xref: /qemu/hw/audio/hda-codec.c (revision 14f5a7ba)
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)
967