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/pci/pci.h"
  22 #include "hw/qdev-properties.h"
  23 #include "intel-hda.h"
  24 #include "migration/vmstate.h"
  25 #include "qemu/module.h"
  26 #include "intel-hda-defs.h"
  27 #include "audio/audio.h"
  28 #include "trace.h"
  29 #include "qom/object.h"
  30
  31 /* -------------------------------------------------------------------------- */
  32
  33 typedef struct desc_param {
  34     uint32_t id;
  35     uint32_t val;
  36 } desc_param;
  37
  38 typedef struct desc_node {
  39     uint32_t nid;
  40     const char *name;
  41     const desc_param *params;
  42     uint32_t nparams;
  43     uint32_t config;
  44     uint32_t pinctl;
  45     uint32_t *conn;
  46     uint32_t stindex;
  47 } desc_node;
  48
  49 typedef struct desc_codec {
  50     const char *name;
  51     uint32_t iid;
  52     const desc_node *nodes;
  53     uint32_t nnodes;
  54 } desc_codec;
  55
  56 static const desc_param* hda_codec_find_param(const desc_node *node, uint32_t id)
  57 {
  58     int i;
  59
  60     for (i = 0; i < node->nparams; i++) {
  61         if (node->params[i].id == id) {
  62             return &node->params[i];
  63         }
  64     }
  65     return NULL;
  66 }
  67
  68 static const desc_node* hda_codec_find_node(const desc_codec *codec, uint32_t nid)
  69 {
  70     int i;
  71
  72     for (i = 0; i < codec->nnodes; i++) {
  73         if (codec->nodes[i].nid == nid) {
  74             return &codec->nodes[i];
  75         }
  76     }
  77     return NULL;
  78 }
  79
  80 static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
  81 {
  82     if (format & AC_FMT_TYPE_NON_PCM) {
  83         return;
  84     }
  85
  86     as->freq = (format & AC_FMT_BASE_44K) ? 44100 : 48000;
  87
  88     switch ((format & AC_FMT_MULT_MASK) >> AC_FMT_MULT_SHIFT) {
  89     case 1: as->freq *= 2; break;
  90     case 2: as->freq *= 3; break;
  91     case 3: as->freq *= 4; break;
  92     }
  93
  94     switch ((format & AC_FMT_DIV_MASK) >> AC_FMT_DIV_SHIFT) {
  95     case 1: as->freq /= 2; break;
  96     case 2: as->freq /= 3; break;
  97     case 3: as->freq /= 4; break;
  98     case 4: as->freq /= 5; break;
  99     case 5: as->freq /= 6; break;
 100     case 6: as->freq /= 7; break;
 101     case 7: as->freq /= 8; break;
 102     }
 103
 104     switch (format & AC_FMT_BITS_MASK) {
 105     case AC_FMT_BITS_8:  as->fmt = AUDIO_FORMAT_S8;  break;
 106     case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
 107     case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
 108     }
 109
 110     as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
 111 }
 112
 113 /* -------------------------------------------------------------------------- */
 114 /*
 115  * HDA codec descriptions
 116  */
 117
 118 /* some defines */
 119
 120 #define QEMU_HDA_ID_VENDOR  0x1af4
 121 #define QEMU_HDA_PCM_FORMATS (AC_SUPPCM_BITS_16 |       \
 122                               0x1fc /* 16 -> 96 kHz */)
 123 #define QEMU_HDA_AMP_NONE    (0)
 124 #define QEMU_HDA_AMP_STEPS   0x4a
 125
 126 #define   PARAM mixemu
 127 #define   HDA_MIXER
 128 #include "hda-codec-common.h"
 129
 130 #define   PARAM nomixemu
 131 #include  "hda-codec-common.h"
 132
 133 #define HDA_TIMER_TICKS (SCALE_MS)
 134 #define B_SIZE sizeof(st->buf)
 135 #define B_MASK (sizeof(st->buf) - 1)
 136
 137 /* -------------------------------------------------------------------------- */
 138
 139 static const char *fmt2name[] = {
 140     [ AUDIO_FORMAT_U8  ] = "PCM-U8",
 141     [ AUDIO_FORMAT_S8  ] = "PCM-S8",
 142     [ AUDIO_FORMAT_U16 ] = "PCM-U16",
 143     [ AUDIO_FORMAT_S16 ] = "PCM-S16",
 144     [ AUDIO_FORMAT_U32 ] = "PCM-U32",
 145     [ AUDIO_FORMAT_S32 ] = "PCM-S32",
 146 };
 147
 148 #define TYPE_HDA_AUDIO "hda-audio"
 149 OBJECT_DECLARE_SIMPLE_TYPE(HDAAudioState, HDA_AUDIO)
 150
 151 typedef struct HDAAudioStream HDAAudioStream;
 152
 153 struct HDAAudioStream {
 154     HDAAudioState *state;
 155     const desc_node *node;
 156     bool output, running;
 157     uint32_t stream;
 158     uint32_t channel;
 159     uint32_t format;
 160     uint32_t gain_left, gain_right;
 161     bool mute_left, mute_right;
 162     struct audsettings as;
 163     union {
 164         SWVoiceIn *in;
 165         SWVoiceOut *out;
 166     } voice;
 167     uint8_t compat_buf[HDA_BUFFER_SIZE];
 168     uint32_t compat_bpos;
 169     uint8_t buf[8192]; /* size must be power of two */
 170     int64_t rpos;
 171     int64_t wpos;
 172     QEMUTimer *buft;
 173     int64_t buft_start;
 174 };
 175
 176 struct HDAAudioState {
 177     HDACodecDevice hda;
 178     const char *name;
 179
 180     QEMUSoundCard card;
 181     const desc_codec *desc;
 182     HDAAudioStream st[4];
 183     bool running_compat[16];
 184     bool running_real[2 * 16];
 185
 186     /* properties */
 187     uint32_t debug;
 188     bool     mixer;
 189     bool     use_timer;
 190 };
 191
 192 static inline int64_t hda_bytes_per_second(HDAAudioStream *st)
 193 {
 194     return 2LL * st->as.nchannels * st->as.freq;
 195 }
 196
 197 static inline void hda_timer_sync_adjust(HDAAudioStream *st, int64_t target_pos)
 198 {
 199     int64_t limit = B_SIZE / 8;
 200     int64_t corr = 0;
 201
 202     if (target_pos > limit) {
 203         corr = HDA_TIMER_TICKS;
 204     }
 205     if (target_pos < -limit) {
 206         corr = -HDA_TIMER_TICKS;
 207     }
 208     if (target_pos < -(2 * limit)) {
 209         corr = -(4 * HDA_TIMER_TICKS);
 210     }
 211     if (corr == 0) {
 212         return;
 213     }
 214
 215     trace_hda_audio_adjust(st->node->name, target_pos);
 216     st->buft_start += corr;
 217 }
 218
 219 static void hda_audio_input_timer(void *opaque)
 220 {
 221     HDAAudioStream *st = opaque;
 222
 223     int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 224
 225     int64_t buft_start = st->buft_start;
 226     int64_t wpos = st->wpos;
 227     int64_t rpos = st->rpos;
 228
 229     int64_t wanted_rpos = hda_bytes_per_second(st) * (now - buft_start)
 230                           / NANOSECONDS_PER_SECOND;
 231     wanted_rpos &= -4; /* IMPORTANT! clip to frames */
 232
 233     if (wanted_rpos <= rpos) {
 234         /* we already transmitted the data */
 235         goto out_timer;
 236     }
 237
 238     int64_t to_transfer = MIN(wpos - rpos, wanted_rpos - rpos);
 239     while (to_transfer) {
 240         uint32_t start = (rpos & B_MASK);
 241         uint32_t chunk = MIN(B_SIZE - start, to_transfer);
 242         int rc = hda_codec_xfer(
 243                 &st->state->hda, st->stream, false, st->buf + start, chunk);
 244         if (!rc) {
 245             break;
 246         }
 247         rpos += chunk;
 248         to_transfer -= chunk;
 249         st->rpos += chunk;
 250     }
 251
 252 out_timer:
 253
 254     if (st->running) {
 255         timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
 256     }
 257 }
 258
 259 static void hda_audio_input_cb(void *opaque, int avail)
 260 {
 261     HDAAudioStream *st = opaque;
 262
 263     int64_t wpos = st->wpos;
 264     int64_t rpos = st->rpos;
 265
 266     int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail);
 267
 268     while (to_transfer) {
 269         uint32_t start = (uint32_t) (wpos & B_MASK);
 270         uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
 271         uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
 272         wpos += read;
 273         to_transfer -= read;
 274         st->wpos += read;
 275         if (chunk != read) {
 276             break;
 277         }
 278     }
 279
 280     hda_timer_sync_adjust(st, -((wpos - rpos) - (B_SIZE >> 1)));
 281 }
 282
 283 static void hda_audio_output_timer(void *opaque)
 284 {
 285     HDAAudioStream *st = opaque;
 286
 287     int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 288
 289     int64_t buft_start = st->buft_start;
 290     int64_t wpos = st->wpos;
 291     int64_t rpos = st->rpos;
 292
 293     int64_t wanted_wpos = hda_bytes_per_second(st) * (now - buft_start)
 294                           / NANOSECONDS_PER_SECOND;
 295     wanted_wpos &= -4; /* IMPORTANT! clip to frames */
 296
 297     if (wanted_wpos <= wpos) {
 298         /* we already received the data */
 299         goto out_timer;
 300     }
 301
 302     int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
 303     while (to_transfer) {
 304         uint32_t start = (wpos & B_MASK);
 305         uint32_t chunk = MIN(B_SIZE - start, to_transfer);
 306         int rc = hda_codec_xfer(
 307                 &st->state->hda, st->stream, true, st->buf + start, chunk);
 308         if (!rc) {
 309             break;
 310         }
 311         wpos += chunk;
 312         to_transfer -= chunk;
 313         st->wpos += chunk;
 314     }
 315
 316 out_timer:
 317
 318     if (st->running) {
 319         timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
 320     }
 321 }
 322
 323 static void hda_audio_output_cb(void *opaque, int avail)
 324 {
 325     HDAAudioStream *st = opaque;
 326
 327     int64_t wpos = st->wpos;
 328     int64_t rpos = st->rpos;
 329
 330     int64_t to_transfer = MIN(wpos - rpos, avail);
 331
 332     if (wpos - rpos == B_SIZE) {
 333         /* drop buffer, reset timer adjust */
 334         st->rpos = 0;
 335         st->wpos = 0;
 336         st->buft_start = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 337         trace_hda_audio_overrun(st->node->name);
 338         return;
 339     }
 340
 341     while (to_transfer) {
 342         uint32_t start = (uint32_t) (rpos & B_MASK);
 343         uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
 344         uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
 345         rpos += written;
 346         to_transfer -= written;
 347         st->rpos += written;
 348         if (chunk != written) {
 349             break;
 350         }
 351     }
 352
 353     hda_timer_sync_adjust(st, (wpos - rpos) - (B_SIZE >> 1));
 354 }
 355
 356 static void hda_audio_compat_input_cb(void *opaque, int avail)
 357 {
 358     HDAAudioStream *st = opaque;
 359     int recv = 0;
 360     int len;
 361     bool rc;
 362
 363     while (avail - recv >= sizeof(st->compat_buf)) {
 364         if (st->compat_bpos != sizeof(st->compat_buf)) {
 365             len = AUD_read(st->voice.in, st->compat_buf + st->compat_bpos,
 366                            sizeof(st->compat_buf) - st->compat_bpos);
 367             st->compat_bpos += len;
 368             recv += len;
 369             if (st->compat_bpos != sizeof(st->compat_buf)) {
 370                 break;
 371             }
 372         }
 373         rc = hda_codec_xfer(&st->state->hda, st->stream, false,
 374                             st->compat_buf, sizeof(st->compat_buf));
 375         if (!rc) {
 376             break;
 377         }
 378         st->compat_bpos = 0;
 379     }
 380 }
 381
 382 static void hda_audio_compat_output_cb(void *opaque, int avail)
 383 {
 384     HDAAudioStream *st = opaque;
 385     int sent = 0;
 386     int len;
 387     bool rc;
 388
 389     while (avail - sent >= sizeof(st->compat_buf)) {
 390         if (st->compat_bpos == sizeof(st->compat_buf)) {
 391             rc = hda_codec_xfer(&st->state->hda, st->stream, true,
 392                                 st->compat_buf, sizeof(st->compat_buf));
 393             if (!rc) {
 394                 break;
 395             }
 396             st->compat_bpos = 0;
 397         }
 398         len = AUD_write(st->voice.out, st->compat_buf + st->compat_bpos,
 399                         sizeof(st->compat_buf) - st->compat_bpos);
 400         st->compat_bpos += len;
 401         sent += len;
 402         if (st->compat_bpos != sizeof(st->compat_buf)) {
 403             break;
 404         }
 405     }
 406 }
 407
 408 static void hda_audio_set_running(HDAAudioStream *st, bool running)
 409 {
 410     if (st->node == NULL) {
 411         return;
 412     }
 413     if (st->running == running) {
 414         return;
 415     }
 416     st->running = running;
 417     trace_hda_audio_running(st->node->name, st->stream, st->running);
 418     if (st->state->use_timer) {
 419         if (running) {
 420             int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 421             st->rpos = 0;
 422             st->wpos = 0;
 423             st->buft_start = now;
 424             timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
 425         } else {
 426             timer_del(st->buft);
 427         }
 428     }
 429     if (st->output) {
 430         AUD_set_active_out(st->voice.out, st->running);
 431     } else {
 432         AUD_set_active_in(st->voice.in, st->running);
 433     }
 434 }
 435
 436 static void hda_audio_set_amp(HDAAudioStream *st)
 437 {
 438     bool muted;
 439     uint32_t left, right;
 440
 441     if (st->node == NULL) {
 442         return;
 443     }
 444
 445     muted = st->mute_left && st->mute_right;
 446     left  = st->mute_left  ? 0 : st->gain_left;
 447     right = st->mute_right ? 0 : st->gain_right;
 448
 449     left = left * 255 / QEMU_HDA_AMP_STEPS;
 450     right = right * 255 / QEMU_HDA_AMP_STEPS;
 451
 452     if (!st->state->mixer) {
 453         return;
 454     }
 455     if (st->output) {
 456         AUD_set_volume_out(st->voice.out, muted, left, right);
 457     } else {
 458         AUD_set_volume_in(st->voice.in, muted, left, right);
 459     }
 460 }
 461
 462 static void hda_audio_setup(HDAAudioStream *st)
 463 {
 464     bool use_timer = st->state->use_timer;
 465     audio_callback_fn cb;
 466
 467     if (st->node == NULL) {
 468         return;
 469     }
 470
 471     trace_hda_audio_format(st->node->name, st->as.nchannels,
 472                            fmt2name[st->as.fmt], st->as.freq);
 473
 474     if (st->output) {
 475         if (use_timer) {
 476             cb = hda_audio_output_cb;
 477             st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
 478                                     hda_audio_output_timer, st);
 479         } else {
 480             cb = hda_audio_compat_output_cb;
 481         }
 482         st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
 483                                      st->node->name, st, cb, &st->as);
 484     } else {
 485         if (use_timer) {
 486             cb = hda_audio_input_cb;
 487             st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
 488                                     hda_audio_input_timer, st);
 489         } else {
 490             cb = hda_audio_compat_input_cb;
 491         }
 492         st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
 493                                    st->node->name, st, cb, &st->as);
 494     }
 495 }
 496
 497 static void hda_audio_command(HDACodecDevice *hda, uint32_t nid, uint32_t data)
 498 {
 499     HDAAudioState *a = HDA_AUDIO(hda);
 500     HDAAudioStream *st;
 501     const desc_node *node = NULL;
 502     const desc_param *param;
 503     uint32_t verb, payload, response, count, shift;
 504
 505     if ((data & 0x70000) == 0x70000) {
 506         /* 12/8 id/payload */
 507         verb = (data >> 8) & 0xfff;
 508         payload = data & 0x00ff;
 509     } else {
 510         /* 4/16 id/payload */
 511         verb = (data >> 8) & 0xf00;
 512         payload = data & 0xffff;
 513     }
 514
 515     node = hda_codec_find_node(a->desc, nid);
 516     if (node == NULL) {
 517         goto fail;
 518     }
 519     dprint(a, 2, "%s: nid %d (%s), verb 0x%x, payload 0x%x\n",
 520            __func__, nid, node->name, verb, payload);
 521
 522     switch (verb) {
 523     /* all nodes */
 524     case AC_VERB_PARAMETERS:
 525         param = hda_codec_find_param(node, payload);
 526         if (param == NULL) {
 527             goto fail;
 528         }
 529         hda_codec_response(hda, true, param->val);
 530         break;
 531     case AC_VERB_GET_SUBSYSTEM_ID:
 532         hda_codec_response(hda, true, a->desc->iid);
 533         break;
 534
 535     /* all functions */
 536     case AC_VERB_GET_CONNECT_LIST:
 537         param = hda_codec_find_param(node, AC_PAR_CONNLIST_LEN);
 538         count = param ? param->val : 0;
 539         response = 0;
 540         shift = 0;
 541         while (payload < count && shift < 32) {
 542             response |= node->conn[payload] << shift;
 543             payload++;
 544             shift += 8;
 545         }
 546         hda_codec_response(hda, true, response);
 547         break;
 548
 549     /* pin widget */
 550     case AC_VERB_GET_CONFIG_DEFAULT:
 551         hda_codec_response(hda, true, node->config);
 552         break;
 553     case AC_VERB_GET_PIN_WIDGET_CONTROL:
 554         hda_codec_response(hda, true, node->pinctl);
 555         break;
 556     case AC_VERB_SET_PIN_WIDGET_CONTROL:
 557         if (node->pinctl != payload) {
 558             dprint(a, 1, "unhandled pin control bit\n");
 559         }
 560         hda_codec_response(hda, true, 0);
 561         break;
 562
 563     /* audio in/out widget */
 564     case AC_VERB_SET_CHANNEL_STREAMID:
 565         st = a->st + node->stindex;
 566         if (st->node == NULL) {
 567             goto fail;
 568         }
 569         hda_audio_set_running(st, false);
 570         st->stream = (payload >> 4) & 0x0f;
 571         st->channel = payload & 0x0f;
 572         dprint(a, 2, "%s: stream %d, channel %d\n",
 573                st->node->name, st->stream, st->channel);
 574         hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
 575         hda_codec_response(hda, true, 0);
 576         break;
 577     case AC_VERB_GET_CONV:
 578         st = a->st + node->stindex;
 579         if (st->node == NULL) {
 580             goto fail;
 581         }
 582         response = st->stream << 4 | st->channel;
 583         hda_codec_response(hda, true, response);
 584         break;
 585     case AC_VERB_SET_STREAM_FORMAT:
 586         st = a->st + node->stindex;
 587         if (st->node == NULL) {
 588             goto fail;
 589         }
 590         st->format = payload;
 591         hda_codec_parse_fmt(st->format, &st->as);
 592         hda_audio_setup(st);
 593         hda_codec_response(hda, true, 0);
 594         break;
 595     case AC_VERB_GET_STREAM_FORMAT:
 596         st = a->st + node->stindex;
 597         if (st->node == NULL) {
 598             goto fail;
 599         }
 600         hda_codec_response(hda, true, st->format);
 601         break;
 602     case AC_VERB_GET_AMP_GAIN_MUTE:
 603         st = a->st + node->stindex;
 604         if (st->node == NULL) {
 605             goto fail;
 606         }
 607         if (payload & AC_AMP_GET_LEFT) {
 608             response = st->gain_left | (st->mute_left ? AC_AMP_MUTE : 0);
 609         } else {
 610             response = st->gain_right | (st->mute_right ? AC_AMP_MUTE : 0);
 611         }
 612         hda_codec_response(hda, true, response);
 613         break;
 614     case AC_VERB_SET_AMP_GAIN_MUTE:
 615         st = a->st + node->stindex;
 616         if (st->node == NULL) {
 617             goto fail;
 618         }
 619         dprint(a, 1, "amp (%s): %s%s%s%s index %d  gain %3d %s\n",
 620                st->node->name,
 621                (payload & AC_AMP_SET_OUTPUT) ? "o" : "-",
 622                (payload & AC_AMP_SET_INPUT)  ? "i" : "-",
 623                (payload & AC_AMP_SET_LEFT)   ? "l" : "-",
 624                (payload & AC_AMP_SET_RIGHT)  ? "r" : "-",
 625                (payload & AC_AMP_SET_INDEX) >> AC_AMP_SET_INDEX_SHIFT,
 626                (payload & AC_AMP_GAIN),
 627                (payload & AC_AMP_MUTE) ? "muted" : "");
 628         if (payload & AC_AMP_SET_LEFT) {
 629             st->gain_left = payload & AC_AMP_GAIN;
 630             st->mute_left = payload & AC_AMP_MUTE;
 631         }
 632         if (payload & AC_AMP_SET_RIGHT) {
 633             st->gain_right = payload & AC_AMP_GAIN;
 634             st->mute_right = payload & AC_AMP_MUTE;
 635         }
 636         hda_audio_set_amp(st);
 637         hda_codec_response(hda, true, 0);
 638         break;
 639
 640     /* not supported */
 641     case AC_VERB_SET_POWER_STATE:
 642     case AC_VERB_GET_POWER_STATE:
 643     case AC_VERB_GET_SDI_SELECT:
 644         hda_codec_response(hda, true, 0);
 645         break;
 646     default:
 647         goto fail;
 648     }
 649     return;
 650
 651 fail:
 652     dprint(a, 1, "%s: not handled: nid %d (%s), verb 0x%x, payload 0x%x\n",
 653            __func__, nid, node ? node->name : "?", verb, payload);
 654     hda_codec_response(hda, true, 0);
 655 }
 656
 657 static void hda_audio_stream(HDACodecDevice *hda, uint32_t stnr, bool running, bool output)
 658 {
 659     HDAAudioState *a = HDA_AUDIO(hda);
 660     int s;
 661
 662     a->running_compat[stnr] = running;
 663     a->running_real[output * 16 + stnr] = running;
 664     for (s = 0; s < ARRAY_SIZE(a->st); s++) {
 665         if (a->st[s].node == NULL) {
 666             continue;
 667         }
 668         if (a->st[s].output != output) {
 669             continue;
 670         }
 671         if (a->st[s].stream != stnr) {
 672             continue;
 673         }
 674         hda_audio_set_running(&a->st[s], running);
 675     }
 676 }
 677
 678 static void hda_audio_init(HDACodecDevice *hda,
 679                            const struct desc_codec *desc,
 680                            Error **errp)
 681 {
 682     HDAAudioState *a = HDA_AUDIO(hda);
 683     HDAAudioStream *st;
 684     const desc_node *node;
 685     const desc_param *param;
 686     uint32_t i, type;
 687
 688     if (!AUD_register_card("hda", &a->card, errp)) {
 689         return;
 690     }
 691
 692     a->desc = desc;
 693     a->name = object_get_typename(OBJECT(a));
 694     dprint(a, 1, "%s: cad %d\n", __func__, a->hda.cad);
 695
 696     for (i = 0; i < a->desc->nnodes; i++) {
 697         node = a->desc->nodes + i;
 698         param = hda_codec_find_param(node, AC_PAR_AUDIO_WIDGET_CAP);
 699         if (param == NULL) {
 700             continue;
 701         }
 702         type = (param->val & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
 703         switch (type) {
 704         case AC_WID_AUD_OUT:
 705         case AC_WID_AUD_IN:
 706             assert(node->stindex < ARRAY_SIZE(a->st));
 707             st = a->st + node->stindex;
 708             st->state = a;
 709             st->node = node;
 710             if (type == AC_WID_AUD_OUT) {
 711                 /* unmute output by default */
 712                 st->gain_left = QEMU_HDA_AMP_STEPS;
 713                 st->gain_right = QEMU_HDA_AMP_STEPS;
 714                 st->compat_bpos = sizeof(st->compat_buf);
 715                 st->output = true;
 716             } else {
 717                 st->output = false;
 718             }
 719             st->format = AC_FMT_TYPE_PCM | AC_FMT_BITS_16 |
 720                 (1 << AC_FMT_CHAN_SHIFT);
 721             hda_codec_parse_fmt(st->format, &st->as);
 722             hda_audio_setup(st);
 723             break;
 724         }
 725     }
 726 }
 727
 728 static void hda_audio_exit(HDACodecDevice *hda)
 729 {
 730     HDAAudioState *a = HDA_AUDIO(hda);
 731     HDAAudioStream *st;
 732     int i;
 733
 734     dprint(a, 1, "%s\n", __func__);
 735     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
 736         st = a->st + i;
 737         if (st->node == NULL) {
 738             continue;
 739         }
 740         if (a->use_timer) {
 741             timer_del(st->buft);
 742         }
 743         if (st->output) {
 744             AUD_close_out(&a->card, st->voice.out);
 745         } else {
 746             AUD_close_in(&a->card, st->voice.in);
 747         }
 748     }
 749     AUD_remove_card(&a->card);
 750 }
 751
 752 static int hda_audio_post_load(void *opaque, int version)
 753 {
 754     HDAAudioState *a = opaque;
 755     HDAAudioStream *st;
 756     int i;
 757
 758     dprint(a, 1, "%s\n", __func__);
 759     if (version == 1) {
 760         /* assume running_compat[] is for output streams */
 761         for (i = 0; i < ARRAY_SIZE(a->running_compat); i++)
 762             a->running_real[16 + i] = a->running_compat[i];
 763     }
 764
 765     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
 766         st = a->st + i;
 767         if (st->node == NULL)
 768             continue;
 769         hda_codec_parse_fmt(st->format, &st->as);
 770         hda_audio_setup(st);
 771         hda_audio_set_amp(st);
 772         hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
 773     }
 774     return 0;
 775 }
 776
 777 static void hda_audio_reset(DeviceState *dev)
 778 {
 779     HDAAudioState *a = HDA_AUDIO(dev);
 780     HDAAudioStream *st;
 781     int i;
 782
 783     dprint(a, 1, "%s\n", __func__);
 784     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
 785         st = a->st + i;
 786         if (st->node != NULL) {
 787             hda_audio_set_running(st, false);
 788         }
 789     }
 790 }
 791
 792 static bool vmstate_hda_audio_stream_buf_needed(void *opaque)
 793 {
 794     HDAAudioStream *st = opaque;
 795     return st->state && st->state->use_timer;
 796 }
 797
 798 static const VMStateDescription vmstate_hda_audio_stream_buf = {
 799     .name = "hda-audio-stream/buffer",
 800     .version_id = 1,
 801     .needed = vmstate_hda_audio_stream_buf_needed,
 802     .fields = (VMStateField[]) {
 803         VMSTATE_BUFFER(buf, HDAAudioStream),
 804         VMSTATE_INT64(rpos, HDAAudioStream),
 805         VMSTATE_INT64(wpos, HDAAudioStream),
 806         VMSTATE_TIMER_PTR(buft, HDAAudioStream),
 807         VMSTATE_INT64(buft_start, HDAAudioStream),
 808         VMSTATE_END_OF_LIST()
 809     }
 810 };
 811
 812 static const VMStateDescription vmstate_hda_audio_stream = {
 813     .name = "hda-audio-stream",
 814     .version_id = 1,
 815     .fields = (VMStateField[]) {
 816         VMSTATE_UINT32(stream, HDAAudioStream),
 817         VMSTATE_UINT32(channel, HDAAudioStream),
 818         VMSTATE_UINT32(format, HDAAudioStream),
 819         VMSTATE_UINT32(gain_left, HDAAudioStream),
 820         VMSTATE_UINT32(gain_right, HDAAudioStream),
 821         VMSTATE_BOOL(mute_left, HDAAudioStream),
 822         VMSTATE_BOOL(mute_right, HDAAudioStream),
 823         VMSTATE_UINT32(compat_bpos, HDAAudioStream),
 824         VMSTATE_BUFFER(compat_buf, HDAAudioStream),
 825         VMSTATE_END_OF_LIST()
 826     },
 827     .subsections = (const VMStateDescription * []) {
 828         &vmstate_hda_audio_stream_buf,
 829         NULL
 830     }
 831 };
 832
 833 static const VMStateDescription vmstate_hda_audio = {
 834     .name = "hda-audio",
 835     .version_id = 2,
 836     .post_load = hda_audio_post_load,
 837     .fields = (VMStateField[]) {
 838         VMSTATE_STRUCT_ARRAY(st, HDAAudioState, 4, 0,
 839                              vmstate_hda_audio_stream,
 840                              HDAAudioStream),
 841         VMSTATE_BOOL_ARRAY(running_compat, HDAAudioState, 16),
 842         VMSTATE_BOOL_ARRAY_V(running_real, HDAAudioState, 2 * 16, 2),
 843         VMSTATE_END_OF_LIST()
 844     }
 845 };
 846
 847 static Property hda_audio_properties[] = {
 848     DEFINE_AUDIO_PROPERTIES(HDAAudioState, card),
 849     DEFINE_PROP_UINT32("debug", HDAAudioState, debug,   0),
 850     DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer,  true),
 851     DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer,  true),
 852     DEFINE_PROP_END_OF_LIST(),
 853 };
 854
 855 static void hda_audio_init_output(HDACodecDevice *hda, Error **errp)
 856 {
 857     HDAAudioState *a = HDA_AUDIO(hda);
 858     const struct desc_codec *desc = &output_nomixemu;
 859
 860     if (!a->mixer) {
 861         desc = &output_mixemu;
 862     }
 863
 864     hda_audio_init(hda, desc, errp);
 865 }
 866
 867 static void hda_audio_init_duplex(HDACodecDevice *hda, Error **errp)
 868 {
 869     HDAAudioState *a = HDA_AUDIO(hda);
 870     const struct desc_codec *desc = &duplex_nomixemu;
 871
 872     if (!a->mixer) {
 873         desc = &duplex_mixemu;
 874     }
 875
 876     hda_audio_init(hda, desc, errp);
 877 }
 878
 879 static void hda_audio_init_micro(HDACodecDevice *hda, Error **errp)
 880 {
 881     HDAAudioState *a = HDA_AUDIO(hda);
 882     const struct desc_codec *desc = &micro_nomixemu;
 883
 884     if (!a->mixer) {
 885         desc = &micro_mixemu;
 886     }
 887
 888     hda_audio_init(hda, desc, errp);
 889 }
 890
 891 static void hda_audio_base_class_init(ObjectClass *klass, void *data)
 892 {
 893     DeviceClass *dc = DEVICE_CLASS(klass);
 894     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
 895
 896     k->exit = hda_audio_exit;
 897     k->command = hda_audio_command;
 898     k->stream = hda_audio_stream;
 899     set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
 900     dc->reset = hda_audio_reset;
 901     dc->vmsd = &vmstate_hda_audio;
 902     device_class_set_props(dc, hda_audio_properties);
 903 }
 904
 905 static const TypeInfo hda_audio_info = {
 906     .name          = TYPE_HDA_AUDIO,
 907     .parent        = TYPE_HDA_CODEC_DEVICE,
 908     .instance_size = sizeof(HDAAudioState),
 909     .class_init    = hda_audio_base_class_init,
 910     .abstract      = true,
 911 };
 912
 913 static void hda_audio_output_class_init(ObjectClass *klass, void *data)
 914 {
 915     DeviceClass *dc = DEVICE_CLASS(klass);
 916     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
 917
 918     k->init = hda_audio_init_output;
 919     dc->desc = "HDA Audio Codec, output-only (line-out)";
 920 }
 921
 922 static const TypeInfo hda_audio_output_info = {
 923     .name          = "hda-output",
 924     .parent        = TYPE_HDA_AUDIO,
 925     .class_init    = hda_audio_output_class_init,
 926 };
 927
 928 static void hda_audio_duplex_class_init(ObjectClass *klass, void *data)
 929 {
 930     DeviceClass *dc = DEVICE_CLASS(klass);
 931     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
 932
 933     k->init = hda_audio_init_duplex;
 934     dc->desc = "HDA Audio Codec, duplex (line-out, line-in)";
 935 }
 936
 937 static const TypeInfo hda_audio_duplex_info = {
 938     .name          = "hda-duplex",
 939     .parent        = TYPE_HDA_AUDIO,
 940     .class_init    = hda_audio_duplex_class_init,
 941 };
 942
 943 static void hda_audio_micro_class_init(ObjectClass *klass, void *data)
 944 {
 945     DeviceClass *dc = DEVICE_CLASS(klass);
 946     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
 947
 948     k->init = hda_audio_init_micro;
 949     dc->desc = "HDA Audio Codec, duplex (speaker, microphone)";
 950 }
 951
 952 static const TypeInfo hda_audio_micro_info = {
 953     .name          = "hda-micro",
 954     .parent        = TYPE_HDA_AUDIO,
 955     .class_init    = hda_audio_micro_class_init,
 956 };
 957
 958 static void hda_audio_register_types(void)
 959 {
 960     type_register_static(&hda_audio_info);
 961     type_register_static(&hda_audio_output_info);
 962     type_register_static(&hda_audio_duplex_info);
 963     type_register_static(&hda_audio_micro_info);
 964 }
 965
 966 type_init(hda_audio_register_types)