aboutsummaryrefslogtreecommitdiff
path: root/hw/audio/hda-codec.c
blob: c25bfa38b143df22781414a28ecc35633fa87ef5 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
/*
 * Copyright (C) 2010 Red Hat, Inc.
 *
 * written by Gerd Hoffmann <kraxel@redhat.com>
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU General Public License as
 * published by the Free Software Foundation; either version 2 or
 * (at your option) version 3 of the License.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, see <http://www.gnu.org/licenses/>.
 */

#include "qemu/osdep.h"
#include "hw/hw.h"
#include "hw/pci/pci.h"
#include "intel-hda.h"
#include "intel-hda-defs.h"
#include "audio/audio.h"
#include "trace.h"

/* -------------------------------------------------------------------------- */

typedef struct desc_param {
    uint32_t id;
    uint32_t val;
} desc_param;

typedef struct desc_node {
    uint32_t nid;
    const char *name;
    const desc_param *params;
    uint32_t nparams;
    uint32_t config;
    uint32_t pinctl;
    uint32_t *conn;
    uint32_t stindex;
} desc_node;

typedef struct desc_codec {
    const char *name;
    uint32_t iid;
    const desc_node *nodes;
    uint32_t nnodes;
} desc_codec;

static const desc_param* hda_codec_find_param(const desc_node *node, uint32_t id)
{
    int i;

    for (i = 0; i < node->nparams; i++) {
        if (node->params[i].id == id) {
            return &node->params[i];
        }
    }
    return NULL;
}

static const desc_node* hda_codec_find_node(const desc_codec *codec, uint32_t nid)
{
    int i;

    for (i = 0; i < codec->nnodes; i++) {
        if (codec->nodes[i].nid == nid) {
            return &codec->nodes[i];
        }
    }
    return NULL;
}

static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
{
    if (format & AC_FMT_TYPE_NON_PCM) {
        return;
    }

    as->freq = (format & AC_FMT_BASE_44K) ? 44100 : 48000;

    switch ((format & AC_FMT_MULT_MASK) >> AC_FMT_MULT_SHIFT) {
    case 1: as->freq *= 2; break;
    case 2: as->freq *= 3; break;
    case 3: as->freq *= 4; break;
    }

    switch ((format & AC_FMT_DIV_MASK) >> AC_FMT_DIV_SHIFT) {
    case 1: as->freq /= 2; break;
    case 2: as->freq /= 3; break;
    case 3: as->freq /= 4; break;
    case 4: as->freq /= 5; break;
    case 5: as->freq /= 6; break;
    case 6: as->freq /= 7; break;
    case 7: as->freq /= 8; break;
    }

    switch (format & AC_FMT_BITS_MASK) {
    case AC_FMT_BITS_8:  as->fmt = AUDIO_FORMAT_S8;  break;
    case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
    case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
    }

    as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
}

/* -------------------------------------------------------------------------- */
/*
 * HDA codec descriptions
 */

/* some defines */

#define QEMU_HDA_ID_VENDOR  0x1af4
#define QEMU_HDA_PCM_FORMATS (AC_SUPPCM_BITS_16 |       \
                              0x1fc /* 16 -> 96 kHz */)
#define QEMU_HDA_AMP_NONE    (0)
#define QEMU_HDA_AMP_STEPS   0x4a

#define   PARAM mixemu
#define   HDA_MIXER
#include "hda-codec-common.h"

#define   PARAM nomixemu
#include  "hda-codec-common.h"

#define HDA_TIMER_TICKS (SCALE_MS)
#define B_SIZE sizeof(st->buf)
#define B_MASK (sizeof(st->buf) - 1)

/* -------------------------------------------------------------------------- */

static const char *fmt2name[] = {
    [ AUDIO_FORMAT_U8  ] = "PCM-U8",
    [ AUDIO_FORMAT_S8  ] = "PCM-S8",
    [ AUDIO_FORMAT_U16 ] = "PCM-U16",
    [ AUDIO_FORMAT_S16 ] = "PCM-S16",
    [ AUDIO_FORMAT_U32 ] = "PCM-U32",
    [ AUDIO_FORMAT_S32 ] = "PCM-S32",
};

typedef struct HDAAudioState HDAAudioState;
typedef struct HDAAudioStream HDAAudioStream;

struct HDAAudioStream {
    HDAAudioState *state;
    const desc_node *node;
    bool output, running;
    uint32_t stream;
    uint32_t channel;
    uint32_t format;
    uint32_t gain_left, gain_right;
    bool mute_left, mute_right;
    struct audsettings as;
    union {
        SWVoiceIn *in;
        SWVoiceOut *out;
    } voice;
    uint8_t compat_buf[HDA_BUFFER_SIZE];
    uint32_t compat_bpos;
    uint8_t buf[8192]; /* size must be power of two */
    int64_t rpos;
    int64_t wpos;
    QEMUTimer *buft;
    int64_t buft_start;
};

#define TYPE_HDA_AUDIO "hda-audio"
#define HDA_AUDIO(obj) OBJECT_CHECK(HDAAudioState, (obj), TYPE_HDA_AUDIO)

struct HDAAudioState {
    HDACodecDevice hda;
    const char *name;

    QEMUSoundCard card;
    const desc_codec *desc;
    HDAAudioStream st[4];
    bool running_compat[16];
    bool running_real[2 * 16];

    /* properties */
    uint32_t debug;
    bool     mixer;
    bool     use_timer;
};

static inline int64_t hda_bytes_per_second(HDAAudioStream *st)
{
    return 2LL * st->as.nchannels * st->as.freq;
}

static inline void hda_timer_sync_adjust(HDAAudioStream *st, int64_t target_pos)
{
    int64_t limit = B_SIZE / 8;
    int64_t corr = 0;

    if (target_pos > limit) {
        corr = HDA_TIMER_TICKS;
    }
    if (target_pos < -limit) {
        corr = -HDA_TIMER_TICKS;
    }
    if (target_pos < -(2 * limit)) {
        corr = -(4 * HDA_TIMER_TICKS);
    }
    if (corr == 0) {
        return;
    }

    trace_hda_audio_adjust(st->node->name, target_pos);
    st->buft_start += corr;
}

static void hda_audio_input_timer(void *opaque)
{
    HDAAudioStream *st = opaque;

    int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);

    int64_t buft_start = st->buft_start;
    int64_t wpos = st->wpos;
    int64_t rpos = st->rpos;

    int64_t wanted_rpos = hda_bytes_per_second(st) * (now - buft_start)
                          / NANOSECONDS_PER_SECOND;
    wanted_rpos &= -4; /* IMPORTANT! clip to frames */

    if (wanted_rpos <= rpos) {
        /* we already transmitted the data */
        goto out_timer;
    }

    int64_t to_transfer = audio_MIN(wpos - rpos, wanted_rpos - rpos);
    while (to_transfer) {
        uint32_t start = (rpos & B_MASK);
        uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
        int rc = hda_codec_xfer(
                &st->state->hda, st->stream, false, st->buf + start, chunk);
        if (!rc) {
            break;
        }
        rpos += chunk;
        to_transfer -= chunk;
        st->rpos += chunk;
    }

out_timer:

    if (st->running) {
        timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
    }
}

static void hda_audio_input_cb(void *opaque, int avail)
{
    HDAAudioStream *st = opaque;

    int64_t wpos = st->wpos;
    int64_t rpos = st->rpos;

    int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), avail);

    hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1)));

    while (to_transfer) {
        uint32_t start = (uint32_t) (wpos & B_MASK);
        uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
        uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
        wpos += read;
        to_transfer -= read;
        st->wpos += read;
        if (chunk != read) {
            break;
        }
    }
}

static void hda_audio_output_timer(void *opaque)
{
    HDAAudioStream *st = opaque;

    int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);

    int64_t buft_start = st->buft_start;
    int64_t wpos = st->wpos;
    int64_t rpos = st->rpos;

    int64_t wanted_wpos = hda_bytes_per_second(st) * (now - buft_start)
                          / NANOSECONDS_PER_SECOND;
    wanted_wpos &= -4; /* IMPORTANT! clip to frames */

    if (wanted_wpos <= wpos) {
        /* we already received the data */
        goto out_timer;
    }

    int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
    while (to_transfer) {
        uint32_t start = (wpos & B_MASK);
        uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
        int rc = hda_codec_xfer(
                &st->state->hda, st->stream, true, st->buf + start, chunk);
        if (!rc) {
            break;
        }
        wpos += chunk;
        to_transfer -= chunk;
        st->wpos += chunk;
    }

out_timer:

    if (st->running) {
        timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
    }
}

static void hda_audio_output_cb(void *opaque, int avail)
{
    HDAAudioStream *st = opaque;

    int64_t wpos = st->wpos;
    int64_t rpos = st->rpos;

    int64_t to_transfer = audio_MIN(wpos - rpos, avail);

    if (wpos - rpos == B_SIZE) {
        /* drop buffer, reset timer adjust */
        st->rpos = 0;
        st->wpos = 0;
        st->buft_start = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
        trace_hda_audio_overrun(st->node->name);
        return;
    }

    hda_timer_sync_adjust(st, (wpos - rpos) - to_transfer - (B_SIZE >> 1));

    while (to_transfer) {
        uint32_t start = (uint32_t) (rpos & B_MASK);
        uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
        uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
        rpos += written;
        to_transfer -= written;
        st->rpos += written;
        if (chunk != written) {
            break;
        }
    }
}

static void hda_audio_compat_input_cb(void *opaque, int avail)
{
    HDAAudioStream *st = opaque;
    int recv = 0;
    int len;
    bool rc;

    while (avail - recv >= sizeof(st->compat_buf)) {
        if (st->compat_bpos != sizeof(st->compat_buf)) {
            len = AUD_read(st->voice.in, st->compat_buf + st->compat_bpos,
                           sizeof(st->compat_buf) - st->compat_bpos);
            st->compat_bpos += len;
            recv += len;
            if (st->compat_bpos != sizeof(st->compat_buf)) {
                break;
            }
        }
        rc = hda_codec_xfer(&st->state->hda, st->stream, false,
                            st->compat_buf, sizeof(st->compat_buf));
        if (!rc) {
            break;
        }
        st->compat_bpos = 0;
    }
}

static void hda_audio_compat_output_cb(void *opaque, int avail)
{
    HDAAudioStream *st = opaque;
    int sent = 0;
    int len;
    bool rc;

    while (avail - sent >= sizeof(st->compat_buf)) {
        if (st->compat_bpos == sizeof(st->compat_buf)) {
            rc = hda_codec_xfer(&st->state->hda, st->stream, true,
                                st->compat_buf, sizeof(st->compat_buf));
            if (!rc) {
                break;
            }
            st->compat_bpos = 0;
        }
        len = AUD_write(st->voice.out, st->compat_buf + st->compat_bpos,
                        sizeof(st->compat_buf) - st->compat_bpos);
        st->compat_bpos += len;
        sent += len;
        if (st->compat_bpos != sizeof(st->compat_buf)) {
            break;
        }
    }
}

static void hda_audio_set_running(HDAAudioStream *st, bool running)
{
    if (st->node == NULL) {
        return;
    }
    if (st->running == running) {
        return;
    }
    st->running = running;
    trace_hda_audio_running(st->node->name, st->stream, st->running);
    if (st->state->use_timer) {
        if (running) {
            int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
            st->rpos = 0;
            st->wpos = 0;
            st->buft_start = now;
            timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
        } else {
            timer_del(st->buft);
        }
    }
    if (st->output) {
        AUD_set_active_out(st->voice.out, st->running);
    } else {
        AUD_set_active_in(st->voice.in, st->running);
    }
}

static void hda_audio_set_amp(HDAAudioStream *st)
{
    bool muted;
    uint32_t left, right;

    if (st->node == NULL) {
        return;
    }

    muted = st->mute_left && st->mute_right;
    left  = st->mute_left  ? 0 : st->gain_left;
    right = st->mute_right ? 0 : st->gain_right;

    left = left * 255 / QEMU_HDA_AMP_STEPS;
    right = right * 255 / QEMU_HDA_AMP_STEPS;

    if (!st->state->mixer) {
        return;
    }
    if (st->output) {
        AUD_set_volume_out(st->voice.out, muted, left, right);
    } else {
        AUD_set_volume_in(st->voice.in, muted, left, right);
    }
}

static void hda_audio_setup(HDAAudioStream *st)
{
    bool use_timer = st->state->use_timer;
    audio_callback_fn cb;

    if (st->node == NULL) {
        return;
    }

    trace_hda_audio_format(st->node->name, st->as.nchannels,
                           fmt2name[st->as.fmt], st->as.freq);

    if (st->output) {
        if (use_timer) {
            cb = hda_audio_output_cb;
            st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
                                    hda_audio_output_timer, st);
        } else {
            cb = hda_audio_compat_output_cb;
        }
        st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
                                     st->node->name, st, cb, &st->as);
    } else {
        if (use_timer) {
            cb = hda_audio_input_cb;
            st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
                                    hda_audio_input_timer, st);
        } else {
            cb = hda_audio_compat_input_cb;
        }
        st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
                                   st->node->name, st, cb, &st->as);
    }
}

static void hda_audio_command(HDACodecDevice *hda, uint32_t nid, uint32_t data)
{
    HDAAudioState *a = HDA_AUDIO(hda);
    HDAAudioStream *st;
    const desc_node *node = NULL;
    const desc_param *param;
    uint32_t verb, payload, response, count, shift;

    if ((data & 0x70000) == 0x70000) {
        /* 12/8 id/payload */
        verb = (data >> 8) & 0xfff;
        payload = data & 0x00ff;
    } else {
        /* 4/16 id/payload */
        verb = (data >> 8) & 0xf00;
        payload = data & 0xffff;
    }

    node = hda_codec_find_node(a->desc, nid);
    if (node == NULL) {
        goto fail;
    }
    dprint(a, 2, "%s: nid %d (%s), verb 0x%x, payload 0x%x\n",
           __func__, nid, node->name, verb, payload);

    switch (verb) {
    /* all nodes */
    case AC_VERB_PARAMETERS:
        param = hda_codec_find_param(node, payload);
        if (param == NULL) {
            goto fail;
        }
        hda_codec_response(hda, true, param->val);
        break;
    case AC_VERB_GET_SUBSYSTEM_ID:
        hda_codec_response(hda, true, a->desc->iid);
        break;

    /* all functions */
    case AC_VERB_GET_CONNECT_LIST:
        param = hda_codec_find_param(node, AC_PAR_CONNLIST_LEN);
        count = param ? param->val : 0;
        response = 0;
        shift = 0;
        while (payload < count && shift < 32) {
            response |= node->conn[payload] << shift;
            payload++;
            shift += 8;
        }
        hda_codec_response(hda, true, response);
        break;

    /* pin widget */
    case AC_VERB_GET_CONFIG_DEFAULT:
        hda_codec_response(hda, true, node->config);
        break;
    case AC_VERB_GET_PIN_WIDGET_CONTROL:
        hda_codec_response(hda, true, node->pinctl);
        break;
    case AC_VERB_SET_PIN_WIDGET_CONTROL:
        if (node->pinctl != payload) {
            dprint(a, 1, "unhandled pin control bit\n");
        }
        hda_codec_response(hda, true, 0);
        break;

    /* audio in/out widget */
    case AC_VERB_SET_CHANNEL_STREAMID:
        st = a->st + node->stindex;
        if (st->node == NULL) {
            goto fail;
        }
        hda_audio_set_running(st, false);
        st->stream = (payload >> 4) & 0x0f;
        st->channel = payload & 0x0f;
        dprint(a, 2, "%s: stream %d, channel %d\n",
               st->node->name, st->stream, st->channel);
        hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
        hda_codec_response(hda, true, 0);
        break;
    case AC_VERB_GET_CONV:
        st = a->st + node->stindex;
        if (st->node == NULL) {
            goto fail;
        }
        response = st->stream << 4 | st->channel;
        hda_codec_response(hda, true, response);
        break;
    case AC_VERB_SET_STREAM_FORMAT:
        st = a->st + node->stindex;
        if (st->node == NULL) {
            goto fail;
        }
        st->format = payload;
        hda_codec_parse_fmt(st->format, &st->as);
        hda_audio_setup(st);
        hda_codec_response(hda, true, 0);
        break;
    case AC_VERB_GET_STREAM_FORMAT:
        st = a->st + node->stindex;
        if (st->node == NULL) {
            goto fail;
        }
        hda_codec_response(hda, true, st->format);
        break;
    case AC_VERB_GET_AMP_GAIN_MUTE:
        st = a->st + node->stindex;
        if (st->node == NULL) {
            goto fail;
        }
        if (payload & AC_AMP_GET_LEFT) {
            response = st->gain_left | (st->mute_left ? AC_AMP_MUTE : 0);
        } else {
            response = st->gain_right | (st->mute_right ? AC_AMP_MUTE : 0);
        }
        hda_codec_response(hda, true, response);
        break;
    case AC_VERB_SET_AMP_GAIN_MUTE:
        st = a->st + node->stindex;
        if (st->node == NULL) {
            goto fail;
        }
        dprint(a, 1, "amp (%s): %s%s%s%s index %d  gain %3d %s\n",
               st->node->name,
               (payload & AC_AMP_SET_OUTPUT) ? "o" : "-",
               (payload & AC_AMP_SET_INPUT)  ? "i" : "-",
               (payload & AC_AMP_SET_LEFT)   ? "l" : "-",
               (payload & AC_AMP_SET_RIGHT)  ? "r" : "-",
               (payload & AC_AMP_SET_INDEX) >> AC_AMP_SET_INDEX_SHIFT,
               (payload & AC_AMP_GAIN),
               (payload & AC_AMP_MUTE) ? "muted" : "");
        if (payload & AC_AMP_SET_LEFT) {
            st->gain_left = payload & AC_AMP_GAIN;
            st->mute_left = payload & AC_AMP_MUTE;
        }
        if (payload & AC_AMP_SET_RIGHT) {
            st->gain_right = payload & AC_AMP_GAIN;
            st->mute_right = payload & AC_AMP_MUTE;
        }
        hda_audio_set_amp(st);
        hda_codec_response(hda, true, 0);
        break;

    /* not supported */
    case AC_VERB_SET_POWER_STATE:
    case AC_VERB_GET_POWER_STATE:
    case AC_VERB_GET_SDI_SELECT:
        hda_codec_response(hda, true, 0);
        break;
    default:
        goto fail;
    }
    return;

fail:
    dprint(a, 1, "%s: not handled: nid %d (%s), verb 0x%x, payload 0x%x\n",
           __func__, nid, node ? node->name : "?", verb, payload);
    hda_codec_response(hda, true, 0);
}

static void hda_audio_stream(HDACodecDevice *hda, uint32_t stnr, bool running, bool output)
{
    HDAAudioState *a = HDA_AUDIO(hda);
    int s;

    a->running_compat[stnr] = running;
    a->running_real[output * 16 + stnr] = running;
    for (s = 0; s < ARRAY_SIZE(a->st); s++) {
        if (a->st[s].node == NULL) {
            continue;
        }
        if (a->st[s].output != output) {
            continue;
        }
        if (a->st[s].stream != stnr) {
            continue;
        }
        hda_audio_set_running(&a->st[s], running);
    }
}

static int hda_audio_init(HDACodecDevice *hda, const struct desc_codec *desc)
{
    HDAAudioState *a = HDA_AUDIO(hda);
    HDAAudioStream *st;
    const desc_node *node;
    const desc_param *param;
    uint32_t i, type;

    a->desc = desc;
    a->name = object_get_typename(OBJECT(a));
    dprint(a, 1, "%s: cad %d\n", __func__, a->hda.cad);

    AUD_register_card("hda", &a->card);
    for (i = 0; i < a->desc->nnodes; i++) {
        node = a->desc->nodes + i;
        param = hda_codec_find_param(node, AC_PAR_AUDIO_WIDGET_CAP);
        if (param == NULL) {
            continue;
        }
        type = (param->val & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
        switch (type) {
        case AC_WID_AUD_OUT:
        case AC_WID_AUD_IN:
            assert(node->stindex < ARRAY_SIZE(a->st));
            st = a->st + node->stindex;
            st->state = a;
            st->node = node;
            if (type == AC_WID_AUD_OUT) {
                /* unmute output by default */
                st->gain_left = QEMU_HDA_AMP_STEPS;
                st->gain_right = QEMU_HDA_AMP_STEPS;
                st->compat_bpos = sizeof(st->compat_buf);
                st->output = true;
            } else {
                st->output = false;
            }
            st->format = AC_FMT_TYPE_PCM | AC_FMT_BITS_16 |
                (1 << AC_FMT_CHAN_SHIFT);
            hda_codec_parse_fmt(st->format, &st->as);
            hda_audio_setup(st);
            break;
        }
    }
    return 0;
}

static void hda_audio_exit(HDACodecDevice *hda)
{
    HDAAudioState *a = HDA_AUDIO(hda);
    HDAAudioStream *st;
    int i;

    dprint(a, 1, "%s\n", __func__);
    for (i = 0; i < ARRAY_SIZE(a->st); i++) {
        st = a->st + i;
        if (st->node == NULL) {
            continue;
        }
        if (a->use_timer) {
            timer_del(st->buft);
        }
        if (st->output) {
            AUD_close_out(&a->card, st->voice.out);
        } else {
            AUD_close_in(&a->card, st->voice.in);
        }
    }
    AUD_remove_card(&a->card);
}

static int hda_audio_post_load(void *opaque, int version)
{
    HDAAudioState *a = opaque;
    HDAAudioStream *st;
    int i;

    dprint(a, 1, "%s\n", __func__);
    if (version == 1) {
        /* assume running_compat[] is for output streams */
        for (i = 0; i < ARRAY_SIZE(a->running_compat); i++)
            a->running_real[16 + i] = a->running_compat[i];
    }

    for (i = 0; i < ARRAY_SIZE(a->st); i++) {
        st = a->st + i;
        if (st->node == NULL)
            continue;
        hda_codec_parse_fmt(st->format, &st->as);
        hda_audio_setup(st);
        hda_audio_set_amp(st);
        hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
    }
    return 0;
}

static void hda_audio_reset(DeviceState *dev)
{
    HDAAudioState *a = HDA_AUDIO(dev);
    HDAAudioStream *st;
    int i;

    dprint(a, 1, "%s\n", __func__);
    for (i = 0; i < ARRAY_SIZE(a->st); i++) {
        st = a->st + i;
        if (st->node != NULL) {
            hda_audio_set_running(st, false);
        }
    }
}

static bool vmstate_hda_audio_stream_buf_needed(void *opaque)
{
    HDAAudioStream *st = opaque;
    return st->state && st->state->use_timer;
}

static const VMStateDescription vmstate_hda_audio_stream_buf = {
    .name = "hda-audio-stream/buffer",
    .version_id = 1,
    .needed = vmstate_hda_audio_stream_buf_needed,
    .fields = (VMStateField[]) {
        VMSTATE_BUFFER(buf, HDAAudioStream),
        VMSTATE_INT64(rpos, HDAAudioStream),
        VMSTATE_INT64(wpos, HDAAudioStream),
        VMSTATE_TIMER_PTR(buft, HDAAudioStream),
        VMSTATE_INT64(buft_start, HDAAudioStream),
        VMSTATE_END_OF_LIST()
    }
};

static const VMStateDescription vmstate_hda_audio_stream = {
    .name = "hda-audio-stream",
    .version_id = 1,
    .fields = (VMStateField[]) {
        VMSTATE_UINT32(stream, HDAAudioStream),
        VMSTATE_UINT32(channel, HDAAudioStream),
        VMSTATE_UINT32(format, HDAAudioStream),
        VMSTATE_UINT32(gain_left, HDAAudioStream),
        VMSTATE_UINT32(gain_right, HDAAudioStream),
        VMSTATE_BOOL(mute_left, HDAAudioStream),
        VMSTATE_BOOL(mute_right, HDAAudioStream),
        VMSTATE_UINT32(compat_bpos, HDAAudioStream),
        VMSTATE_BUFFER(compat_buf, HDAAudioStream),
        VMSTATE_END_OF_LIST()
    },
    .subsections = (const VMStateDescription * []) {
        &vmstate_hda_audio_stream_buf,
        NULL
    }
};

static const VMStateDescription vmstate_hda_audio = {
    .name = "hda-audio",
    .version_id = 2,
    .post_load = hda_audio_post_load,
    .fields = (VMStateField[]) {
        VMSTATE_STRUCT_ARRAY(st, HDAAudioState, 4, 0,
                             vmstate_hda_audio_stream,
                             HDAAudioStream),
        VMSTATE_BOOL_ARRAY(running_compat, HDAAudioState, 16),
        VMSTATE_BOOL_ARRAY_V(running_real, HDAAudioState, 2 * 16, 2),
        VMSTATE_END_OF_LIST()
    }
};

static Property hda_audio_properties[] = {
    DEFINE_PROP_UINT32("debug", HDAAudioState, debug,   0),
    DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer,  true),
    DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer,  true),
    DEFINE_PROP_END_OF_LIST(),
};

static int hda_audio_init_output(HDACodecDevice *hda)
{
    HDAAudioState *a = HDA_AUDIO(hda);

    if (!a->mixer) {
        return hda_audio_init(hda, &output_nomixemu);
    } else {
        return hda_audio_init(hda, &output_mixemu);
    }
}

static int hda_audio_init_duplex(HDACodecDevice *hda)
{
    HDAAudioState *a = HDA_AUDIO(hda);

    if (!a->mixer) {
        return hda_audio_init(hda, &duplex_nomixemu);
    } else {
        return hda_audio_init(hda, &duplex_mixemu);
    }
}

static int hda_audio_init_micro(HDACodecDevice *hda)
{
    HDAAudioState *a = HDA_AUDIO(hda);

    if (!a->mixer) {
        return hda_audio_init(hda, &micro_nomixemu);
    } else {
        return hda_audio_init(hda, &micro_mixemu);
    }
}

static void hda_audio_base_class_init(ObjectClass *klass, void *data)
{
    DeviceClass *dc = DEVICE_CLASS(klass);
    HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);

    k->exit = hda_audio_exit;
    k->command = hda_audio_command;
    k->stream = hda_audio_stream;
    set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
    dc->reset = hda_audio_reset;
    dc->vmsd = &vmstate_hda_audio;
    dc->props = hda_audio_properties;
}

static const TypeInfo hda_audio_info = {
    .name          = TYPE_HDA_AUDIO,
    .parent        = TYPE_HDA_CODEC_DEVICE,
    .class_init    = hda_audio_base_class_init,
    .abstract      = true,
};

static void hda_audio_output_class_init(ObjectClass *klass, void *data)
{
    DeviceClass *dc = DEVICE_CLASS(klass);
    HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);

    k->init = hda_audio_init_output;
    dc->desc = "HDA Audio Codec, output-only (line-out)";
}

static const TypeInfo hda_audio_output_info = {
    .name          = "hda-output",
    .parent        = TYPE_HDA_AUDIO,
    .instance_size = sizeof(HDAAudioState),
    .class_init    = hda_audio_output_class_init,
};

static void hda_audio_duplex_class_init(ObjectClass *klass, void *data)
{
    DeviceClass *dc = DEVICE_CLASS(klass);
    HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);

    k->init = hda_audio_init_duplex;
    dc->desc = "HDA Audio Codec, duplex (line-out, line-in)";
}

static const TypeInfo hda_audio_duplex_info = {
    .name          = "hda-duplex",
    .parent        = TYPE_HDA_AUDIO,
    .instance_size = sizeof(HDAAudioState),
    .class_init    = hda_audio_duplex_class_init,
};

static void hda_audio_micro_class_init(ObjectClass *klass, void *data)
{
    DeviceClass *dc = DEVICE_CLASS(klass);
    HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);

    k->init = hda_audio_init_micro;
    dc->desc = "HDA Audio Codec, duplex (speaker, microphone)";
}

static const TypeInfo hda_audio_micro_info = {
    .name          = "hda-micro",
    .parent        = TYPE_HDA_AUDIO,
    .instance_size = sizeof(HDAAudioState),
    .class_init    = hda_audio_micro_class_init,
};

static void hda_audio_register_types(void)
{
    type_register_static(&hda_audio_info);
    type_register_static(&hda_audio_output_info);
    type_register_static(&hda_audio_duplex_info);
    type_register_static(&hda_audio_micro_info);
}

type_init(hda_audio_register_types)