xref: /openbmc/qemu/audio/audio.c (revision 587adaca)
1 /*
2  * QEMU Audio subsystem
3  *
4  * Copyright (c) 2003-2005 Vassili Karpov (malc)
5  *
6  * Permission is hereby granted, free of charge, to any person obtaining a copy
7  * of this software and associated documentation files (the "Software"), to deal
8  * in the Software without restriction, including without limitation the rights
9  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
10  * copies of the Software, and to permit persons to whom the Software is
11  * furnished to do so, subject to the following conditions:
12  *
13  * The above copyright notice and this permission notice shall be included in
14  * all copies or substantial portions of the Software.
15  *
16  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
17  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
18  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
19  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
20  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
21  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
22  * THE SOFTWARE.
23  */
24 
25 #include "qemu/osdep.h"
26 #include "audio.h"
27 #include "migration/vmstate.h"
28 #include "monitor/monitor.h"
29 #include "qemu/timer.h"
30 #include "qapi/error.h"
31 #include "qapi/qobject-input-visitor.h"
32 #include "qapi/qapi-visit-audio.h"
33 #include "qemu/cutils.h"
34 #include "qemu/module.h"
35 #include "qemu-common.h"
36 #include "sysemu/replay.h"
37 #include "sysemu/runstate.h"
38 #include "ui/qemu-spice.h"
39 #include "trace.h"
40 
41 #define AUDIO_CAP "audio"
42 #include "audio_int.h"
43 
44 /* #define DEBUG_LIVE */
45 /* #define DEBUG_OUT */
46 /* #define DEBUG_CAPTURE */
47 /* #define DEBUG_POLL */
48 
49 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
50 
51 
52 /* Order of CONFIG_AUDIO_DRIVERS is import.
53    The 1st one is the one used by default, that is the reason
54     that we generate the list.
55 */
56 const char *audio_prio_list[] = {
57     "spice",
58     CONFIG_AUDIO_DRIVERS
59     "none",
60     "wav",
61     NULL
62 };
63 
64 static QLIST_HEAD(, audio_driver) audio_drivers;
65 static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
66 
67 void audio_driver_register(audio_driver *drv)
68 {
69     QLIST_INSERT_HEAD(&audio_drivers, drv, next);
70 }
71 
72 audio_driver *audio_driver_lookup(const char *name)
73 {
74     struct audio_driver *d;
75 
76     QLIST_FOREACH(d, &audio_drivers, next) {
77         if (strcmp(name, d->name) == 0) {
78             return d;
79         }
80     }
81 
82     audio_module_load_one(name);
83     QLIST_FOREACH(d, &audio_drivers, next) {
84         if (strcmp(name, d->name) == 0) {
85             return d;
86         }
87     }
88 
89     return NULL;
90 }
91 
92 static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
93     QTAILQ_HEAD_INITIALIZER(audio_states);
94 
95 const struct mixeng_volume nominal_volume = {
96     .mute = 0,
97 #ifdef FLOAT_MIXENG
98     .r = 1.0,
99     .l = 1.0,
100 #else
101     .r = 1ULL << 32,
102     .l = 1ULL << 32,
103 #endif
104 };
105 
106 static bool legacy_config = true;
107 
108 int audio_bug (const char *funcname, int cond)
109 {
110     if (cond) {
111         static int shown;
112 
113         AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
114         if (!shown) {
115             shown = 1;
116             AUD_log (NULL, "Save all your work and restart without audio\n");
117             AUD_log (NULL, "I am sorry\n");
118         }
119         AUD_log (NULL, "Context:\n");
120         abort();
121     }
122 
123     return cond;
124 }
125 
126 static inline int audio_bits_to_index (int bits)
127 {
128     switch (bits) {
129     case 8:
130         return 0;
131 
132     case 16:
133         return 1;
134 
135     case 32:
136         return 2;
137 
138     default:
139         audio_bug ("bits_to_index", 1);
140         AUD_log (NULL, "invalid bits %d\n", bits);
141         return 0;
142     }
143 }
144 
145 void *audio_calloc (const char *funcname, int nmemb, size_t size)
146 {
147     int cond;
148     size_t len;
149 
150     len = nmemb * size;
151     cond = !nmemb || !size;
152     cond |= nmemb < 0;
153     cond |= len < size;
154 
155     if (audio_bug ("audio_calloc", cond)) {
156         AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
157                  funcname);
158         AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
159         return NULL;
160     }
161 
162     return g_malloc0 (len);
163 }
164 
165 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
166 {
167     if (cap) {
168         fprintf(stderr, "%s: ", cap);
169     }
170 
171     vfprintf(stderr, fmt, ap);
172 }
173 
174 void AUD_log (const char *cap, const char *fmt, ...)
175 {
176     va_list ap;
177 
178     va_start (ap, fmt);
179     AUD_vlog (cap, fmt, ap);
180     va_end (ap);
181 }
182 
183 static void audio_print_settings (struct audsettings *as)
184 {
185     dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
186 
187     switch (as->fmt) {
188     case AUDIO_FORMAT_S8:
189         AUD_log (NULL, "S8");
190         break;
191     case AUDIO_FORMAT_U8:
192         AUD_log (NULL, "U8");
193         break;
194     case AUDIO_FORMAT_S16:
195         AUD_log (NULL, "S16");
196         break;
197     case AUDIO_FORMAT_U16:
198         AUD_log (NULL, "U16");
199         break;
200     case AUDIO_FORMAT_S32:
201         AUD_log (NULL, "S32");
202         break;
203     case AUDIO_FORMAT_U32:
204         AUD_log (NULL, "U32");
205         break;
206     case AUDIO_FORMAT_F32:
207         AUD_log (NULL, "F32");
208         break;
209     default:
210         AUD_log (NULL, "invalid(%d)", as->fmt);
211         break;
212     }
213 
214     AUD_log (NULL, " endianness=");
215     switch (as->endianness) {
216     case 0:
217         AUD_log (NULL, "little");
218         break;
219     case 1:
220         AUD_log (NULL, "big");
221         break;
222     default:
223         AUD_log (NULL, "invalid");
224         break;
225     }
226     AUD_log (NULL, "\n");
227 }
228 
229 static int audio_validate_settings (struct audsettings *as)
230 {
231     int invalid;
232 
233     invalid = as->nchannels < 1;
234     invalid |= as->endianness != 0 && as->endianness != 1;
235 
236     switch (as->fmt) {
237     case AUDIO_FORMAT_S8:
238     case AUDIO_FORMAT_U8:
239     case AUDIO_FORMAT_S16:
240     case AUDIO_FORMAT_U16:
241     case AUDIO_FORMAT_S32:
242     case AUDIO_FORMAT_U32:
243     case AUDIO_FORMAT_F32:
244         break;
245     default:
246         invalid = 1;
247         break;
248     }
249 
250     invalid |= as->freq <= 0;
251     return invalid ? -1 : 0;
252 }
253 
254 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
255 {
256     int bits = 8;
257     bool is_signed = false, is_float = false;
258 
259     switch (as->fmt) {
260     case AUDIO_FORMAT_S8:
261         is_signed = true;
262         /* fall through */
263     case AUDIO_FORMAT_U8:
264         break;
265 
266     case AUDIO_FORMAT_S16:
267         is_signed = true;
268         /* fall through */
269     case AUDIO_FORMAT_U16:
270         bits = 16;
271         break;
272 
273     case AUDIO_FORMAT_F32:
274         is_float = true;
275         /* fall through */
276     case AUDIO_FORMAT_S32:
277         is_signed = true;
278         /* fall through */
279     case AUDIO_FORMAT_U32:
280         bits = 32;
281         break;
282 
283     default:
284         abort();
285     }
286     return info->freq == as->freq
287         && info->nchannels == as->nchannels
288         && info->is_signed == is_signed
289         && info->is_float == is_float
290         && info->bits == bits
291         && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
292 }
293 
294 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
295 {
296     int bits = 8, mul;
297     bool is_signed = false, is_float = false;
298 
299     switch (as->fmt) {
300     case AUDIO_FORMAT_S8:
301         is_signed = true;
302         /* fall through */
303     case AUDIO_FORMAT_U8:
304         mul = 1;
305         break;
306 
307     case AUDIO_FORMAT_S16:
308         is_signed = true;
309         /* fall through */
310     case AUDIO_FORMAT_U16:
311         bits = 16;
312         mul = 2;
313         break;
314 
315     case AUDIO_FORMAT_F32:
316         is_float = true;
317         /* fall through */
318     case AUDIO_FORMAT_S32:
319         is_signed = true;
320         /* fall through */
321     case AUDIO_FORMAT_U32:
322         bits = 32;
323         mul = 4;
324         break;
325 
326     default:
327         abort();
328     }
329 
330     info->freq = as->freq;
331     info->bits = bits;
332     info->is_signed = is_signed;
333     info->is_float = is_float;
334     info->nchannels = as->nchannels;
335     info->bytes_per_frame = as->nchannels * mul;
336     info->bytes_per_second = info->freq * info->bytes_per_frame;
337     info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
338 }
339 
340 void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
341 {
342     if (!len) {
343         return;
344     }
345 
346     if (info->is_signed || info->is_float) {
347         memset(buf, 0x00, len * info->bytes_per_frame);
348     } else {
349         switch (info->bits) {
350         case 8:
351             memset(buf, 0x80, len * info->bytes_per_frame);
352             break;
353 
354         case 16:
355             {
356                 int i;
357                 uint16_t *p = buf;
358                 short s = INT16_MAX;
359 
360                 if (info->swap_endianness) {
361                     s = bswap16 (s);
362                 }
363 
364                 for (i = 0; i < len * info->nchannels; i++) {
365                     p[i] = s;
366                 }
367             }
368             break;
369 
370         case 32:
371             {
372                 int i;
373                 uint32_t *p = buf;
374                 int32_t s = INT32_MAX;
375 
376                 if (info->swap_endianness) {
377                     s = bswap32 (s);
378                 }
379 
380                 for (i = 0; i < len * info->nchannels; i++) {
381                     p[i] = s;
382                 }
383             }
384             break;
385 
386         default:
387             AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
388                      info->bits);
389             break;
390         }
391     }
392 }
393 
394 /*
395  * Capture
396  */
397 static void noop_conv (struct st_sample *dst, const void *src, int samples)
398 {
399     (void) src;
400     (void) dst;
401     (void) samples;
402 }
403 
404 static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
405                                                         struct audsettings *as)
406 {
407     CaptureVoiceOut *cap;
408 
409     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
410         if (audio_pcm_info_eq (&cap->hw.info, as)) {
411             return cap;
412         }
413     }
414     return NULL;
415 }
416 
417 static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
418 {
419     struct capture_callback *cb;
420 
421 #ifdef DEBUG_CAPTURE
422     dolog ("notification %d sent\n", cmd);
423 #endif
424     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
425         cb->ops.notify (cb->opaque, cmd);
426     }
427 }
428 
429 static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
430 {
431     if (cap->hw.enabled != enabled) {
432         audcnotification_e cmd;
433         cap->hw.enabled = enabled;
434         cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
435         audio_notify_capture (cap, cmd);
436     }
437 }
438 
439 static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
440 {
441     HWVoiceOut *hw = &cap->hw;
442     SWVoiceOut *sw;
443     int enabled = 0;
444 
445     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
446         if (sw->active) {
447             enabled = 1;
448             break;
449         }
450     }
451     audio_capture_maybe_changed (cap, enabled);
452 }
453 
454 static void audio_detach_capture (HWVoiceOut *hw)
455 {
456     SWVoiceCap *sc = hw->cap_head.lh_first;
457 
458     while (sc) {
459         SWVoiceCap *sc1 = sc->entries.le_next;
460         SWVoiceOut *sw = &sc->sw;
461         CaptureVoiceOut *cap = sc->cap;
462         int was_active = sw->active;
463 
464         if (sw->rate) {
465             st_rate_stop (sw->rate);
466             sw->rate = NULL;
467         }
468 
469         QLIST_REMOVE (sw, entries);
470         QLIST_REMOVE (sc, entries);
471         g_free (sc);
472         if (was_active) {
473             /* We have removed soft voice from the capture:
474                this might have changed the overall status of the capture
475                since this might have been the only active voice */
476             audio_recalc_and_notify_capture (cap);
477         }
478         sc = sc1;
479     }
480 }
481 
482 static int audio_attach_capture (HWVoiceOut *hw)
483 {
484     AudioState *s = hw->s;
485     CaptureVoiceOut *cap;
486 
487     audio_detach_capture (hw);
488     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
489         SWVoiceCap *sc;
490         SWVoiceOut *sw;
491         HWVoiceOut *hw_cap = &cap->hw;
492 
493         sc = g_malloc0(sizeof(*sc));
494 
495         sc->cap = cap;
496         sw = &sc->sw;
497         sw->hw = hw_cap;
498         sw->info = hw->info;
499         sw->empty = 1;
500         sw->active = hw->enabled;
501         sw->conv = noop_conv;
502         sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
503         sw->vol = nominal_volume;
504         sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
505         if (!sw->rate) {
506             dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
507             g_free (sw);
508             return -1;
509         }
510         QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
511         QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
512 #ifdef DEBUG_CAPTURE
513         sw->name = g_strdup_printf ("for %p %d,%d,%d",
514                                     hw, sw->info.freq, sw->info.bits,
515                                     sw->info.nchannels);
516         dolog ("Added %s active = %d\n", sw->name, sw->active);
517 #endif
518         if (sw->active) {
519             audio_capture_maybe_changed (cap, 1);
520         }
521     }
522     return 0;
523 }
524 
525 /*
526  * Hard voice (capture)
527  */
528 static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
529 {
530     SWVoiceIn *sw;
531     size_t m = hw->total_samples_captured;
532 
533     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
534         if (sw->active) {
535             m = MIN (m, sw->total_hw_samples_acquired);
536         }
537     }
538     return m;
539 }
540 
541 static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
542 {
543     size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
544     if (audio_bug(__func__, live > hw->conv_buf->size)) {
545         dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
546         return 0;
547     }
548     return live;
549 }
550 
551 static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
552 {
553     size_t clipped = 0;
554     size_t pos = hw->mix_buf->pos;
555 
556     while (len) {
557         st_sample *src = hw->mix_buf->samples + pos;
558         uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
559         size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
560         size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
561 
562         hw->clip(dst, src, samples_to_clip);
563 
564         pos = (pos + samples_to_clip) % hw->mix_buf->size;
565         len -= samples_to_clip;
566         clipped += samples_to_clip;
567     }
568 }
569 
570 /*
571  * Soft voice (capture)
572  */
573 static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
574 {
575     HWVoiceIn *hw = sw->hw;
576     ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
577     ssize_t rpos;
578 
579     if (audio_bug(__func__, live < 0 || live > hw->conv_buf->size)) {
580         dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
581         return 0;
582     }
583 
584     rpos = hw->conv_buf->pos - live;
585     if (rpos >= 0) {
586         return rpos;
587     } else {
588         return hw->conv_buf->size + rpos;
589     }
590 }
591 
592 static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
593 {
594     HWVoiceIn *hw = sw->hw;
595     size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
596     struct st_sample *src, *dst = sw->buf;
597 
598     rpos = audio_pcm_sw_get_rpos_in(sw) % hw->conv_buf->size;
599 
600     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
601     if (audio_bug(__func__, live > hw->conv_buf->size)) {
602         dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
603         return 0;
604     }
605 
606     samples = size / sw->info.bytes_per_frame;
607     if (!live) {
608         return 0;
609     }
610 
611     swlim = (live * sw->ratio) >> 32;
612     swlim = MIN (swlim, samples);
613 
614     while (swlim) {
615         src = hw->conv_buf->samples + rpos;
616         if (hw->conv_buf->pos > rpos) {
617             isamp = hw->conv_buf->pos - rpos;
618         } else {
619             isamp = hw->conv_buf->size - rpos;
620         }
621 
622         if (!isamp) {
623             break;
624         }
625         osamp = swlim;
626 
627         st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
628         swlim -= osamp;
629         rpos = (rpos + isamp) % hw->conv_buf->size;
630         dst += osamp;
631         ret += osamp;
632         total += isamp;
633     }
634 
635     if (hw->pcm_ops && !hw->pcm_ops->volume_in) {
636         mixeng_volume (sw->buf, ret, &sw->vol);
637     }
638 
639     sw->clip (buf, sw->buf, ret);
640     sw->total_hw_samples_acquired += total;
641     return ret * sw->info.bytes_per_frame;
642 }
643 
644 /*
645  * Hard voice (playback)
646  */
647 static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
648 {
649     SWVoiceOut *sw;
650     size_t m = SIZE_MAX;
651     int nb_live = 0;
652 
653     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
654         if (sw->active || !sw->empty) {
655             m = MIN (m, sw->total_hw_samples_mixed);
656             nb_live += 1;
657         }
658     }
659 
660     *nb_livep = nb_live;
661     return m;
662 }
663 
664 static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
665 {
666     size_t smin;
667     int nb_live1;
668 
669     smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
670     if (nb_live) {
671         *nb_live = nb_live1;
672     }
673 
674     if (nb_live1) {
675         size_t live = smin;
676 
677         if (audio_bug(__func__, live > hw->mix_buf->size)) {
678             dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
679             return 0;
680         }
681         return live;
682     }
683     return 0;
684 }
685 
686 /*
687  * Soft voice (playback)
688  */
689 static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
690 {
691     size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
692     size_t ret = 0, pos = 0, total = 0;
693 
694     if (!sw) {
695         return size;
696     }
697 
698     hwsamples = sw->hw->mix_buf->size;
699 
700     live = sw->total_hw_samples_mixed;
701     if (audio_bug(__func__, live > hwsamples)) {
702         dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
703         return 0;
704     }
705 
706     if (live == hwsamples) {
707 #ifdef DEBUG_OUT
708         dolog ("%s is full %zu\n", sw->name, live);
709 #endif
710         return 0;
711     }
712 
713     wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
714     samples = size / sw->info.bytes_per_frame;
715 
716     dead = hwsamples - live;
717     swlim = ((int64_t) dead << 32) / sw->ratio;
718     swlim = MIN (swlim, samples);
719     if (swlim) {
720         sw->conv (sw->buf, buf, swlim);
721 
722         if (sw->hw->pcm_ops && !sw->hw->pcm_ops->volume_out) {
723             mixeng_volume (sw->buf, swlim, &sw->vol);
724         }
725     }
726 
727     while (swlim) {
728         dead = hwsamples - live;
729         left = hwsamples - wpos;
730         blck = MIN (dead, left);
731         if (!blck) {
732             break;
733         }
734         isamp = swlim;
735         osamp = blck;
736         st_rate_flow_mix (
737             sw->rate,
738             sw->buf + pos,
739             sw->hw->mix_buf->samples + wpos,
740             &isamp,
741             &osamp
742             );
743         ret += isamp;
744         swlim -= isamp;
745         pos += isamp;
746         live += osamp;
747         wpos = (wpos + osamp) % hwsamples;
748         total += osamp;
749     }
750 
751     sw->total_hw_samples_mixed += total;
752     sw->empty = sw->total_hw_samples_mixed == 0;
753 
754 #ifdef DEBUG_OUT
755     dolog (
756         "%s: write size %zu ret %zu total sw %zu\n",
757         SW_NAME (sw),
758         size / sw->info.bytes_per_frame,
759         ret,
760         sw->total_hw_samples_mixed
761         );
762 #endif
763 
764     return ret * sw->info.bytes_per_frame;
765 }
766 
767 #ifdef DEBUG_AUDIO
768 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
769 {
770     dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
771           cap, info->bits, info->is_signed, info->is_float, info->freq,
772           info->nchannels);
773 }
774 #endif
775 
776 #define DAC
777 #include "audio_template.h"
778 #undef DAC
779 #include "audio_template.h"
780 
781 /*
782  * Timer
783  */
784 static int audio_is_timer_needed(AudioState *s)
785 {
786     HWVoiceIn *hwi = NULL;
787     HWVoiceOut *hwo = NULL;
788 
789     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
790         if (!hwo->poll_mode) {
791             return 1;
792         }
793     }
794     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
795         if (!hwi->poll_mode) {
796             return 1;
797         }
798     }
799     return 0;
800 }
801 
802 static void audio_reset_timer (AudioState *s)
803 {
804     if (audio_is_timer_needed(s)) {
805         timer_mod_anticipate_ns(s->ts,
806             qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
807         if (!s->timer_running) {
808             s->timer_running = true;
809             s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
810             trace_audio_timer_start(s->period_ticks / SCALE_MS);
811         }
812     } else {
813         timer_del(s->ts);
814         if (s->timer_running) {
815             s->timer_running = false;
816             trace_audio_timer_stop();
817         }
818     }
819 }
820 
821 static void audio_timer (void *opaque)
822 {
823     int64_t now, diff;
824     AudioState *s = opaque;
825 
826     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
827     diff = now - s->timer_last;
828     if (diff > s->period_ticks * 3 / 2) {
829         trace_audio_timer_delayed(diff / SCALE_MS);
830     }
831     s->timer_last = now;
832 
833     audio_run(s, "timer");
834     audio_reset_timer(s);
835 }
836 
837 /*
838  * Public API
839  */
840 size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
841 {
842     HWVoiceOut *hw;
843 
844     if (!sw) {
845         /* XXX: Consider options */
846         return size;
847     }
848     hw = sw->hw;
849 
850     if (!hw->enabled) {
851         dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
852         return 0;
853     }
854 
855     if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
856         return audio_pcm_sw_write(sw, buf, size);
857     } else {
858         return hw->pcm_ops->write(hw, buf, size);
859     }
860 }
861 
862 size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
863 {
864     HWVoiceIn *hw;
865 
866     if (!sw) {
867         /* XXX: Consider options */
868         return size;
869     }
870     hw = sw->hw;
871 
872     if (!hw->enabled) {
873         dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
874         return 0;
875     }
876 
877     if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
878         return audio_pcm_sw_read(sw, buf, size);
879     } else {
880         return hw->pcm_ops->read(hw, buf, size);
881     }
882 }
883 
884 int AUD_get_buffer_size_out(SWVoiceOut *sw)
885 {
886     return sw->hw->samples * sw->hw->info.bytes_per_frame;
887 }
888 
889 void AUD_set_active_out (SWVoiceOut *sw, int on)
890 {
891     HWVoiceOut *hw;
892 
893     if (!sw) {
894         return;
895     }
896 
897     hw = sw->hw;
898     if (sw->active != on) {
899         AudioState *s = sw->s;
900         SWVoiceOut *temp_sw;
901         SWVoiceCap *sc;
902 
903         if (on) {
904             hw->pending_disable = 0;
905             if (!hw->enabled) {
906                 hw->enabled = 1;
907                 if (s->vm_running) {
908                     if (hw->pcm_ops->enable_out) {
909                         hw->pcm_ops->enable_out(hw, true);
910                     }
911                     audio_reset_timer (s);
912                 }
913             }
914         } else {
915             if (hw->enabled) {
916                 int nb_active = 0;
917 
918                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
919                      temp_sw = temp_sw->entries.le_next) {
920                     nb_active += temp_sw->active != 0;
921                 }
922 
923                 hw->pending_disable = nb_active == 1;
924             }
925         }
926 
927         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
928             sc->sw.active = hw->enabled;
929             if (hw->enabled) {
930                 audio_capture_maybe_changed (sc->cap, 1);
931             }
932         }
933         sw->active = on;
934     }
935 }
936 
937 void AUD_set_active_in (SWVoiceIn *sw, int on)
938 {
939     HWVoiceIn *hw;
940 
941     if (!sw) {
942         return;
943     }
944 
945     hw = sw->hw;
946     if (sw->active != on) {
947         AudioState *s = sw->s;
948         SWVoiceIn *temp_sw;
949 
950         if (on) {
951             if (!hw->enabled) {
952                 hw->enabled = 1;
953                 if (s->vm_running) {
954                     if (hw->pcm_ops->enable_in) {
955                         hw->pcm_ops->enable_in(hw, true);
956                     }
957                     audio_reset_timer (s);
958                 }
959             }
960             sw->total_hw_samples_acquired = hw->total_samples_captured;
961         } else {
962             if (hw->enabled) {
963                 int nb_active = 0;
964 
965                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
966                      temp_sw = temp_sw->entries.le_next) {
967                     nb_active += temp_sw->active != 0;
968                 }
969 
970                 if (nb_active == 1) {
971                     hw->enabled = 0;
972                     if (hw->pcm_ops->enable_in) {
973                         hw->pcm_ops->enable_in(hw, false);
974                     }
975                 }
976             }
977         }
978         sw->active = on;
979     }
980 }
981 
982 static size_t audio_get_avail (SWVoiceIn *sw)
983 {
984     size_t live;
985 
986     if (!sw) {
987         return 0;
988     }
989 
990     live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
991     if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
992         dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
993               sw->hw->conv_buf->size);
994         return 0;
995     }
996 
997     ldebug (
998         "%s: get_avail live %zu ret %" PRId64 "\n",
999         SW_NAME (sw),
1000         live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
1001         );
1002 
1003     return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
1004 }
1005 
1006 static size_t audio_get_free(SWVoiceOut *sw)
1007 {
1008     size_t live, dead;
1009 
1010     if (!sw) {
1011         return 0;
1012     }
1013 
1014     live = sw->total_hw_samples_mixed;
1015 
1016     if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
1017         dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
1018               sw->hw->mix_buf->size);
1019         return 0;
1020     }
1021 
1022     dead = sw->hw->mix_buf->size - live;
1023 
1024 #ifdef DEBUG_OUT
1025     dolog ("%s: get_free live %zu dead %zu ret %" PRId64 "\n",
1026            SW_NAME (sw),
1027            live, dead, (((int64_t) dead << 32) / sw->ratio) *
1028            sw->info.bytes_per_frame);
1029 #endif
1030 
1031     return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
1032 }
1033 
1034 static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
1035                                         size_t samples)
1036 {
1037     size_t n;
1038 
1039     if (hw->enabled) {
1040         SWVoiceCap *sc;
1041 
1042         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1043             SWVoiceOut *sw = &sc->sw;
1044             int rpos2 = rpos;
1045 
1046             n = samples;
1047             while (n) {
1048                 size_t till_end_of_hw = hw->mix_buf->size - rpos2;
1049                 size_t to_write = MIN(till_end_of_hw, n);
1050                 size_t bytes = to_write * hw->info.bytes_per_frame;
1051                 size_t written;
1052 
1053                 sw->buf = hw->mix_buf->samples + rpos2;
1054                 written = audio_pcm_sw_write (sw, NULL, bytes);
1055                 if (written - bytes) {
1056                     dolog("Could not mix %zu bytes into a capture "
1057                           "buffer, mixed %zu\n",
1058                           bytes, written);
1059                     break;
1060                 }
1061                 n -= to_write;
1062                 rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
1063             }
1064         }
1065     }
1066 
1067     n = MIN(samples, hw->mix_buf->size - rpos);
1068     mixeng_clear(hw->mix_buf->samples + rpos, n);
1069     mixeng_clear(hw->mix_buf->samples, samples - n);
1070 }
1071 
1072 static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
1073 {
1074     size_t clipped = 0;
1075 
1076     while (live) {
1077         size_t size = live * hw->info.bytes_per_frame;
1078         size_t decr, proc;
1079         void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
1080 
1081         if (size == 0) {
1082             break;
1083         }
1084 
1085         decr = MIN(size / hw->info.bytes_per_frame, live);
1086         if (buf) {
1087             audio_pcm_hw_clip_out(hw, buf, decr);
1088         }
1089         proc = hw->pcm_ops->put_buffer_out(hw, buf,
1090                                            decr * hw->info.bytes_per_frame) /
1091             hw->info.bytes_per_frame;
1092 
1093         live -= proc;
1094         clipped += proc;
1095         hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
1096 
1097         if (proc == 0 || proc < decr) {
1098             break;
1099         }
1100     }
1101 
1102     if (hw->pcm_ops->run_buffer_out) {
1103         hw->pcm_ops->run_buffer_out(hw);
1104     }
1105 
1106     return clipped;
1107 }
1108 
1109 static void audio_run_out (AudioState *s)
1110 {
1111     HWVoiceOut *hw = NULL;
1112     SWVoiceOut *sw;
1113 
1114     if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1115         while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
1116             /* there is exactly 1 sw for each hw with no mixeng */
1117             sw = hw->sw_head.lh_first;
1118 
1119             if (hw->pending_disable) {
1120                 hw->enabled = 0;
1121                 hw->pending_disable = 0;
1122                 if (hw->pcm_ops->enable_out) {
1123                     hw->pcm_ops->enable_out(hw, false);
1124                 }
1125             }
1126 
1127             if (sw->active) {
1128                 sw->callback.fn(sw->callback.opaque, INT_MAX);
1129             }
1130         }
1131         return;
1132     }
1133 
1134     while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
1135         size_t played, live, prev_rpos, free;
1136         int nb_live;
1137 
1138         live = audio_pcm_hw_get_live_out (hw, &nb_live);
1139         if (!nb_live) {
1140             live = 0;
1141         }
1142 
1143         if (audio_bug(__func__, live > hw->mix_buf->size)) {
1144             dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
1145             continue;
1146         }
1147 
1148         if (hw->pending_disable && !nb_live) {
1149             SWVoiceCap *sc;
1150 #ifdef DEBUG_OUT
1151             dolog ("Disabling voice\n");
1152 #endif
1153             hw->enabled = 0;
1154             hw->pending_disable = 0;
1155             if (hw->pcm_ops->enable_out) {
1156                 hw->pcm_ops->enable_out(hw, false);
1157             }
1158             for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1159                 sc->sw.active = 0;
1160                 audio_recalc_and_notify_capture (sc->cap);
1161             }
1162             continue;
1163         }
1164 
1165         if (!live) {
1166             for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1167                 if (sw->active) {
1168                     free = audio_get_free (sw);
1169                     if (free > 0) {
1170                         sw->callback.fn (sw->callback.opaque, free);
1171                     }
1172                 }
1173             }
1174             if (hw->pcm_ops->run_buffer_out) {
1175                 hw->pcm_ops->run_buffer_out(hw);
1176             }
1177             continue;
1178         }
1179 
1180         prev_rpos = hw->mix_buf->pos;
1181         played = audio_pcm_hw_run_out(hw, live);
1182         replay_audio_out(&played);
1183         if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
1184             dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
1185                   hw->mix_buf->pos, hw->mix_buf->size, played);
1186             hw->mix_buf->pos = 0;
1187         }
1188 
1189 #ifdef DEBUG_OUT
1190         dolog("played=%zu\n", played);
1191 #endif
1192 
1193         if (played) {
1194             hw->ts_helper += played;
1195             audio_capture_mix_and_clear (hw, prev_rpos, played);
1196         }
1197 
1198         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1199             if (!sw->active && sw->empty) {
1200                 continue;
1201             }
1202 
1203             if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
1204                 dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
1205                       played, sw->total_hw_samples_mixed);
1206                 played = sw->total_hw_samples_mixed;
1207             }
1208 
1209             sw->total_hw_samples_mixed -= played;
1210 
1211             if (!sw->total_hw_samples_mixed) {
1212                 sw->empty = 1;
1213             }
1214 
1215             if (sw->active) {
1216                 free = audio_get_free (sw);
1217                 if (free > 0) {
1218                     sw->callback.fn (sw->callback.opaque, free);
1219                 }
1220             }
1221         }
1222     }
1223 }
1224 
1225 static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
1226 {
1227     size_t conv = 0;
1228     STSampleBuffer *conv_buf = hw->conv_buf;
1229 
1230     if (hw->pcm_ops->run_buffer_in) {
1231         hw->pcm_ops->run_buffer_in(hw);
1232     }
1233 
1234     while (samples) {
1235         size_t proc;
1236         size_t size = samples * hw->info.bytes_per_frame;
1237         void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
1238 
1239         assert(size % hw->info.bytes_per_frame == 0);
1240         if (size == 0) {
1241             break;
1242         }
1243 
1244         proc = MIN(size / hw->info.bytes_per_frame,
1245                    conv_buf->size - conv_buf->pos);
1246 
1247         hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
1248         conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
1249 
1250         samples -= proc;
1251         conv += proc;
1252         hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
1253     }
1254 
1255     return conv;
1256 }
1257 
1258 static void audio_run_in (AudioState *s)
1259 {
1260     HWVoiceIn *hw = NULL;
1261 
1262     if (!audio_get_pdo_in(s->dev)->mixing_engine) {
1263         while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1264             /* there is exactly 1 sw for each hw with no mixeng */
1265             SWVoiceIn *sw = hw->sw_head.lh_first;
1266             if (sw->active) {
1267                 sw->callback.fn(sw->callback.opaque, INT_MAX);
1268             }
1269         }
1270         return;
1271     }
1272 
1273     while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1274         SWVoiceIn *sw;
1275         size_t captured = 0, min;
1276 
1277         if (replay_mode != REPLAY_MODE_PLAY) {
1278             captured = audio_pcm_hw_run_in(
1279                 hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
1280         }
1281         replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
1282                         hw->conv_buf->size);
1283 
1284         min = audio_pcm_hw_find_min_in (hw);
1285         hw->total_samples_captured += captured - min;
1286         hw->ts_helper += captured;
1287 
1288         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1289             sw->total_hw_samples_acquired -= min;
1290 
1291             if (sw->active) {
1292                 size_t avail;
1293 
1294                 avail = audio_get_avail (sw);
1295                 if (avail > 0) {
1296                     sw->callback.fn (sw->callback.opaque, avail);
1297                 }
1298             }
1299         }
1300     }
1301 }
1302 
1303 static void audio_run_capture (AudioState *s)
1304 {
1305     CaptureVoiceOut *cap;
1306 
1307     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
1308         size_t live, rpos, captured;
1309         HWVoiceOut *hw = &cap->hw;
1310         SWVoiceOut *sw;
1311 
1312         captured = live = audio_pcm_hw_get_live_out (hw, NULL);
1313         rpos = hw->mix_buf->pos;
1314         while (live) {
1315             size_t left = hw->mix_buf->size - rpos;
1316             size_t to_capture = MIN(live, left);
1317             struct st_sample *src;
1318             struct capture_callback *cb;
1319 
1320             src = hw->mix_buf->samples + rpos;
1321             hw->clip (cap->buf, src, to_capture);
1322             mixeng_clear (src, to_capture);
1323 
1324             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1325                 cb->ops.capture (cb->opaque, cap->buf,
1326                                  to_capture * hw->info.bytes_per_frame);
1327             }
1328             rpos = (rpos + to_capture) % hw->mix_buf->size;
1329             live -= to_capture;
1330         }
1331         hw->mix_buf->pos = rpos;
1332 
1333         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1334             if (!sw->active && sw->empty) {
1335                 continue;
1336             }
1337 
1338             if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
1339                 dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
1340                       captured, sw->total_hw_samples_mixed);
1341                 captured = sw->total_hw_samples_mixed;
1342             }
1343 
1344             sw->total_hw_samples_mixed -= captured;
1345             sw->empty = sw->total_hw_samples_mixed == 0;
1346         }
1347     }
1348 }
1349 
1350 void audio_run(AudioState *s, const char *msg)
1351 {
1352     audio_run_out(s);
1353     audio_run_in(s);
1354     audio_run_capture(s);
1355 
1356 #ifdef DEBUG_POLL
1357     {
1358         static double prevtime;
1359         double currtime;
1360         struct timeval tv;
1361 
1362         if (gettimeofday (&tv, NULL)) {
1363             perror ("audio_run: gettimeofday");
1364             return;
1365         }
1366 
1367         currtime = tv.tv_sec + tv.tv_usec * 1e-6;
1368         dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
1369         prevtime = currtime;
1370     }
1371 #endif
1372 }
1373 
1374 void audio_generic_run_buffer_in(HWVoiceIn *hw)
1375 {
1376     if (unlikely(!hw->buf_emul)) {
1377         hw->size_emul = hw->samples * hw->info.bytes_per_frame;
1378         hw->buf_emul = g_malloc(hw->size_emul);
1379         hw->pos_emul = hw->pending_emul = 0;
1380     }
1381 
1382     while (hw->pending_emul < hw->size_emul) {
1383         size_t read_len = MIN(hw->size_emul - hw->pos_emul,
1384                               hw->size_emul - hw->pending_emul);
1385         size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
1386                                         read_len);
1387         hw->pending_emul += read;
1388         hw->pos_emul = (hw->pos_emul + read) % hw->size_emul;
1389         if (read < read_len) {
1390             break;
1391         }
1392     }
1393 }
1394 
1395 void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
1396 {
1397     ssize_t start = (ssize_t)hw->pos_emul - hw->pending_emul;
1398 
1399     if (start < 0) {
1400         start += hw->size_emul;
1401     }
1402     assert(start >= 0 && start < hw->size_emul);
1403 
1404     *size = MIN(*size, hw->pending_emul);
1405     *size = MIN(*size, hw->size_emul - start);
1406     return hw->buf_emul + start;
1407 }
1408 
1409 void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
1410 {
1411     assert(size <= hw->pending_emul);
1412     hw->pending_emul -= size;
1413 }
1414 
1415 void audio_generic_run_buffer_out(HWVoiceOut *hw)
1416 {
1417     while (hw->pending_emul) {
1418         size_t write_len, written;
1419         ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
1420 
1421         if (start < 0) {
1422             start += hw->size_emul;
1423         }
1424         assert(start >= 0 && start < hw->size_emul);
1425 
1426         write_len = MIN(hw->pending_emul, hw->size_emul - start);
1427 
1428         written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
1429         hw->pending_emul -= written;
1430 
1431         if (written < write_len) {
1432             break;
1433         }
1434     }
1435 }
1436 
1437 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
1438 {
1439     if (unlikely(!hw->buf_emul)) {
1440         hw->size_emul = hw->samples * hw->info.bytes_per_frame;
1441         hw->buf_emul = g_malloc(hw->size_emul);
1442         hw->pos_emul = hw->pending_emul = 0;
1443     }
1444 
1445     *size = MIN(hw->size_emul - hw->pending_emul,
1446                 hw->size_emul - hw->pos_emul);
1447     return hw->buf_emul + hw->pos_emul;
1448 }
1449 
1450 size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
1451 {
1452     assert(buf == hw->buf_emul + hw->pos_emul &&
1453            size + hw->pending_emul <= hw->size_emul);
1454 
1455     hw->pending_emul += size;
1456     hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
1457 
1458     return size;
1459 }
1460 
1461 size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
1462 {
1463     size_t total = 0;
1464 
1465     while (total < size) {
1466         size_t dst_size = size - total;
1467         size_t copy_size, proc;
1468         void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
1469 
1470         if (dst_size == 0) {
1471             break;
1472         }
1473 
1474         copy_size = MIN(size - total, dst_size);
1475         if (dst) {
1476             memcpy(dst, (char *)buf + total, copy_size);
1477         }
1478         proc = hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
1479         total += proc;
1480 
1481         if (proc == 0 || proc < copy_size) {
1482             break;
1483         }
1484     }
1485 
1486     if (hw->pcm_ops->run_buffer_out) {
1487         hw->pcm_ops->run_buffer_out(hw);
1488     }
1489 
1490     return total;
1491 }
1492 
1493 size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
1494 {
1495     size_t total = 0;
1496 
1497     if (hw->pcm_ops->run_buffer_in) {
1498         hw->pcm_ops->run_buffer_in(hw);
1499     }
1500 
1501     while (total < size) {
1502         size_t src_size = size - total;
1503         void *src = hw->pcm_ops->get_buffer_in(hw, &src_size);
1504 
1505         if (src_size == 0) {
1506             break;
1507         }
1508 
1509         memcpy((char *)buf + total, src, src_size);
1510         hw->pcm_ops->put_buffer_in(hw, src, src_size);
1511         total += src_size;
1512     }
1513 
1514     return total;
1515 }
1516 
1517 static int audio_driver_init(AudioState *s, struct audio_driver *drv,
1518                              bool msg, Audiodev *dev)
1519 {
1520     s->drv_opaque = drv->init(dev);
1521 
1522     if (s->drv_opaque) {
1523         if (!drv->pcm_ops->get_buffer_in) {
1524             drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
1525             drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
1526         }
1527         if (!drv->pcm_ops->get_buffer_out) {
1528             drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
1529             drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
1530         }
1531 
1532         audio_init_nb_voices_out(s, drv);
1533         audio_init_nb_voices_in(s, drv);
1534         s->drv = drv;
1535         return 0;
1536     } else {
1537         if (msg) {
1538             dolog("Could not init `%s' audio driver\n", drv->name);
1539         }
1540         return -1;
1541     }
1542 }
1543 
1544 static void audio_vm_change_state_handler (void *opaque, bool running,
1545                                            RunState state)
1546 {
1547     AudioState *s = opaque;
1548     HWVoiceOut *hwo = NULL;
1549     HWVoiceIn *hwi = NULL;
1550 
1551     s->vm_running = running;
1552     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
1553         if (hwo->pcm_ops->enable_out) {
1554             hwo->pcm_ops->enable_out(hwo, running);
1555         }
1556     }
1557 
1558     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
1559         if (hwi->pcm_ops->enable_in) {
1560             hwi->pcm_ops->enable_in(hwi, running);
1561         }
1562     }
1563     audio_reset_timer (s);
1564 }
1565 
1566 static void free_audio_state(AudioState *s)
1567 {
1568     HWVoiceOut *hwo, *hwon;
1569     HWVoiceIn *hwi, *hwin;
1570 
1571     QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
1572         SWVoiceCap *sc;
1573 
1574         if (hwo->enabled && hwo->pcm_ops->enable_out) {
1575             hwo->pcm_ops->enable_out(hwo, false);
1576         }
1577         hwo->pcm_ops->fini_out (hwo);
1578 
1579         for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1580             CaptureVoiceOut *cap = sc->cap;
1581             struct capture_callback *cb;
1582 
1583             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1584                 cb->ops.destroy (cb->opaque);
1585             }
1586         }
1587         QLIST_REMOVE(hwo, entries);
1588     }
1589 
1590     QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
1591         if (hwi->enabled && hwi->pcm_ops->enable_in) {
1592             hwi->pcm_ops->enable_in(hwi, false);
1593         }
1594         hwi->pcm_ops->fini_in (hwi);
1595         QLIST_REMOVE(hwi, entries);
1596     }
1597 
1598     if (s->drv) {
1599         s->drv->fini (s->drv_opaque);
1600         s->drv = NULL;
1601     }
1602 
1603     if (s->dev) {
1604         qapi_free_Audiodev(s->dev);
1605         s->dev = NULL;
1606     }
1607 
1608     if (s->ts) {
1609         timer_free(s->ts);
1610         s->ts = NULL;
1611     }
1612 
1613     g_free(s);
1614 }
1615 
1616 void audio_cleanup(void)
1617 {
1618     while (!QTAILQ_EMPTY(&audio_states)) {
1619         AudioState *s = QTAILQ_FIRST(&audio_states);
1620         QTAILQ_REMOVE(&audio_states, s, list);
1621         free_audio_state(s);
1622     }
1623 }
1624 
1625 static const VMStateDescription vmstate_audio = {
1626     .name = "audio",
1627     .version_id = 1,
1628     .minimum_version_id = 1,
1629     .fields = (VMStateField[]) {
1630         VMSTATE_END_OF_LIST()
1631     }
1632 };
1633 
1634 static void audio_validate_opts(Audiodev *dev, Error **errp);
1635 
1636 static AudiodevListEntry *audiodev_find(
1637     AudiodevListHead *head, const char *drvname)
1638 {
1639     AudiodevListEntry *e;
1640     QSIMPLEQ_FOREACH(e, head, next) {
1641         if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) {
1642             return e;
1643         }
1644     }
1645 
1646     return NULL;
1647 }
1648 
1649 /*
1650  * if we have dev, this function was called because of an -audiodev argument =>
1651  *   initialize a new state with it
1652  * if dev == NULL => legacy implicit initialization, return the already created
1653  *   state or create a new one
1654  */
1655 static AudioState *audio_init(Audiodev *dev, const char *name)
1656 {
1657     static bool atexit_registered;
1658     size_t i;
1659     int done = 0;
1660     const char *drvname = NULL;
1661     VMChangeStateEntry *e;
1662     AudioState *s;
1663     struct audio_driver *driver;
1664     /* silence gcc warning about uninitialized variable */
1665     AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
1666 
1667     if (using_spice) {
1668         /*
1669          * When using spice allow the spice audio driver being picked
1670          * as default.
1671          *
1672          * Temporary hack.  Using audio devices without explicit
1673          * audiodev= property is already deprecated.  Same goes for
1674          * the -soundhw switch.  Once this support gets finally
1675          * removed we can also drop the concept of a default audio
1676          * backend and this can go away.
1677          */
1678         driver = audio_driver_lookup("spice");
1679         if (driver) {
1680             driver->can_be_default = 1;
1681         }
1682     }
1683 
1684     if (dev) {
1685         /* -audiodev option */
1686         legacy_config = false;
1687         drvname = AudiodevDriver_str(dev->driver);
1688     } else if (!QTAILQ_EMPTY(&audio_states)) {
1689         if (!legacy_config) {
1690             dolog("Device %s: audiodev default parameter is deprecated, please "
1691                   "specify audiodev=%s\n", name,
1692                   QTAILQ_FIRST(&audio_states)->dev->id);
1693         }
1694         return QTAILQ_FIRST(&audio_states);
1695     } else {
1696         /* legacy implicit initialization */
1697         head = audio_handle_legacy_opts();
1698         /*
1699          * In case of legacy initialization, all Audiodevs in the list will have
1700          * the same configuration (except the driver), so it doesn't matter which
1701          * one we chose.  We need an Audiodev to set up AudioState before we can
1702          * init a driver.  Also note that dev at this point is still in the
1703          * list.
1704          */
1705         dev = QSIMPLEQ_FIRST(&head)->dev;
1706         audio_validate_opts(dev, &error_abort);
1707     }
1708 
1709     s = g_malloc0(sizeof(AudioState));
1710     s->dev = dev;
1711 
1712     QLIST_INIT (&s->hw_head_out);
1713     QLIST_INIT (&s->hw_head_in);
1714     QLIST_INIT (&s->cap_head);
1715     if (!atexit_registered) {
1716         atexit(audio_cleanup);
1717         atexit_registered = true;
1718     }
1719     QTAILQ_INSERT_TAIL(&audio_states, s, list);
1720 
1721     s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
1722 
1723     s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices;
1724     s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices;
1725 
1726     if (s->nb_hw_voices_out <= 0) {
1727         dolog ("Bogus number of playback voices %d, setting to 1\n",
1728                s->nb_hw_voices_out);
1729         s->nb_hw_voices_out = 1;
1730     }
1731 
1732     if (s->nb_hw_voices_in <= 0) {
1733         dolog ("Bogus number of capture voices %d, setting to 0\n",
1734                s->nb_hw_voices_in);
1735         s->nb_hw_voices_in = 0;
1736     }
1737 
1738     if (drvname) {
1739         driver = audio_driver_lookup(drvname);
1740         if (driver) {
1741             done = !audio_driver_init(s, driver, true, dev);
1742         } else {
1743             dolog ("Unknown audio driver `%s'\n", drvname);
1744         }
1745     } else {
1746         for (i = 0; audio_prio_list[i]; i++) {
1747             AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]);
1748             driver = audio_driver_lookup(audio_prio_list[i]);
1749 
1750             if (e && driver) {
1751                 s->dev = dev = e->dev;
1752                 audio_validate_opts(dev, &error_abort);
1753                 done = !audio_driver_init(s, driver, false, dev);
1754                 if (done) {
1755                     e->dev = NULL;
1756                     break;
1757                 }
1758             }
1759         }
1760     }
1761     audio_free_audiodev_list(&head);
1762 
1763     if (!done) {
1764         driver = audio_driver_lookup("none");
1765         done = !audio_driver_init(s, driver, false, dev);
1766         assert(done);
1767         dolog("warning: Using timer based audio emulation\n");
1768     }
1769 
1770     if (dev->timer_period <= 0) {
1771         s->period_ticks = 1;
1772     } else {
1773         s->period_ticks = dev->timer_period * (int64_t)SCALE_US;
1774     }
1775 
1776     e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
1777     if (!e) {
1778         dolog ("warning: Could not register change state handler\n"
1779                "(Audio can continue looping even after stopping the VM)\n");
1780     }
1781 
1782     QLIST_INIT (&s->card_head);
1783     vmstate_register (NULL, 0, &vmstate_audio, s);
1784     return s;
1785 }
1786 
1787 void audio_free_audiodev_list(AudiodevListHead *head)
1788 {
1789     AudiodevListEntry *e;
1790     while ((e = QSIMPLEQ_FIRST(head))) {
1791         QSIMPLEQ_REMOVE_HEAD(head, next);
1792         qapi_free_Audiodev(e->dev);
1793         g_free(e);
1794     }
1795 }
1796 
1797 void AUD_register_card (const char *name, QEMUSoundCard *card)
1798 {
1799     if (!card->state) {
1800         card->state = audio_init(NULL, name);
1801     }
1802 
1803     card->name = g_strdup (name);
1804     memset (&card->entries, 0, sizeof (card->entries));
1805     QLIST_INSERT_HEAD(&card->state->card_head, card, entries);
1806 }
1807 
1808 void AUD_remove_card (QEMUSoundCard *card)
1809 {
1810     QLIST_REMOVE (card, entries);
1811     g_free (card->name);
1812 }
1813 
1814 
1815 CaptureVoiceOut *AUD_add_capture(
1816     AudioState *s,
1817     struct audsettings *as,
1818     struct audio_capture_ops *ops,
1819     void *cb_opaque
1820     )
1821 {
1822     CaptureVoiceOut *cap;
1823     struct capture_callback *cb;
1824 
1825     if (!s) {
1826         if (!legacy_config) {
1827             dolog("Capturing without setting an audiodev is deprecated\n");
1828         }
1829         s = audio_init(NULL, NULL);
1830     }
1831 
1832     if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1833         dolog("Can't capture with mixeng disabled\n");
1834         return NULL;
1835     }
1836 
1837     if (audio_validate_settings (as)) {
1838         dolog ("Invalid settings were passed when trying to add capture\n");
1839         audio_print_settings (as);
1840         return NULL;
1841     }
1842 
1843     cb = g_malloc0(sizeof(*cb));
1844     cb->ops = *ops;
1845     cb->opaque = cb_opaque;
1846 
1847     cap = audio_pcm_capture_find_specific(s, as);
1848     if (cap) {
1849         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1850         return cap;
1851     } else {
1852         HWVoiceOut *hw;
1853         CaptureVoiceOut *cap;
1854 
1855         cap = g_malloc0(sizeof(*cap));
1856 
1857         hw = &cap->hw;
1858         hw->s = s;
1859         QLIST_INIT (&hw->sw_head);
1860         QLIST_INIT (&cap->cb_head);
1861 
1862         /* XXX find a more elegant way */
1863         hw->samples = 4096 * 4;
1864         audio_pcm_hw_alloc_resources_out(hw);
1865 
1866         audio_pcm_init_info (&hw->info, as);
1867 
1868         cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
1869 
1870         if (hw->info.is_float) {
1871             hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
1872         } else {
1873             hw->clip = mixeng_clip
1874                 [hw->info.nchannels == 2]
1875                 [hw->info.is_signed]
1876                 [hw->info.swap_endianness]
1877                 [audio_bits_to_index(hw->info.bits)];
1878         }
1879 
1880         QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
1881         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1882 
1883         QLIST_FOREACH(hw, &s->hw_head_out, entries) {
1884             audio_attach_capture (hw);
1885         }
1886         return cap;
1887     }
1888 }
1889 
1890 void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
1891 {
1892     struct capture_callback *cb;
1893 
1894     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1895         if (cb->opaque == cb_opaque) {
1896             cb->ops.destroy (cb_opaque);
1897             QLIST_REMOVE (cb, entries);
1898             g_free (cb);
1899 
1900             if (!cap->cb_head.lh_first) {
1901                 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
1902 
1903                 while (sw) {
1904                     SWVoiceCap *sc = (SWVoiceCap *) sw;
1905 #ifdef DEBUG_CAPTURE
1906                     dolog ("freeing %s\n", sw->name);
1907 #endif
1908 
1909                     sw1 = sw->entries.le_next;
1910                     if (sw->rate) {
1911                         st_rate_stop (sw->rate);
1912                         sw->rate = NULL;
1913                     }
1914                     QLIST_REMOVE (sw, entries);
1915                     QLIST_REMOVE (sc, entries);
1916                     g_free (sc);
1917                     sw = sw1;
1918                 }
1919                 QLIST_REMOVE (cap, entries);
1920                 g_free (cap->hw.mix_buf);
1921                 g_free (cap->buf);
1922                 g_free (cap);
1923             }
1924             return;
1925         }
1926     }
1927 }
1928 
1929 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
1930 {
1931     Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1932     audio_set_volume_out(sw, &vol);
1933 }
1934 
1935 void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
1936 {
1937     if (sw) {
1938         HWVoiceOut *hw = sw->hw;
1939 
1940         sw->vol.mute = vol->mute;
1941         sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
1942         sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
1943             255;
1944 
1945         if (hw->pcm_ops->volume_out) {
1946             hw->pcm_ops->volume_out(hw, vol);
1947         }
1948     }
1949 }
1950 
1951 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
1952 {
1953     Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1954     audio_set_volume_in(sw, &vol);
1955 }
1956 
1957 void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
1958 {
1959     if (sw) {
1960         HWVoiceIn *hw = sw->hw;
1961 
1962         sw->vol.mute = vol->mute;
1963         sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
1964         sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
1965             255;
1966 
1967         if (hw->pcm_ops->volume_in) {
1968             hw->pcm_ops->volume_in(hw, vol);
1969         }
1970     }
1971 }
1972 
1973 void audio_create_pdos(Audiodev *dev)
1974 {
1975     switch (dev->driver) {
1976 #define CASE(DRIVER, driver, pdo_name)                              \
1977     case AUDIODEV_DRIVER_##DRIVER:                                  \
1978         if (!dev->u.driver.has_in) {                                \
1979             dev->u.driver.in = g_malloc0(                           \
1980                 sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
1981             dev->u.driver.has_in = true;                            \
1982         }                                                           \
1983         if (!dev->u.driver.has_out) {                               \
1984             dev->u.driver.out = g_malloc0(                          \
1985                 sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
1986             dev->u.driver.has_out = true;                           \
1987         }                                                           \
1988         break
1989 
1990         CASE(NONE, none, );
1991         CASE(ALSA, alsa, Alsa);
1992         CASE(COREAUDIO, coreaudio, Coreaudio);
1993         CASE(DSOUND, dsound, );
1994         CASE(JACK, jack, Jack);
1995         CASE(OSS, oss, Oss);
1996         CASE(PA, pa, Pa);
1997         CASE(SDL, sdl, Sdl);
1998         CASE(SPICE, spice, );
1999         CASE(WAV, wav, );
2000 
2001     case AUDIODEV_DRIVER__MAX:
2002         abort();
2003     };
2004 }
2005 
2006 static void audio_validate_per_direction_opts(
2007     AudiodevPerDirectionOptions *pdo, Error **errp)
2008 {
2009     if (!pdo->has_mixing_engine) {
2010         pdo->has_mixing_engine = true;
2011         pdo->mixing_engine = true;
2012     }
2013     if (!pdo->has_fixed_settings) {
2014         pdo->has_fixed_settings = true;
2015         pdo->fixed_settings = pdo->mixing_engine;
2016     }
2017     if (!pdo->fixed_settings &&
2018         (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
2019         error_setg(errp,
2020                    "You can't use frequency, channels or format with fixed-settings=off");
2021         return;
2022     }
2023     if (!pdo->mixing_engine && pdo->fixed_settings) {
2024         error_setg(errp, "You can't use fixed-settings without mixeng");
2025         return;
2026     }
2027 
2028     if (!pdo->has_frequency) {
2029         pdo->has_frequency = true;
2030         pdo->frequency = 44100;
2031     }
2032     if (!pdo->has_channels) {
2033         pdo->has_channels = true;
2034         pdo->channels = 2;
2035     }
2036     if (!pdo->has_voices) {
2037         pdo->has_voices = true;
2038         pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
2039     }
2040     if (!pdo->has_format) {
2041         pdo->has_format = true;
2042         pdo->format = AUDIO_FORMAT_S16;
2043     }
2044 }
2045 
2046 static void audio_validate_opts(Audiodev *dev, Error **errp)
2047 {
2048     Error *err = NULL;
2049 
2050     audio_create_pdos(dev);
2051 
2052     audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err);
2053     if (err) {
2054         error_propagate(errp, err);
2055         return;
2056     }
2057 
2058     audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err);
2059     if (err) {
2060         error_propagate(errp, err);
2061         return;
2062     }
2063 
2064     if (!dev->has_timer_period) {
2065         dev->has_timer_period = true;
2066         dev->timer_period = 10000; /* 100Hz -> 10ms */
2067     }
2068 }
2069 
2070 void audio_parse_option(const char *opt)
2071 {
2072     AudiodevListEntry *e;
2073     Audiodev *dev = NULL;
2074 
2075     Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
2076     visit_type_Audiodev(v, NULL, &dev, &error_fatal);
2077     visit_free(v);
2078 
2079     audio_validate_opts(dev, &error_fatal);
2080 
2081     e = g_malloc0(sizeof(AudiodevListEntry));
2082     e->dev = dev;
2083     QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
2084 }
2085 
2086 void audio_init_audiodevs(void)
2087 {
2088     AudiodevListEntry *e;
2089 
2090     QSIMPLEQ_FOREACH(e, &audiodevs, next) {
2091         audio_init(e->dev, NULL);
2092     }
2093 }
2094 
2095 audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
2096 {
2097     return (audsettings) {
2098         .freq = pdo->frequency,
2099         .nchannels = pdo->channels,
2100         .fmt = pdo->format,
2101         .endianness = AUDIO_HOST_ENDIANNESS,
2102     };
2103 }
2104 
2105 int audioformat_bytes_per_sample(AudioFormat fmt)
2106 {
2107     switch (fmt) {
2108     case AUDIO_FORMAT_U8:
2109     case AUDIO_FORMAT_S8:
2110         return 1;
2111 
2112     case AUDIO_FORMAT_U16:
2113     case AUDIO_FORMAT_S16:
2114         return 2;
2115 
2116     case AUDIO_FORMAT_U32:
2117     case AUDIO_FORMAT_S32:
2118     case AUDIO_FORMAT_F32:
2119         return 4;
2120 
2121     case AUDIO_FORMAT__MAX:
2122         ;
2123     }
2124     abort();
2125 }
2126 
2127 
2128 /* frames = freq * usec / 1e6 */
2129 int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
2130                         audsettings *as, int def_usecs)
2131 {
2132     uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs;
2133     return (as->freq * usecs + 500000) / 1000000;
2134 }
2135 
2136 /* samples = channels * frames = channels * freq * usec / 1e6 */
2137 int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
2138                          audsettings *as, int def_usecs)
2139 {
2140     return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
2141 }
2142 
2143 /*
2144  * bytes = bytes_per_sample * samples =
2145  *     bytes_per_sample * channels * freq * usec / 1e6
2146  */
2147 int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
2148                        audsettings *as, int def_usecs)
2149 {
2150     return audio_buffer_samples(pdo, as, def_usecs) *
2151         audioformat_bytes_per_sample(as->fmt);
2152 }
2153 
2154 AudioState *audio_state_by_name(const char *name)
2155 {
2156     AudioState *s;
2157     QTAILQ_FOREACH(s, &audio_states, list) {
2158         assert(s->dev);
2159         if (strcmp(name, s->dev->id) == 0) {
2160             return s;
2161         }
2162     }
2163     return NULL;
2164 }
2165 
2166 const char *audio_get_id(QEMUSoundCard *card)
2167 {
2168     if (card->state) {
2169         assert(card->state->dev);
2170         return card->state->dev->id;
2171     } else {
2172         return "";
2173     }
2174 }
2175 
2176 const char *audio_application_name(void)
2177 {
2178     const char *vm_name;
2179 
2180     vm_name = qemu_get_vm_name();
2181     return vm_name ? vm_name : "qemu";
2182 }
2183 
2184 void audio_rate_start(RateCtl *rate)
2185 {
2186     memset(rate, 0, sizeof(RateCtl));
2187     rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2188 }
2189 
2190 size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
2191                             size_t bytes_avail)
2192 {
2193     int64_t now;
2194     int64_t ticks;
2195     int64_t bytes;
2196     int64_t samples;
2197     size_t ret;
2198 
2199     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2200     ticks = now - rate->start_ticks;
2201     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
2202     samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
2203     if (samples < 0 || samples > 65536) {
2204         AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
2205         audio_rate_start(rate);
2206         samples = 0;
2207     }
2208 
2209     ret = MIN(samples * info->bytes_per_frame, bytes_avail);
2210     rate->bytes_sent += ret;
2211     return ret;
2212 }
2213