xref: /qemu/hw/audio/hda-codec.c (revision 8110fa1d)
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 typedef struct HDAAudioState HDAAudioState;
149 typedef struct HDAAudioStream HDAAudioStream;
150 
151 struct HDAAudioStream {
152     HDAAudioState *state;
153     const desc_node *node;
154     bool output, running;
155     uint32_t stream;
156     uint32_t channel;
157     uint32_t format;
158     uint32_t gain_left, gain_right;
159     bool mute_left, mute_right;
160     struct audsettings as;
161     union {
162         SWVoiceIn *in;
163         SWVoiceOut *out;
164     } voice;
165     uint8_t compat_buf[HDA_BUFFER_SIZE];
166     uint32_t compat_bpos;
167     uint8_t buf[8192]; /* size must be power of two */
168     int64_t rpos;
169     int64_t wpos;
170     QEMUTimer *buft;
171     int64_t buft_start;
172 };
173 
174 #define TYPE_HDA_AUDIO "hda-audio"
175 DECLARE_INSTANCE_CHECKER(HDAAudioState, HDA_AUDIO,
176                          TYPE_HDA_AUDIO)
177 
178 struct HDAAudioState {
179     HDACodecDevice hda;
180     const char *name;
181 
182     QEMUSoundCard card;
183     const desc_codec *desc;
184     HDAAudioStream st[4];
185     bool running_compat[16];
186     bool running_real[2 * 16];
187 
188     /* properties */
189     uint32_t debug;
190     bool     mixer;
191     bool     use_timer;
192 };
193 
194 static inline int64_t hda_bytes_per_second(HDAAudioStream *st)
195 {
196     return 2LL * st->as.nchannels * st->as.freq;
197 }
198 
199 static inline void hda_timer_sync_adjust(HDAAudioStream *st, int64_t target_pos)
200 {
201     int64_t limit = B_SIZE / 8;
202     int64_t corr = 0;
203 
204     if (target_pos > limit) {
205         corr = HDA_TIMER_TICKS;
206     }
207     if (target_pos < -limit) {
208         corr = -HDA_TIMER_TICKS;
209     }
210     if (target_pos < -(2 * limit)) {
211         corr = -(4 * HDA_TIMER_TICKS);
212     }
213     if (corr == 0) {
214         return;
215     }
216 
217     trace_hda_audio_adjust(st->node->name, target_pos);
218     st->buft_start += corr;
219 }
220 
221 static void hda_audio_input_timer(void *opaque)
222 {
223     HDAAudioStream *st = opaque;
224 
225     int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
226 
227     int64_t buft_start = st->buft_start;
228     int64_t wpos = st->wpos;
229     int64_t rpos = st->rpos;
230 
231     int64_t wanted_rpos = hda_bytes_per_second(st) * (now - buft_start)
232                           / NANOSECONDS_PER_SECOND;
233     wanted_rpos &= -4; /* IMPORTANT! clip to frames */
234 
235     if (wanted_rpos <= rpos) {
236         /* we already transmitted the data */
237         goto out_timer;
238     }
239 
240     int64_t to_transfer = MIN(wpos - rpos, wanted_rpos - rpos);
241     while (to_transfer) {
242         uint32_t start = (rpos & B_MASK);
243         uint32_t chunk = MIN(B_SIZE - start, to_transfer);
244         int rc = hda_codec_xfer(
245                 &st->state->hda, st->stream, false, st->buf + start, chunk);
246         if (!rc) {
247             break;
248         }
249         rpos += chunk;
250         to_transfer -= chunk;
251         st->rpos += chunk;
252     }
253 
254 out_timer:
255 
256     if (st->running) {
257         timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
258     }
259 }
260 
261 static void hda_audio_input_cb(void *opaque, int avail)
262 {
263     HDAAudioStream *st = opaque;
264 
265     int64_t wpos = st->wpos;
266     int64_t rpos = st->rpos;
267 
268     int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail);
269 
270     while (to_transfer) {
271         uint32_t start = (uint32_t) (wpos & B_MASK);
272         uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
273         uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
274         wpos += read;
275         to_transfer -= read;
276         st->wpos += read;
277         if (chunk != read) {
278             break;
279         }
280     }
281 
282     hda_timer_sync_adjust(st, -((wpos - rpos) - (B_SIZE >> 1)));
283 }
284 
285 static void hda_audio_output_timer(void *opaque)
286 {
287     HDAAudioStream *st = opaque;
288 
289     int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
290 
291     int64_t buft_start = st->buft_start;
292     int64_t wpos = st->wpos;
293     int64_t rpos = st->rpos;
294 
295     int64_t wanted_wpos = hda_bytes_per_second(st) * (now - buft_start)
296                           / NANOSECONDS_PER_SECOND;
297     wanted_wpos &= -4; /* IMPORTANT! clip to frames */
298 
299     if (wanted_wpos <= wpos) {
300         /* we already received the data */
301         goto out_timer;
302     }
303 
304     int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
305     while (to_transfer) {
306         uint32_t start = (wpos & B_MASK);
307         uint32_t chunk = MIN(B_SIZE - start, to_transfer);
308         int rc = hda_codec_xfer(
309                 &st->state->hda, st->stream, true, st->buf + start, chunk);
310         if (!rc) {
311             break;
312         }
313         wpos += chunk;
314         to_transfer -= chunk;
315         st->wpos += chunk;
316     }
317 
318 out_timer:
319 
320     if (st->running) {
321         timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
322     }
323 }
324 
325 static void hda_audio_output_cb(void *opaque, int avail)
326 {
327     HDAAudioStream *st = opaque;
328 
329     int64_t wpos = st->wpos;
330     int64_t rpos = st->rpos;
331 
332     int64_t to_transfer = MIN(wpos - rpos, avail);
333 
334     if (wpos - rpos == B_SIZE) {
335         /* drop buffer, reset timer adjust */
336         st->rpos = 0;
337         st->wpos = 0;
338         st->buft_start = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
339         trace_hda_audio_overrun(st->node->name);
340         return;
341     }
342 
343     while (to_transfer) {
344         uint32_t start = (uint32_t) (rpos & B_MASK);
345         uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
346         uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
347         rpos += written;
348         to_transfer -= written;
349         st->rpos += written;
350         if (chunk != written) {
351             break;
352         }
353     }
354 
355     hda_timer_sync_adjust(st, (wpos - rpos) - (B_SIZE >> 1));
356 }
357 
358 static void hda_audio_compat_input_cb(void *opaque, int avail)
359 {
360     HDAAudioStream *st = opaque;
361     int recv = 0;
362     int len;
363     bool rc;
364 
365     while (avail - recv >= sizeof(st->compat_buf)) {
366         if (st->compat_bpos != sizeof(st->compat_buf)) {
367             len = AUD_read(st->voice.in, st->compat_buf + st->compat_bpos,
368                            sizeof(st->compat_buf) - st->compat_bpos);
369             st->compat_bpos += len;
370             recv += len;
371             if (st->compat_bpos != sizeof(st->compat_buf)) {
372                 break;
373             }
374         }
375         rc = hda_codec_xfer(&st->state->hda, st->stream, false,
376                             st->compat_buf, sizeof(st->compat_buf));
377         if (!rc) {
378             break;
379         }
380         st->compat_bpos = 0;
381     }
382 }
383 
384 static void hda_audio_compat_output_cb(void *opaque, int avail)
385 {
386     HDAAudioStream *st = opaque;
387     int sent = 0;
388     int len;
389     bool rc;
390 
391     while (avail - sent >= sizeof(st->compat_buf)) {
392         if (st->compat_bpos == sizeof(st->compat_buf)) {
393             rc = hda_codec_xfer(&st->state->hda, st->stream, true,
394                                 st->compat_buf, sizeof(st->compat_buf));
395             if (!rc) {
396                 break;
397             }
398             st->compat_bpos = 0;
399         }
400         len = AUD_write(st->voice.out, st->compat_buf + st->compat_bpos,
401                         sizeof(st->compat_buf) - st->compat_bpos);
402         st->compat_bpos += len;
403         sent += len;
404         if (st->compat_bpos != sizeof(st->compat_buf)) {
405             break;
406         }
407     }
408 }
409 
410 static void hda_audio_set_running(HDAAudioStream *st, bool running)
411 {
412     if (st->node == NULL) {
413         return;
414     }
415     if (st->running == running) {
416         return;
417     }
418     st->running = running;
419     trace_hda_audio_running(st->node->name, st->stream, st->running);
420     if (st->state->use_timer) {
421         if (running) {
422             int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
423             st->rpos = 0;
424             st->wpos = 0;
425             st->buft_start = now;
426             timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
427         } else {
428             timer_del(st->buft);
429         }
430     }
431     if (st->output) {
432         AUD_set_active_out(st->voice.out, st->running);
433     } else {
434         AUD_set_active_in(st->voice.in, st->running);
435     }
436 }
437 
438 static void hda_audio_set_amp(HDAAudioStream *st)
439 {
440     bool muted;
441     uint32_t left, right;
442 
443     if (st->node == NULL) {
444         return;
445     }
446 
447     muted = st->mute_left && st->mute_right;
448     left  = st->mute_left  ? 0 : st->gain_left;
449     right = st->mute_right ? 0 : st->gain_right;
450 
451     left = left * 255 / QEMU_HDA_AMP_STEPS;
452     right = right * 255 / QEMU_HDA_AMP_STEPS;
453 
454     if (!st->state->mixer) {
455         return;
456     }
457     if (st->output) {
458         AUD_set_volume_out(st->voice.out, muted, left, right);
459     } else {
460         AUD_set_volume_in(st->voice.in, muted, left, right);
461     }
462 }
463 
464 static void hda_audio_setup(HDAAudioStream *st)
465 {
466     bool use_timer = st->state->use_timer;
467     audio_callback_fn cb;
468 
469     if (st->node == NULL) {
470         return;
471     }
472 
473     trace_hda_audio_format(st->node->name, st->as.nchannels,
474                            fmt2name[st->as.fmt], st->as.freq);
475 
476     if (st->output) {
477         if (use_timer) {
478             cb = hda_audio_output_cb;
479             st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
480                                     hda_audio_output_timer, st);
481         } else {
482             cb = hda_audio_compat_output_cb;
483         }
484         st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
485                                      st->node->name, st, cb, &st->as);
486     } else {
487         if (use_timer) {
488             cb = hda_audio_input_cb;
489             st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
490                                     hda_audio_input_timer, st);
491         } else {
492             cb = hda_audio_compat_input_cb;
493         }
494         st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
495                                    st->node->name, st, cb, &st->as);
496     }
497 }
498 
499 static void hda_audio_command(HDACodecDevice *hda, uint32_t nid, uint32_t data)
500 {
501     HDAAudioState *a = HDA_AUDIO(hda);
502     HDAAudioStream *st;
503     const desc_node *node = NULL;
504     const desc_param *param;
505     uint32_t verb, payload, response, count, shift;
506 
507     if ((data & 0x70000) == 0x70000) {
508         /* 12/8 id/payload */
509         verb = (data >> 8) & 0xfff;
510         payload = data & 0x00ff;
511     } else {
512         /* 4/16 id/payload */
513         verb = (data >> 8) & 0xf00;
514         payload = data & 0xffff;
515     }
516 
517     node = hda_codec_find_node(a->desc, nid);
518     if (node == NULL) {
519         goto fail;
520     }
521     dprint(a, 2, "%s: nid %d (%s), verb 0x%x, payload 0x%x\n",
522            __func__, nid, node->name, verb, payload);
523 
524     switch (verb) {
525     /* all nodes */
526     case AC_VERB_PARAMETERS:
527         param = hda_codec_find_param(node, payload);
528         if (param == NULL) {
529             goto fail;
530         }
531         hda_codec_response(hda, true, param->val);
532         break;
533     case AC_VERB_GET_SUBSYSTEM_ID:
534         hda_codec_response(hda, true, a->desc->iid);
535         break;
536 
537     /* all functions */
538     case AC_VERB_GET_CONNECT_LIST:
539         param = hda_codec_find_param(node, AC_PAR_CONNLIST_LEN);
540         count = param ? param->val : 0;
541         response = 0;
542         shift = 0;
543         while (payload < count && shift < 32) {
544             response |= node->conn[payload] << shift;
545             payload++;
546             shift += 8;
547         }
548         hda_codec_response(hda, true, response);
549         break;
550 
551     /* pin widget */
552     case AC_VERB_GET_CONFIG_DEFAULT:
553         hda_codec_response(hda, true, node->config);
554         break;
555     case AC_VERB_GET_PIN_WIDGET_CONTROL:
556         hda_codec_response(hda, true, node->pinctl);
557         break;
558     case AC_VERB_SET_PIN_WIDGET_CONTROL:
559         if (node->pinctl != payload) {
560             dprint(a, 1, "unhandled pin control bit\n");
561         }
562         hda_codec_response(hda, true, 0);
563         break;
564 
565     /* audio in/out widget */
566     case AC_VERB_SET_CHANNEL_STREAMID:
567         st = a->st + node->stindex;
568         if (st->node == NULL) {
569             goto fail;
570         }
571         hda_audio_set_running(st, false);
572         st->stream = (payload >> 4) & 0x0f;
573         st->channel = payload & 0x0f;
574         dprint(a, 2, "%s: stream %d, channel %d\n",
575                st->node->name, st->stream, st->channel);
576         hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
577         hda_codec_response(hda, true, 0);
578         break;
579     case AC_VERB_GET_CONV:
580         st = a->st + node->stindex;
581         if (st->node == NULL) {
582             goto fail;
583         }
584         response = st->stream << 4 | st->channel;
585         hda_codec_response(hda, true, response);
586         break;
587     case AC_VERB_SET_STREAM_FORMAT:
588         st = a->st + node->stindex;
589         if (st->node == NULL) {
590             goto fail;
591         }
592         st->format = payload;
593         hda_codec_parse_fmt(st->format, &st->as);
594         hda_audio_setup(st);
595         hda_codec_response(hda, true, 0);
596         break;
597     case AC_VERB_GET_STREAM_FORMAT:
598         st = a->st + node->stindex;
599         if (st->node == NULL) {
600             goto fail;
601         }
602         hda_codec_response(hda, true, st->format);
603         break;
604     case AC_VERB_GET_AMP_GAIN_MUTE:
605         st = a->st + node->stindex;
606         if (st->node == NULL) {
607             goto fail;
608         }
609         if (payload & AC_AMP_GET_LEFT) {
610             response = st->gain_left | (st->mute_left ? AC_AMP_MUTE : 0);
611         } else {
612             response = st->gain_right | (st->mute_right ? AC_AMP_MUTE : 0);
613         }
614         hda_codec_response(hda, true, response);
615         break;
616     case AC_VERB_SET_AMP_GAIN_MUTE:
617         st = a->st + node->stindex;
618         if (st->node == NULL) {
619             goto fail;
620         }
621         dprint(a, 1, "amp (%s): %s%s%s%s index %d  gain %3d %s\n",
622                st->node->name,
623                (payload & AC_AMP_SET_OUTPUT) ? "o" : "-",
624                (payload & AC_AMP_SET_INPUT)  ? "i" : "-",
625                (payload & AC_AMP_SET_LEFT)   ? "l" : "-",
626                (payload & AC_AMP_SET_RIGHT)  ? "r" : "-",
627                (payload & AC_AMP_SET_INDEX) >> AC_AMP_SET_INDEX_SHIFT,
628                (payload & AC_AMP_GAIN),
629                (payload & AC_AMP_MUTE) ? "muted" : "");
630         if (payload & AC_AMP_SET_LEFT) {
631             st->gain_left = payload & AC_AMP_GAIN;
632             st->mute_left = payload & AC_AMP_MUTE;
633         }
634         if (payload & AC_AMP_SET_RIGHT) {
635             st->gain_right = payload & AC_AMP_GAIN;
636             st->mute_right = payload & AC_AMP_MUTE;
637         }
638         hda_audio_set_amp(st);
639         hda_codec_response(hda, true, 0);
640         break;
641 
642     /* not supported */
643     case AC_VERB_SET_POWER_STATE:
644     case AC_VERB_GET_POWER_STATE:
645     case AC_VERB_GET_SDI_SELECT:
646         hda_codec_response(hda, true, 0);
647         break;
648     default:
649         goto fail;
650     }
651     return;
652 
653 fail:
654     dprint(a, 1, "%s: not handled: nid %d (%s), verb 0x%x, payload 0x%x\n",
655            __func__, nid, node ? node->name : "?", verb, payload);
656     hda_codec_response(hda, true, 0);
657 }
658 
659 static void hda_audio_stream(HDACodecDevice *hda, uint32_t stnr, bool running, bool output)
660 {
661     HDAAudioState *a = HDA_AUDIO(hda);
662     int s;
663 
664     a->running_compat[stnr] = running;
665     a->running_real[output * 16 + stnr] = running;
666     for (s = 0; s < ARRAY_SIZE(a->st); s++) {
667         if (a->st[s].node == NULL) {
668             continue;
669         }
670         if (a->st[s].output != output) {
671             continue;
672         }
673         if (a->st[s].stream != stnr) {
674             continue;
675         }
676         hda_audio_set_running(&a->st[s], running);
677     }
678 }
679 
680 static int hda_audio_init(HDACodecDevice *hda, const struct desc_codec *desc)
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     a->desc = desc;
689     a->name = object_get_typename(OBJECT(a));
690     dprint(a, 1, "%s: cad %d\n", __func__, a->hda.cad);
691 
692     AUD_register_card("hda", &a->card);
693     for (i = 0; i < a->desc->nnodes; i++) {
694         node = a->desc->nodes + i;
695         param = hda_codec_find_param(node, AC_PAR_AUDIO_WIDGET_CAP);
696         if (param == NULL) {
697             continue;
698         }
699         type = (param->val & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
700         switch (type) {
701         case AC_WID_AUD_OUT:
702         case AC_WID_AUD_IN:
703             assert(node->stindex < ARRAY_SIZE(a->st));
704             st = a->st + node->stindex;
705             st->state = a;
706             st->node = node;
707             if (type == AC_WID_AUD_OUT) {
708                 /* unmute output by default */
709                 st->gain_left = QEMU_HDA_AMP_STEPS;
710                 st->gain_right = QEMU_HDA_AMP_STEPS;
711                 st->compat_bpos = sizeof(st->compat_buf);
712                 st->output = true;
713             } else {
714                 st->output = false;
715             }
716             st->format = AC_FMT_TYPE_PCM | AC_FMT_BITS_16 |
717                 (1 << AC_FMT_CHAN_SHIFT);
718             hda_codec_parse_fmt(st->format, &st->as);
719             hda_audio_setup(st);
720             break;
721         }
722     }
723     return 0;
724 }
725 
726 static void hda_audio_exit(HDACodecDevice *hda)
727 {
728     HDAAudioState *a = HDA_AUDIO(hda);
729     HDAAudioStream *st;
730     int i;
731 
732     dprint(a, 1, "%s\n", __func__);
733     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
734         st = a->st + i;
735         if (st->node == NULL) {
736             continue;
737         }
738         if (a->use_timer) {
739             timer_del(st->buft);
740         }
741         if (st->output) {
742             AUD_close_out(&a->card, st->voice.out);
743         } else {
744             AUD_close_in(&a->card, st->voice.in);
745         }
746     }
747     AUD_remove_card(&a->card);
748 }
749 
750 static int hda_audio_post_load(void *opaque, int version)
751 {
752     HDAAudioState *a = opaque;
753     HDAAudioStream *st;
754     int i;
755 
756     dprint(a, 1, "%s\n", __func__);
757     if (version == 1) {
758         /* assume running_compat[] is for output streams */
759         for (i = 0; i < ARRAY_SIZE(a->running_compat); i++)
760             a->running_real[16 + i] = a->running_compat[i];
761     }
762 
763     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
764         st = a->st + i;
765         if (st->node == NULL)
766             continue;
767         hda_codec_parse_fmt(st->format, &st->as);
768         hda_audio_setup(st);
769         hda_audio_set_amp(st);
770         hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
771     }
772     return 0;
773 }
774 
775 static void hda_audio_reset(DeviceState *dev)
776 {
777     HDAAudioState *a = HDA_AUDIO(dev);
778     HDAAudioStream *st;
779     int i;
780 
781     dprint(a, 1, "%s\n", __func__);
782     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
783         st = a->st + i;
784         if (st->node != NULL) {
785             hda_audio_set_running(st, false);
786         }
787     }
788 }
789 
790 static bool vmstate_hda_audio_stream_buf_needed(void *opaque)
791 {
792     HDAAudioStream *st = opaque;
793     return st->state && st->state->use_timer;
794 }
795 
796 static const VMStateDescription vmstate_hda_audio_stream_buf = {
797     .name = "hda-audio-stream/buffer",
798     .version_id = 1,
799     .needed = vmstate_hda_audio_stream_buf_needed,
800     .fields = (VMStateField[]) {
801         VMSTATE_BUFFER(buf, HDAAudioStream),
802         VMSTATE_INT64(rpos, HDAAudioStream),
803         VMSTATE_INT64(wpos, HDAAudioStream),
804         VMSTATE_TIMER_PTR(buft, HDAAudioStream),
805         VMSTATE_INT64(buft_start, HDAAudioStream),
806         VMSTATE_END_OF_LIST()
807     }
808 };
809 
810 static const VMStateDescription vmstate_hda_audio_stream = {
811     .name = "hda-audio-stream",
812     .version_id = 1,
813     .fields = (VMStateField[]) {
814         VMSTATE_UINT32(stream, HDAAudioStream),
815         VMSTATE_UINT32(channel, HDAAudioStream),
816         VMSTATE_UINT32(format, HDAAudioStream),
817         VMSTATE_UINT32(gain_left, HDAAudioStream),
818         VMSTATE_UINT32(gain_right, HDAAudioStream),
819         VMSTATE_BOOL(mute_left, HDAAudioStream),
820         VMSTATE_BOOL(mute_right, HDAAudioStream),
821         VMSTATE_UINT32(compat_bpos, HDAAudioStream),
822         VMSTATE_BUFFER(compat_buf, HDAAudioStream),
823         VMSTATE_END_OF_LIST()
824     },
825     .subsections = (const VMStateDescription * []) {
826         &vmstate_hda_audio_stream_buf,
827         NULL
828     }
829 };
830 
831 static const VMStateDescription vmstate_hda_audio = {
832     .name = "hda-audio",
833     .version_id = 2,
834     .post_load = hda_audio_post_load,
835     .fields = (VMStateField[]) {
836         VMSTATE_STRUCT_ARRAY(st, HDAAudioState, 4, 0,
837                              vmstate_hda_audio_stream,
838                              HDAAudioStream),
839         VMSTATE_BOOL_ARRAY(running_compat, HDAAudioState, 16),
840         VMSTATE_BOOL_ARRAY_V(running_real, HDAAudioState, 2 * 16, 2),
841         VMSTATE_END_OF_LIST()
842     }
843 };
844 
845 static Property hda_audio_properties[] = {
846     DEFINE_AUDIO_PROPERTIES(HDAAudioState, card),
847     DEFINE_PROP_UINT32("debug", HDAAudioState, debug,   0),
848     DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer,  true),
849     DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer,  true),
850     DEFINE_PROP_END_OF_LIST(),
851 };
852 
853 static int hda_audio_init_output(HDACodecDevice *hda)
854 {
855     HDAAudioState *a = HDA_AUDIO(hda);
856 
857     if (!a->mixer) {
858         return hda_audio_init(hda, &output_nomixemu);
859     } else {
860         return hda_audio_init(hda, &output_mixemu);
861     }
862 }
863 
864 static int hda_audio_init_duplex(HDACodecDevice *hda)
865 {
866     HDAAudioState *a = HDA_AUDIO(hda);
867 
868     if (!a->mixer) {
869         return hda_audio_init(hda, &duplex_nomixemu);
870     } else {
871         return hda_audio_init(hda, &duplex_mixemu);
872     }
873 }
874 
875 static int hda_audio_init_micro(HDACodecDevice *hda)
876 {
877     HDAAudioState *a = HDA_AUDIO(hda);
878 
879     if (!a->mixer) {
880         return hda_audio_init(hda, &micro_nomixemu);
881     } else {
882         return hda_audio_init(hda, &micro_mixemu);
883     }
884 }
885 
886 static void hda_audio_base_class_init(ObjectClass *klass, void *data)
887 {
888     DeviceClass *dc = DEVICE_CLASS(klass);
889     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
890 
891     k->exit = hda_audio_exit;
892     k->command = hda_audio_command;
893     k->stream = hda_audio_stream;
894     set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
895     dc->reset = hda_audio_reset;
896     dc->vmsd = &vmstate_hda_audio;
897     device_class_set_props(dc, hda_audio_properties);
898 }
899 
900 static const TypeInfo hda_audio_info = {
901     .name          = TYPE_HDA_AUDIO,
902     .parent        = TYPE_HDA_CODEC_DEVICE,
903     .instance_size = sizeof(HDAAudioState),
904     .class_init    = hda_audio_base_class_init,
905     .abstract      = true,
906 };
907 
908 static void hda_audio_output_class_init(ObjectClass *klass, void *data)
909 {
910     DeviceClass *dc = DEVICE_CLASS(klass);
911     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
912 
913     k->init = hda_audio_init_output;
914     dc->desc = "HDA Audio Codec, output-only (line-out)";
915 }
916 
917 static const TypeInfo hda_audio_output_info = {
918     .name          = "hda-output",
919     .parent        = TYPE_HDA_AUDIO,
920     .class_init    = hda_audio_output_class_init,
921 };
922 
923 static void hda_audio_duplex_class_init(ObjectClass *klass, void *data)
924 {
925     DeviceClass *dc = DEVICE_CLASS(klass);
926     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
927 
928     k->init = hda_audio_init_duplex;
929     dc->desc = "HDA Audio Codec, duplex (line-out, line-in)";
930 }
931 
932 static const TypeInfo hda_audio_duplex_info = {
933     .name          = "hda-duplex",
934     .parent        = TYPE_HDA_AUDIO,
935     .class_init    = hda_audio_duplex_class_init,
936 };
937 
938 static void hda_audio_micro_class_init(ObjectClass *klass, void *data)
939 {
940     DeviceClass *dc = DEVICE_CLASS(klass);
941     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
942 
943     k->init = hda_audio_init_micro;
944     dc->desc = "HDA Audio Codec, duplex (speaker, microphone)";
945 }
946 
947 static const TypeInfo hda_audio_micro_info = {
948     .name          = "hda-micro",
949     .parent        = TYPE_HDA_AUDIO,
950     .class_init    = hda_audio_micro_class_init,
951 };
952 
953 static void hda_audio_register_types(void)
954 {
955     type_register_static(&hda_audio_info);
956     type_register_static(&hda_audio_output_info);
957     type_register_static(&hda_audio_duplex_info);
958     type_register_static(&hda_audio_micro_info);
959 }
960 
961 type_init(hda_audio_register_types)
962