xref: /openbmc/qemu/hw/audio/hda-codec.c (revision ca693d1c)
1 /*
2  * Copyright (C) 2010 Red Hat, Inc.
3  *
4  * written by Gerd Hoffmann <kraxel@redhat.com>
5  *
6  * This program is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU General Public License as
8  * published by the Free Software Foundation; either version 2 or
9  * (at your option) version 3 of the License.
10  *
11  * This program is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
14  * GNU General Public License for more details.
15  *
16  * You should have received a copy of the GNU General Public License
17  * along with this program; if not, see <http://www.gnu.org/licenses/>.
18  */
19 
20 #include "qemu/osdep.h"
21 #include "hw/hw.h"
22 #include "hw/pci/pci.h"
23 #include "intel-hda.h"
24 #include "intel-hda-defs.h"
25 #include "audio/audio.h"
26 #include "trace.h"
27 
28 /* -------------------------------------------------------------------------- */
29 
30 typedef struct desc_param {
31     uint32_t id;
32     uint32_t val;
33 } desc_param;
34 
35 typedef struct desc_node {
36     uint32_t nid;
37     const char *name;
38     const desc_param *params;
39     uint32_t nparams;
40     uint32_t config;
41     uint32_t pinctl;
42     uint32_t *conn;
43     uint32_t stindex;
44 } desc_node;
45 
46 typedef struct desc_codec {
47     const char *name;
48     uint32_t iid;
49     const desc_node *nodes;
50     uint32_t nnodes;
51 } desc_codec;
52 
53 static const desc_param* hda_codec_find_param(const desc_node *node, uint32_t id)
54 {
55     int i;
56 
57     for (i = 0; i < node->nparams; i++) {
58         if (node->params[i].id == id) {
59             return &node->params[i];
60         }
61     }
62     return NULL;
63 }
64 
65 static const desc_node* hda_codec_find_node(const desc_codec *codec, uint32_t nid)
66 {
67     int i;
68 
69     for (i = 0; i < codec->nnodes; i++) {
70         if (codec->nodes[i].nid == nid) {
71             return &codec->nodes[i];
72         }
73     }
74     return NULL;
75 }
76 
77 static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
78 {
79     if (format & AC_FMT_TYPE_NON_PCM) {
80         return;
81     }
82 
83     as->freq = (format & AC_FMT_BASE_44K) ? 44100 : 48000;
84 
85     switch ((format & AC_FMT_MULT_MASK) >> AC_FMT_MULT_SHIFT) {
86     case 1: as->freq *= 2; break;
87     case 2: as->freq *= 3; break;
88     case 3: as->freq *= 4; break;
89     }
90 
91     switch ((format & AC_FMT_DIV_MASK) >> AC_FMT_DIV_SHIFT) {
92     case 1: as->freq /= 2; break;
93     case 2: as->freq /= 3; break;
94     case 3: as->freq /= 4; break;
95     case 4: as->freq /= 5; break;
96     case 5: as->freq /= 6; break;
97     case 6: as->freq /= 7; break;
98     case 7: as->freq /= 8; break;
99     }
100 
101     switch (format & AC_FMT_BITS_MASK) {
102     case AC_FMT_BITS_8:  as->fmt = AUDIO_FORMAT_S8;  break;
103     case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
104     case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
105     }
106 
107     as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
108 }
109 
110 /* -------------------------------------------------------------------------- */
111 /*
112  * HDA codec descriptions
113  */
114 
115 /* some defines */
116 
117 #define QEMU_HDA_ID_VENDOR  0x1af4
118 #define QEMU_HDA_PCM_FORMATS (AC_SUPPCM_BITS_16 |       \
119                               0x1fc /* 16 -> 96 kHz */)
120 #define QEMU_HDA_AMP_NONE    (0)
121 #define QEMU_HDA_AMP_STEPS   0x4a
122 
123 #define   PARAM mixemu
124 #define   HDA_MIXER
125 #include "hda-codec-common.h"
126 
127 #define   PARAM nomixemu
128 #include  "hda-codec-common.h"
129 
130 #define HDA_TIMER_TICKS (SCALE_MS)
131 #define B_SIZE sizeof(st->buf)
132 #define B_MASK (sizeof(st->buf) - 1)
133 
134 /* -------------------------------------------------------------------------- */
135 
136 static const char *fmt2name[] = {
137     [ AUDIO_FORMAT_U8  ] = "PCM-U8",
138     [ AUDIO_FORMAT_S8  ] = "PCM-S8",
139     [ AUDIO_FORMAT_U16 ] = "PCM-U16",
140     [ AUDIO_FORMAT_S16 ] = "PCM-S16",
141     [ AUDIO_FORMAT_U32 ] = "PCM-U32",
142     [ AUDIO_FORMAT_S32 ] = "PCM-S32",
143 };
144 
145 typedef struct HDAAudioState HDAAudioState;
146 typedef struct HDAAudioStream HDAAudioStream;
147 
148 struct HDAAudioStream {
149     HDAAudioState *state;
150     const desc_node *node;
151     bool output, running;
152     uint32_t stream;
153     uint32_t channel;
154     uint32_t format;
155     uint32_t gain_left, gain_right;
156     bool mute_left, mute_right;
157     struct audsettings as;
158     union {
159         SWVoiceIn *in;
160         SWVoiceOut *out;
161     } voice;
162     uint8_t compat_buf[HDA_BUFFER_SIZE];
163     uint32_t compat_bpos;
164     uint8_t buf[8192]; /* size must be power of two */
165     int64_t rpos;
166     int64_t wpos;
167     QEMUTimer *buft;
168     int64_t buft_start;
169 };
170 
171 #define TYPE_HDA_AUDIO "hda-audio"
172 #define HDA_AUDIO(obj) OBJECT_CHECK(HDAAudioState, (obj), TYPE_HDA_AUDIO)
173 
174 struct HDAAudioState {
175     HDACodecDevice hda;
176     const char *name;
177 
178     QEMUSoundCard card;
179     const desc_codec *desc;
180     HDAAudioStream st[4];
181     bool running_compat[16];
182     bool running_real[2 * 16];
183 
184     /* properties */
185     uint32_t debug;
186     bool     mixer;
187     bool     use_timer;
188 };
189 
190 static inline int64_t hda_bytes_per_second(HDAAudioStream *st)
191 {
192     return 2LL * st->as.nchannels * st->as.freq;
193 }
194 
195 static inline void hda_timer_sync_adjust(HDAAudioStream *st, int64_t target_pos)
196 {
197     int64_t limit = B_SIZE / 8;
198     int64_t corr = 0;
199 
200     if (target_pos > limit) {
201         corr = HDA_TIMER_TICKS;
202     }
203     if (target_pos < -limit) {
204         corr = -HDA_TIMER_TICKS;
205     }
206     if (target_pos < -(2 * limit)) {
207         corr = -(4 * HDA_TIMER_TICKS);
208     }
209     if (corr == 0) {
210         return;
211     }
212 
213     trace_hda_audio_adjust(st->node->name, target_pos);
214     st->buft_start += corr;
215 }
216 
217 static void hda_audio_input_timer(void *opaque)
218 {
219     HDAAudioStream *st = opaque;
220 
221     int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
222 
223     int64_t buft_start = st->buft_start;
224     int64_t wpos = st->wpos;
225     int64_t rpos = st->rpos;
226 
227     int64_t wanted_rpos = hda_bytes_per_second(st) * (now - buft_start)
228                           / NANOSECONDS_PER_SECOND;
229     wanted_rpos &= -4; /* IMPORTANT! clip to frames */
230 
231     if (wanted_rpos <= rpos) {
232         /* we already transmitted the data */
233         goto out_timer;
234     }
235 
236     int64_t to_transfer = audio_MIN(wpos - rpos, wanted_rpos - rpos);
237     while (to_transfer) {
238         uint32_t start = (rpos & B_MASK);
239         uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
240         int rc = hda_codec_xfer(
241                 &st->state->hda, st->stream, false, st->buf + start, chunk);
242         if (!rc) {
243             break;
244         }
245         rpos += chunk;
246         to_transfer -= chunk;
247         st->rpos += chunk;
248     }
249 
250 out_timer:
251 
252     if (st->running) {
253         timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
254     }
255 }
256 
257 static void hda_audio_input_cb(void *opaque, int avail)
258 {
259     HDAAudioStream *st = opaque;
260 
261     int64_t wpos = st->wpos;
262     int64_t rpos = st->rpos;
263 
264     int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), avail);
265 
266     hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1)));
267 
268     while (to_transfer) {
269         uint32_t start = (uint32_t) (wpos & B_MASK);
270         uint32_t chunk = (uint32_t) audio_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 
281 static void hda_audio_output_timer(void *opaque)
282 {
283     HDAAudioStream *st = opaque;
284 
285     int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
286 
287     int64_t buft_start = st->buft_start;
288     int64_t wpos = st->wpos;
289     int64_t rpos = st->rpos;
290 
291     int64_t wanted_wpos = hda_bytes_per_second(st) * (now - buft_start)
292                           / NANOSECONDS_PER_SECOND;
293     wanted_wpos &= -4; /* IMPORTANT! clip to frames */
294 
295     if (wanted_wpos <= wpos) {
296         /* we already received the data */
297         goto out_timer;
298     }
299 
300     int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
301     while (to_transfer) {
302         uint32_t start = (wpos & B_MASK);
303         uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
304         int rc = hda_codec_xfer(
305                 &st->state->hda, st->stream, true, st->buf + start, chunk);
306         if (!rc) {
307             break;
308         }
309         wpos += chunk;
310         to_transfer -= chunk;
311         st->wpos += chunk;
312     }
313 
314 out_timer:
315 
316     if (st->running) {
317         timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
318     }
319 }
320 
321 static void hda_audio_output_cb(void *opaque, int avail)
322 {
323     HDAAudioStream *st = opaque;
324 
325     int64_t wpos = st->wpos;
326     int64_t rpos = st->rpos;
327 
328     int64_t to_transfer = audio_MIN(wpos - rpos, avail);
329 
330     if (wpos - rpos == B_SIZE) {
331         /* drop buffer, reset timer adjust */
332         st->rpos = 0;
333         st->wpos = 0;
334         st->buft_start = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
335         trace_hda_audio_overrun(st->node->name);
336         return;
337     }
338 
339     hda_timer_sync_adjust(st, (wpos - rpos) - to_transfer - (B_SIZE >> 1));
340 
341     while (to_transfer) {
342         uint32_t start = (uint32_t) (rpos & B_MASK);
343         uint32_t chunk = (uint32_t) audio_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 
354 static void hda_audio_compat_input_cb(void *opaque, int avail)
355 {
356     HDAAudioStream *st = opaque;
357     int recv = 0;
358     int len;
359     bool rc;
360 
361     while (avail - recv >= sizeof(st->compat_buf)) {
362         if (st->compat_bpos != sizeof(st->compat_buf)) {
363             len = AUD_read(st->voice.in, st->compat_buf + st->compat_bpos,
364                            sizeof(st->compat_buf) - st->compat_bpos);
365             st->compat_bpos += len;
366             recv += len;
367             if (st->compat_bpos != sizeof(st->compat_buf)) {
368                 break;
369             }
370         }
371         rc = hda_codec_xfer(&st->state->hda, st->stream, false,
372                             st->compat_buf, sizeof(st->compat_buf));
373         if (!rc) {
374             break;
375         }
376         st->compat_bpos = 0;
377     }
378 }
379 
380 static void hda_audio_compat_output_cb(void *opaque, int avail)
381 {
382     HDAAudioStream *st = opaque;
383     int sent = 0;
384     int len;
385     bool rc;
386 
387     while (avail - sent >= sizeof(st->compat_buf)) {
388         if (st->compat_bpos == sizeof(st->compat_buf)) {
389             rc = hda_codec_xfer(&st->state->hda, st->stream, true,
390                                 st->compat_buf, sizeof(st->compat_buf));
391             if (!rc) {
392                 break;
393             }
394             st->compat_bpos = 0;
395         }
396         len = AUD_write(st->voice.out, st->compat_buf + st->compat_bpos,
397                         sizeof(st->compat_buf) - st->compat_bpos);
398         st->compat_bpos += len;
399         sent += len;
400         if (st->compat_bpos != sizeof(st->compat_buf)) {
401             break;
402         }
403     }
404 }
405 
406 static void hda_audio_set_running(HDAAudioStream *st, bool running)
407 {
408     if (st->node == NULL) {
409         return;
410     }
411     if (st->running == running) {
412         return;
413     }
414     st->running = running;
415     trace_hda_audio_running(st->node->name, st->stream, st->running);
416     if (st->state->use_timer) {
417         if (running) {
418             int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
419             st->rpos = 0;
420             st->wpos = 0;
421             st->buft_start = now;
422             timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
423         } else {
424             timer_del(st->buft);
425         }
426     }
427     if (st->output) {
428         AUD_set_active_out(st->voice.out, st->running);
429     } else {
430         AUD_set_active_in(st->voice.in, st->running);
431     }
432 }
433 
434 static void hda_audio_set_amp(HDAAudioStream *st)
435 {
436     bool muted;
437     uint32_t left, right;
438 
439     if (st->node == NULL) {
440         return;
441     }
442 
443     muted = st->mute_left && st->mute_right;
444     left  = st->mute_left  ? 0 : st->gain_left;
445     right = st->mute_right ? 0 : st->gain_right;
446 
447     left = left * 255 / QEMU_HDA_AMP_STEPS;
448     right = right * 255 / QEMU_HDA_AMP_STEPS;
449 
450     if (!st->state->mixer) {
451         return;
452     }
453     if (st->output) {
454         AUD_set_volume_out(st->voice.out, muted, left, right);
455     } else {
456         AUD_set_volume_in(st->voice.in, muted, left, right);
457     }
458 }
459 
460 static void hda_audio_setup(HDAAudioStream *st)
461 {
462     bool use_timer = st->state->use_timer;
463     audio_callback_fn cb;
464 
465     if (st->node == NULL) {
466         return;
467     }
468 
469     trace_hda_audio_format(st->node->name, st->as.nchannels,
470                            fmt2name[st->as.fmt], st->as.freq);
471 
472     if (st->output) {
473         if (use_timer) {
474             cb = hda_audio_output_cb;
475             st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
476                                     hda_audio_output_timer, st);
477         } else {
478             cb = hda_audio_compat_output_cb;
479         }
480         st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
481                                      st->node->name, st, cb, &st->as);
482     } else {
483         if (use_timer) {
484             cb = hda_audio_input_cb;
485             st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
486                                     hda_audio_input_timer, st);
487         } else {
488             cb = hda_audio_compat_input_cb;
489         }
490         st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
491                                    st->node->name, st, cb, &st->as);
492     }
493 }
494 
495 static void hda_audio_command(HDACodecDevice *hda, uint32_t nid, uint32_t data)
496 {
497     HDAAudioState *a = HDA_AUDIO(hda);
498     HDAAudioStream *st;
499     const desc_node *node = NULL;
500     const desc_param *param;
501     uint32_t verb, payload, response, count, shift;
502 
503     if ((data & 0x70000) == 0x70000) {
504         /* 12/8 id/payload */
505         verb = (data >> 8) & 0xfff;
506         payload = data & 0x00ff;
507     } else {
508         /* 4/16 id/payload */
509         verb = (data >> 8) & 0xf00;
510         payload = data & 0xffff;
511     }
512 
513     node = hda_codec_find_node(a->desc, nid);
514     if (node == NULL) {
515         goto fail;
516     }
517     dprint(a, 2, "%s: nid %d (%s), verb 0x%x, payload 0x%x\n",
518            __func__, nid, node->name, verb, payload);
519 
520     switch (verb) {
521     /* all nodes */
522     case AC_VERB_PARAMETERS:
523         param = hda_codec_find_param(node, payload);
524         if (param == NULL) {
525             goto fail;
526         }
527         hda_codec_response(hda, true, param->val);
528         break;
529     case AC_VERB_GET_SUBSYSTEM_ID:
530         hda_codec_response(hda, true, a->desc->iid);
531         break;
532 
533     /* all functions */
534     case AC_VERB_GET_CONNECT_LIST:
535         param = hda_codec_find_param(node, AC_PAR_CONNLIST_LEN);
536         count = param ? param->val : 0;
537         response = 0;
538         shift = 0;
539         while (payload < count && shift < 32) {
540             response |= node->conn[payload] << shift;
541             payload++;
542             shift += 8;
543         }
544         hda_codec_response(hda, true, response);
545         break;
546 
547     /* pin widget */
548     case AC_VERB_GET_CONFIG_DEFAULT:
549         hda_codec_response(hda, true, node->config);
550         break;
551     case AC_VERB_GET_PIN_WIDGET_CONTROL:
552         hda_codec_response(hda, true, node->pinctl);
553         break;
554     case AC_VERB_SET_PIN_WIDGET_CONTROL:
555         if (node->pinctl != payload) {
556             dprint(a, 1, "unhandled pin control bit\n");
557         }
558         hda_codec_response(hda, true, 0);
559         break;
560 
561     /* audio in/out widget */
562     case AC_VERB_SET_CHANNEL_STREAMID:
563         st = a->st + node->stindex;
564         if (st->node == NULL) {
565             goto fail;
566         }
567         hda_audio_set_running(st, false);
568         st->stream = (payload >> 4) & 0x0f;
569         st->channel = payload & 0x0f;
570         dprint(a, 2, "%s: stream %d, channel %d\n",
571                st->node->name, st->stream, st->channel);
572         hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
573         hda_codec_response(hda, true, 0);
574         break;
575     case AC_VERB_GET_CONV:
576         st = a->st + node->stindex;
577         if (st->node == NULL) {
578             goto fail;
579         }
580         response = st->stream << 4 | st->channel;
581         hda_codec_response(hda, true, response);
582         break;
583     case AC_VERB_SET_STREAM_FORMAT:
584         st = a->st + node->stindex;
585         if (st->node == NULL) {
586             goto fail;
587         }
588         st->format = payload;
589         hda_codec_parse_fmt(st->format, &st->as);
590         hda_audio_setup(st);
591         hda_codec_response(hda, true, 0);
592         break;
593     case AC_VERB_GET_STREAM_FORMAT:
594         st = a->st + node->stindex;
595         if (st->node == NULL) {
596             goto fail;
597         }
598         hda_codec_response(hda, true, st->format);
599         break;
600     case AC_VERB_GET_AMP_GAIN_MUTE:
601         st = a->st + node->stindex;
602         if (st->node == NULL) {
603             goto fail;
604         }
605         if (payload & AC_AMP_GET_LEFT) {
606             response = st->gain_left | (st->mute_left ? AC_AMP_MUTE : 0);
607         } else {
608             response = st->gain_right | (st->mute_right ? AC_AMP_MUTE : 0);
609         }
610         hda_codec_response(hda, true, response);
611         break;
612     case AC_VERB_SET_AMP_GAIN_MUTE:
613         st = a->st + node->stindex;
614         if (st->node == NULL) {
615             goto fail;
616         }
617         dprint(a, 1, "amp (%s): %s%s%s%s index %d  gain %3d %s\n",
618                st->node->name,
619                (payload & AC_AMP_SET_OUTPUT) ? "o" : "-",
620                (payload & AC_AMP_SET_INPUT)  ? "i" : "-",
621                (payload & AC_AMP_SET_LEFT)   ? "l" : "-",
622                (payload & AC_AMP_SET_RIGHT)  ? "r" : "-",
623                (payload & AC_AMP_SET_INDEX) >> AC_AMP_SET_INDEX_SHIFT,
624                (payload & AC_AMP_GAIN),
625                (payload & AC_AMP_MUTE) ? "muted" : "");
626         if (payload & AC_AMP_SET_LEFT) {
627             st->gain_left = payload & AC_AMP_GAIN;
628             st->mute_left = payload & AC_AMP_MUTE;
629         }
630         if (payload & AC_AMP_SET_RIGHT) {
631             st->gain_right = payload & AC_AMP_GAIN;
632             st->mute_right = payload & AC_AMP_MUTE;
633         }
634         hda_audio_set_amp(st);
635         hda_codec_response(hda, true, 0);
636         break;
637 
638     /* not supported */
639     case AC_VERB_SET_POWER_STATE:
640     case AC_VERB_GET_POWER_STATE:
641     case AC_VERB_GET_SDI_SELECT:
642         hda_codec_response(hda, true, 0);
643         break;
644     default:
645         goto fail;
646     }
647     return;
648 
649 fail:
650     dprint(a, 1, "%s: not handled: nid %d (%s), verb 0x%x, payload 0x%x\n",
651            __func__, nid, node ? node->name : "?", verb, payload);
652     hda_codec_response(hda, true, 0);
653 }
654 
655 static void hda_audio_stream(HDACodecDevice *hda, uint32_t stnr, bool running, bool output)
656 {
657     HDAAudioState *a = HDA_AUDIO(hda);
658     int s;
659 
660     a->running_compat[stnr] = running;
661     a->running_real[output * 16 + stnr] = running;
662     for (s = 0; s < ARRAY_SIZE(a->st); s++) {
663         if (a->st[s].node == NULL) {
664             continue;
665         }
666         if (a->st[s].output != output) {
667             continue;
668         }
669         if (a->st[s].stream != stnr) {
670             continue;
671         }
672         hda_audio_set_running(&a->st[s], running);
673     }
674 }
675 
676 static int hda_audio_init(HDACodecDevice *hda, const struct desc_codec *desc)
677 {
678     HDAAudioState *a = HDA_AUDIO(hda);
679     HDAAudioStream *st;
680     const desc_node *node;
681     const desc_param *param;
682     uint32_t i, type;
683 
684     a->desc = desc;
685     a->name = object_get_typename(OBJECT(a));
686     dprint(a, 1, "%s: cad %d\n", __func__, a->hda.cad);
687 
688     AUD_register_card("hda", &a->card);
689     for (i = 0; i < a->desc->nnodes; i++) {
690         node = a->desc->nodes + i;
691         param = hda_codec_find_param(node, AC_PAR_AUDIO_WIDGET_CAP);
692         if (param == NULL) {
693             continue;
694         }
695         type = (param->val & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
696         switch (type) {
697         case AC_WID_AUD_OUT:
698         case AC_WID_AUD_IN:
699             assert(node->stindex < ARRAY_SIZE(a->st));
700             st = a->st + node->stindex;
701             st->state = a;
702             st->node = node;
703             if (type == AC_WID_AUD_OUT) {
704                 /* unmute output by default */
705                 st->gain_left = QEMU_HDA_AMP_STEPS;
706                 st->gain_right = QEMU_HDA_AMP_STEPS;
707                 st->compat_bpos = sizeof(st->compat_buf);
708                 st->output = true;
709             } else {
710                 st->output = false;
711             }
712             st->format = AC_FMT_TYPE_PCM | AC_FMT_BITS_16 |
713                 (1 << AC_FMT_CHAN_SHIFT);
714             hda_codec_parse_fmt(st->format, &st->as);
715             hda_audio_setup(st);
716             break;
717         }
718     }
719     return 0;
720 }
721 
722 static void hda_audio_exit(HDACodecDevice *hda)
723 {
724     HDAAudioState *a = HDA_AUDIO(hda);
725     HDAAudioStream *st;
726     int i;
727 
728     dprint(a, 1, "%s\n", __func__);
729     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
730         st = a->st + i;
731         if (st->node == NULL) {
732             continue;
733         }
734         if (a->use_timer) {
735             timer_del(st->buft);
736         }
737         if (st->output) {
738             AUD_close_out(&a->card, st->voice.out);
739         } else {
740             AUD_close_in(&a->card, st->voice.in);
741         }
742     }
743     AUD_remove_card(&a->card);
744 }
745 
746 static int hda_audio_post_load(void *opaque, int version)
747 {
748     HDAAudioState *a = opaque;
749     HDAAudioStream *st;
750     int i;
751 
752     dprint(a, 1, "%s\n", __func__);
753     if (version == 1) {
754         /* assume running_compat[] is for output streams */
755         for (i = 0; i < ARRAY_SIZE(a->running_compat); i++)
756             a->running_real[16 + i] = a->running_compat[i];
757     }
758 
759     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
760         st = a->st + i;
761         if (st->node == NULL)
762             continue;
763         hda_codec_parse_fmt(st->format, &st->as);
764         hda_audio_setup(st);
765         hda_audio_set_amp(st);
766         hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
767     }
768     return 0;
769 }
770 
771 static void hda_audio_reset(DeviceState *dev)
772 {
773     HDAAudioState *a = HDA_AUDIO(dev);
774     HDAAudioStream *st;
775     int i;
776 
777     dprint(a, 1, "%s\n", __func__);
778     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
779         st = a->st + i;
780         if (st->node != NULL) {
781             hda_audio_set_running(st, false);
782         }
783     }
784 }
785 
786 static bool vmstate_hda_audio_stream_buf_needed(void *opaque)
787 {
788     HDAAudioStream *st = opaque;
789     return st->state && st->state->use_timer;
790 }
791 
792 static const VMStateDescription vmstate_hda_audio_stream_buf = {
793     .name = "hda-audio-stream/buffer",
794     .version_id = 1,
795     .needed = vmstate_hda_audio_stream_buf_needed,
796     .fields = (VMStateField[]) {
797         VMSTATE_BUFFER(buf, HDAAudioStream),
798         VMSTATE_INT64(rpos, HDAAudioStream),
799         VMSTATE_INT64(wpos, HDAAudioStream),
800         VMSTATE_TIMER_PTR(buft, HDAAudioStream),
801         VMSTATE_INT64(buft_start, HDAAudioStream),
802         VMSTATE_END_OF_LIST()
803     }
804 };
805 
806 static const VMStateDescription vmstate_hda_audio_stream = {
807     .name = "hda-audio-stream",
808     .version_id = 1,
809     .fields = (VMStateField[]) {
810         VMSTATE_UINT32(stream, HDAAudioStream),
811         VMSTATE_UINT32(channel, HDAAudioStream),
812         VMSTATE_UINT32(format, HDAAudioStream),
813         VMSTATE_UINT32(gain_left, HDAAudioStream),
814         VMSTATE_UINT32(gain_right, HDAAudioStream),
815         VMSTATE_BOOL(mute_left, HDAAudioStream),
816         VMSTATE_BOOL(mute_right, HDAAudioStream),
817         VMSTATE_UINT32(compat_bpos, HDAAudioStream),
818         VMSTATE_BUFFER(compat_buf, HDAAudioStream),
819         VMSTATE_END_OF_LIST()
820     },
821     .subsections = (const VMStateDescription * []) {
822         &vmstate_hda_audio_stream_buf,
823         NULL
824     }
825 };
826 
827 static const VMStateDescription vmstate_hda_audio = {
828     .name = "hda-audio",
829     .version_id = 2,
830     .post_load = hda_audio_post_load,
831     .fields = (VMStateField[]) {
832         VMSTATE_STRUCT_ARRAY(st, HDAAudioState, 4, 0,
833                              vmstate_hda_audio_stream,
834                              HDAAudioStream),
835         VMSTATE_BOOL_ARRAY(running_compat, HDAAudioState, 16),
836         VMSTATE_BOOL_ARRAY_V(running_real, HDAAudioState, 2 * 16, 2),
837         VMSTATE_END_OF_LIST()
838     }
839 };
840 
841 static Property hda_audio_properties[] = {
842     DEFINE_PROP_UINT32("debug", HDAAudioState, debug,   0),
843     DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer,  true),
844     DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer,  true),
845     DEFINE_PROP_END_OF_LIST(),
846 };
847 
848 static int hda_audio_init_output(HDACodecDevice *hda)
849 {
850     HDAAudioState *a = HDA_AUDIO(hda);
851 
852     if (!a->mixer) {
853         return hda_audio_init(hda, &output_nomixemu);
854     } else {
855         return hda_audio_init(hda, &output_mixemu);
856     }
857 }
858 
859 static int hda_audio_init_duplex(HDACodecDevice *hda)
860 {
861     HDAAudioState *a = HDA_AUDIO(hda);
862 
863     if (!a->mixer) {
864         return hda_audio_init(hda, &duplex_nomixemu);
865     } else {
866         return hda_audio_init(hda, &duplex_mixemu);
867     }
868 }
869 
870 static int hda_audio_init_micro(HDACodecDevice *hda)
871 {
872     HDAAudioState *a = HDA_AUDIO(hda);
873 
874     if (!a->mixer) {
875         return hda_audio_init(hda, &micro_nomixemu);
876     } else {
877         return hda_audio_init(hda, &micro_mixemu);
878     }
879 }
880 
881 static void hda_audio_base_class_init(ObjectClass *klass, void *data)
882 {
883     DeviceClass *dc = DEVICE_CLASS(klass);
884     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
885 
886     k->exit = hda_audio_exit;
887     k->command = hda_audio_command;
888     k->stream = hda_audio_stream;
889     set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
890     dc->reset = hda_audio_reset;
891     dc->vmsd = &vmstate_hda_audio;
892     dc->props = hda_audio_properties;
893 }
894 
895 static const TypeInfo hda_audio_info = {
896     .name          = TYPE_HDA_AUDIO,
897     .parent        = TYPE_HDA_CODEC_DEVICE,
898     .class_init    = hda_audio_base_class_init,
899     .abstract      = true,
900 };
901 
902 static void hda_audio_output_class_init(ObjectClass *klass, void *data)
903 {
904     DeviceClass *dc = DEVICE_CLASS(klass);
905     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
906 
907     k->init = hda_audio_init_output;
908     dc->desc = "HDA Audio Codec, output-only (line-out)";
909 }
910 
911 static const TypeInfo hda_audio_output_info = {
912     .name          = "hda-output",
913     .parent        = TYPE_HDA_AUDIO,
914     .instance_size = sizeof(HDAAudioState),
915     .class_init    = hda_audio_output_class_init,
916 };
917 
918 static void hda_audio_duplex_class_init(ObjectClass *klass, void *data)
919 {
920     DeviceClass *dc = DEVICE_CLASS(klass);
921     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
922 
923     k->init = hda_audio_init_duplex;
924     dc->desc = "HDA Audio Codec, duplex (line-out, line-in)";
925 }
926 
927 static const TypeInfo hda_audio_duplex_info = {
928     .name          = "hda-duplex",
929     .parent        = TYPE_HDA_AUDIO,
930     .instance_size = sizeof(HDAAudioState),
931     .class_init    = hda_audio_duplex_class_init,
932 };
933 
934 static void hda_audio_micro_class_init(ObjectClass *klass, void *data)
935 {
936     DeviceClass *dc = DEVICE_CLASS(klass);
937     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
938 
939     k->init = hda_audio_init_micro;
940     dc->desc = "HDA Audio Codec, duplex (speaker, microphone)";
941 }
942 
943 static const TypeInfo hda_audio_micro_info = {
944     .name          = "hda-micro",
945     .parent        = TYPE_HDA_AUDIO,
946     .instance_size = sizeof(HDAAudioState),
947     .class_init    = hda_audio_micro_class_init,
948 };
949 
950 static void hda_audio_register_types(void)
951 {
952     type_register_static(&hda_audio_info);
953     type_register_static(&hda_audio_output_info);
954     type_register_static(&hda_audio_duplex_info);
955     type_register_static(&hda_audio_micro_info);
956 }
957 
958 type_init(hda_audio_register_types)
959