xref: /openbmc/qemu/audio/audio.c (revision 8c6631e6)
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/clone-visitor.h"
32 #include "qapi/qobject-input-visitor.h"
33 #include "qapi/qapi-visit-audio.h"
34 #include "qapi/qapi-commands-audio.h"
35 #include "qemu/cutils.h"
36 #include "qemu/module.h"
37 #include "qemu/help_option.h"
38 #include "sysemu/sysemu.h"
39 #include "sysemu/replay.h"
40 #include "sysemu/runstate.h"
41 #include "ui/qemu-spice.h"
42 #include "trace.h"
43 
44 #define AUDIO_CAP "audio"
45 #include "audio_int.h"
46 
47 /* #define DEBUG_LIVE */
48 /* #define DEBUG_OUT */
49 /* #define DEBUG_CAPTURE */
50 /* #define DEBUG_POLL */
51 
52 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
53 
54 
55 /* Order of CONFIG_AUDIO_DRIVERS is import.
56    The 1st one is the one used by default, that is the reason
57     that we generate the list.
58 */
59 const char *audio_prio_list[] = {
60     "spice",
61     CONFIG_AUDIO_DRIVERS
62     "none",
63     "wav",
64     NULL
65 };
66 
67 static QLIST_HEAD(, audio_driver) audio_drivers;
68 static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
69 
70 void audio_driver_register(audio_driver *drv)
71 {
72     QLIST_INSERT_HEAD(&audio_drivers, drv, next);
73 }
74 
75 audio_driver *audio_driver_lookup(const char *name)
76 {
77     struct audio_driver *d;
78     Error *local_err = NULL;
79     int rv;
80 
81     QLIST_FOREACH(d, &audio_drivers, next) {
82         if (strcmp(name, d->name) == 0) {
83             return d;
84         }
85     }
86     rv = audio_module_load(name, &local_err);
87     if (rv > 0) {
88         QLIST_FOREACH(d, &audio_drivers, next) {
89             if (strcmp(name, d->name) == 0) {
90                 return d;
91             }
92         }
93     } else if (rv < 0) {
94         error_report_err(local_err);
95     }
96     return NULL;
97 }
98 
99 static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
100     QTAILQ_HEAD_INITIALIZER(audio_states);
101 
102 const struct mixeng_volume nominal_volume = {
103     .mute = 0,
104 #ifdef FLOAT_MIXENG
105     .r = 1.0,
106     .l = 1.0,
107 #else
108     .r = 1ULL << 32,
109     .l = 1ULL << 32,
110 #endif
111 };
112 
113 static bool legacy_config = true;
114 
115 int audio_bug (const char *funcname, int cond)
116 {
117     if (cond) {
118         static int shown;
119 
120         AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
121         if (!shown) {
122             shown = 1;
123             AUD_log (NULL, "Save all your work and restart without audio\n");
124             AUD_log (NULL, "I am sorry\n");
125         }
126         AUD_log (NULL, "Context:\n");
127     }
128 
129     return cond;
130 }
131 
132 static inline int audio_bits_to_index (int bits)
133 {
134     switch (bits) {
135     case 8:
136         return 0;
137 
138     case 16:
139         return 1;
140 
141     case 32:
142         return 2;
143 
144     default:
145         audio_bug ("bits_to_index", 1);
146         AUD_log (NULL, "invalid bits %d\n", bits);
147         return 0;
148     }
149 }
150 
151 void *audio_calloc (const char *funcname, int nmemb, size_t size)
152 {
153     int cond;
154     size_t len;
155 
156     len = nmemb * size;
157     cond = !nmemb || !size;
158     cond |= nmemb < 0;
159     cond |= len < size;
160 
161     if (audio_bug ("audio_calloc", cond)) {
162         AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
163                  funcname);
164         AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
165         return NULL;
166     }
167 
168     return g_malloc0 (len);
169 }
170 
171 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
172 {
173     if (cap) {
174         fprintf(stderr, "%s: ", cap);
175     }
176 
177     vfprintf(stderr, fmt, ap);
178 }
179 
180 void AUD_log (const char *cap, const char *fmt, ...)
181 {
182     va_list ap;
183 
184     va_start (ap, fmt);
185     AUD_vlog (cap, fmt, ap);
186     va_end (ap);
187 }
188 
189 static void audio_print_settings (struct audsettings *as)
190 {
191     dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
192 
193     switch (as->fmt) {
194     case AUDIO_FORMAT_S8:
195         AUD_log (NULL, "S8");
196         break;
197     case AUDIO_FORMAT_U8:
198         AUD_log (NULL, "U8");
199         break;
200     case AUDIO_FORMAT_S16:
201         AUD_log (NULL, "S16");
202         break;
203     case AUDIO_FORMAT_U16:
204         AUD_log (NULL, "U16");
205         break;
206     case AUDIO_FORMAT_S32:
207         AUD_log (NULL, "S32");
208         break;
209     case AUDIO_FORMAT_U32:
210         AUD_log (NULL, "U32");
211         break;
212     case AUDIO_FORMAT_F32:
213         AUD_log (NULL, "F32");
214         break;
215     default:
216         AUD_log (NULL, "invalid(%d)", as->fmt);
217         break;
218     }
219 
220     AUD_log (NULL, " endianness=");
221     switch (as->endianness) {
222     case 0:
223         AUD_log (NULL, "little");
224         break;
225     case 1:
226         AUD_log (NULL, "big");
227         break;
228     default:
229         AUD_log (NULL, "invalid");
230         break;
231     }
232     AUD_log (NULL, "\n");
233 }
234 
235 static int audio_validate_settings (struct audsettings *as)
236 {
237     int invalid;
238 
239     invalid = as->nchannels < 1;
240     invalid |= as->endianness != 0 && as->endianness != 1;
241 
242     switch (as->fmt) {
243     case AUDIO_FORMAT_S8:
244     case AUDIO_FORMAT_U8:
245     case AUDIO_FORMAT_S16:
246     case AUDIO_FORMAT_U16:
247     case AUDIO_FORMAT_S32:
248     case AUDIO_FORMAT_U32:
249     case AUDIO_FORMAT_F32:
250         break;
251     default:
252         invalid = 1;
253         break;
254     }
255 
256     invalid |= as->freq <= 0;
257     return invalid ? -1 : 0;
258 }
259 
260 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
261 {
262     int bits = 8;
263     bool is_signed = false, is_float = false;
264 
265     switch (as->fmt) {
266     case AUDIO_FORMAT_S8:
267         is_signed = true;
268         /* fall through */
269     case AUDIO_FORMAT_U8:
270         break;
271 
272     case AUDIO_FORMAT_S16:
273         is_signed = true;
274         /* fall through */
275     case AUDIO_FORMAT_U16:
276         bits = 16;
277         break;
278 
279     case AUDIO_FORMAT_F32:
280         is_float = true;
281         /* fall through */
282     case AUDIO_FORMAT_S32:
283         is_signed = true;
284         /* fall through */
285     case AUDIO_FORMAT_U32:
286         bits = 32;
287         break;
288 
289     default:
290         abort();
291     }
292     return info->freq == as->freq
293         && info->nchannels == as->nchannels
294         && info->is_signed == is_signed
295         && info->is_float == is_float
296         && info->bits == bits
297         && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
298 }
299 
300 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
301 {
302     int bits = 8, mul;
303     bool is_signed = false, is_float = false;
304 
305     switch (as->fmt) {
306     case AUDIO_FORMAT_S8:
307         is_signed = true;
308         /* fall through */
309     case AUDIO_FORMAT_U8:
310         mul = 1;
311         break;
312 
313     case AUDIO_FORMAT_S16:
314         is_signed = true;
315         /* fall through */
316     case AUDIO_FORMAT_U16:
317         bits = 16;
318         mul = 2;
319         break;
320 
321     case AUDIO_FORMAT_F32:
322         is_float = true;
323         /* fall through */
324     case AUDIO_FORMAT_S32:
325         is_signed = true;
326         /* fall through */
327     case AUDIO_FORMAT_U32:
328         bits = 32;
329         mul = 4;
330         break;
331 
332     default:
333         abort();
334     }
335 
336     info->freq = as->freq;
337     info->bits = bits;
338     info->is_signed = is_signed;
339     info->is_float = is_float;
340     info->nchannels = as->nchannels;
341     info->bytes_per_frame = as->nchannels * mul;
342     info->bytes_per_second = info->freq * info->bytes_per_frame;
343     info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
344 }
345 
346 void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
347 {
348     if (!len) {
349         return;
350     }
351 
352     if (info->is_signed || info->is_float) {
353         memset(buf, 0x00, len * info->bytes_per_frame);
354     } else {
355         switch (info->bits) {
356         case 8:
357             memset(buf, 0x80, len * info->bytes_per_frame);
358             break;
359 
360         case 16:
361             {
362                 int i;
363                 uint16_t *p = buf;
364                 short s = INT16_MAX;
365 
366                 if (info->swap_endianness) {
367                     s = bswap16 (s);
368                 }
369 
370                 for (i = 0; i < len * info->nchannels; i++) {
371                     p[i] = s;
372                 }
373             }
374             break;
375 
376         case 32:
377             {
378                 int i;
379                 uint32_t *p = buf;
380                 int32_t s = INT32_MAX;
381 
382                 if (info->swap_endianness) {
383                     s = bswap32 (s);
384                 }
385 
386                 for (i = 0; i < len * info->nchannels; i++) {
387                     p[i] = s;
388                 }
389             }
390             break;
391 
392         default:
393             AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
394                      info->bits);
395             break;
396         }
397     }
398 }
399 
400 /*
401  * Capture
402  */
403 static void noop_conv (struct st_sample *dst, const void *src, int samples)
404 {
405     (void) src;
406     (void) dst;
407     (void) samples;
408 }
409 
410 static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
411                                                         struct audsettings *as)
412 {
413     CaptureVoiceOut *cap;
414 
415     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
416         if (audio_pcm_info_eq (&cap->hw.info, as)) {
417             return cap;
418         }
419     }
420     return NULL;
421 }
422 
423 static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
424 {
425     struct capture_callback *cb;
426 
427 #ifdef DEBUG_CAPTURE
428     dolog ("notification %d sent\n", cmd);
429 #endif
430     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
431         cb->ops.notify (cb->opaque, cmd);
432     }
433 }
434 
435 static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
436 {
437     if (cap->hw.enabled != enabled) {
438         audcnotification_e cmd;
439         cap->hw.enabled = enabled;
440         cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
441         audio_notify_capture (cap, cmd);
442     }
443 }
444 
445 static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
446 {
447     HWVoiceOut *hw = &cap->hw;
448     SWVoiceOut *sw;
449     int enabled = 0;
450 
451     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
452         if (sw->active) {
453             enabled = 1;
454             break;
455         }
456     }
457     audio_capture_maybe_changed (cap, enabled);
458 }
459 
460 static void audio_detach_capture (HWVoiceOut *hw)
461 {
462     SWVoiceCap *sc = hw->cap_head.lh_first;
463 
464     while (sc) {
465         SWVoiceCap *sc1 = sc->entries.le_next;
466         SWVoiceOut *sw = &sc->sw;
467         CaptureVoiceOut *cap = sc->cap;
468         int was_active = sw->active;
469 
470         if (sw->rate) {
471             st_rate_stop (sw->rate);
472             sw->rate = NULL;
473         }
474 
475         QLIST_REMOVE (sw, entries);
476         QLIST_REMOVE (sc, entries);
477         g_free (sc);
478         if (was_active) {
479             /* We have removed soft voice from the capture:
480                this might have changed the overall status of the capture
481                since this might have been the only active voice */
482             audio_recalc_and_notify_capture (cap);
483         }
484         sc = sc1;
485     }
486 }
487 
488 static int audio_attach_capture (HWVoiceOut *hw)
489 {
490     AudioState *s = hw->s;
491     CaptureVoiceOut *cap;
492 
493     audio_detach_capture (hw);
494     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
495         SWVoiceCap *sc;
496         SWVoiceOut *sw;
497         HWVoiceOut *hw_cap = &cap->hw;
498 
499         sc = g_malloc0(sizeof(*sc));
500 
501         sc->cap = cap;
502         sw = &sc->sw;
503         sw->hw = hw_cap;
504         sw->info = hw->info;
505         sw->empty = 1;
506         sw->active = hw->enabled;
507         sw->conv = noop_conv;
508         sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
509         sw->vol = nominal_volume;
510         sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
511         if (!sw->rate) {
512             dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
513             g_free (sw);
514             return -1;
515         }
516         QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
517         QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
518 #ifdef DEBUG_CAPTURE
519         sw->name = g_strdup_printf ("for %p %d,%d,%d",
520                                     hw, sw->info.freq, sw->info.bits,
521                                     sw->info.nchannels);
522         dolog ("Added %s active = %d\n", sw->name, sw->active);
523 #endif
524         if (sw->active) {
525             audio_capture_maybe_changed (cap, 1);
526         }
527     }
528     return 0;
529 }
530 
531 /*
532  * Hard voice (capture)
533  */
534 static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
535 {
536     SWVoiceIn *sw;
537     size_t m = hw->total_samples_captured;
538 
539     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
540         if (sw->active) {
541             m = MIN (m, sw->total_hw_samples_acquired);
542         }
543     }
544     return m;
545 }
546 
547 static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
548 {
549     size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
550     if (audio_bug(__func__, live > hw->conv_buf->size)) {
551         dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
552         return 0;
553     }
554     return live;
555 }
556 
557 static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
558 {
559     size_t conv = 0;
560     STSampleBuffer *conv_buf = hw->conv_buf;
561 
562     while (samples) {
563         uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame);
564         size_t proc = MIN(samples, conv_buf->size - conv_buf->pos);
565 
566         hw->conv(conv_buf->samples + conv_buf->pos, src, proc);
567         conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
568         samples -= proc;
569         conv += proc;
570     }
571 
572     return conv;
573 }
574 
575 /*
576  * Soft voice (capture)
577  */
578 static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
579 {
580     HWVoiceIn *hw = sw->hw;
581     size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
582     struct st_sample *src, *dst = sw->buf;
583 
584     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
585     if (!live) {
586         return 0;
587     }
588     if (audio_bug(__func__, live > hw->conv_buf->size)) {
589         dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
590         return 0;
591     }
592 
593     rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size);
594 
595     samples = size / sw->info.bytes_per_frame;
596 
597     swlim = (live * sw->ratio) >> 32;
598     swlim = MIN (swlim, samples);
599 
600     while (swlim) {
601         src = hw->conv_buf->samples + rpos;
602         if (hw->conv_buf->pos > rpos) {
603             isamp = hw->conv_buf->pos - rpos;
604         } else {
605             isamp = hw->conv_buf->size - rpos;
606         }
607 
608         if (!isamp) {
609             break;
610         }
611         osamp = swlim;
612 
613         st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
614         swlim -= osamp;
615         rpos = (rpos + isamp) % hw->conv_buf->size;
616         dst += osamp;
617         ret += osamp;
618         total += isamp;
619     }
620 
621     if (!hw->pcm_ops->volume_in) {
622         mixeng_volume (sw->buf, ret, &sw->vol);
623     }
624 
625     sw->clip (buf, sw->buf, ret);
626     sw->total_hw_samples_acquired += total;
627     return ret * sw->info.bytes_per_frame;
628 }
629 
630 /*
631  * Hard voice (playback)
632  */
633 static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
634 {
635     SWVoiceOut *sw;
636     size_t m = SIZE_MAX;
637     int nb_live = 0;
638 
639     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
640         if (sw->active || !sw->empty) {
641             m = MIN (m, sw->total_hw_samples_mixed);
642             nb_live += 1;
643         }
644     }
645 
646     *nb_livep = nb_live;
647     return m;
648 }
649 
650 static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
651 {
652     size_t smin;
653     int nb_live1;
654 
655     smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
656     if (nb_live) {
657         *nb_live = nb_live1;
658     }
659 
660     if (nb_live1) {
661         size_t live = smin;
662 
663         if (audio_bug(__func__, live > hw->mix_buf->size)) {
664             dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
665             return 0;
666         }
667         return live;
668     }
669     return 0;
670 }
671 
672 static size_t audio_pcm_hw_get_free(HWVoiceOut *hw)
673 {
674     return (hw->pcm_ops->buffer_get_free ? hw->pcm_ops->buffer_get_free(hw) :
675             INT_MAX) / hw->info.bytes_per_frame;
676 }
677 
678 static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
679 {
680     size_t clipped = 0;
681     size_t pos = hw->mix_buf->pos;
682 
683     while (len) {
684         st_sample *src = hw->mix_buf->samples + pos;
685         uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
686         size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
687         size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
688 
689         hw->clip(dst, src, samples_to_clip);
690 
691         pos = (pos + samples_to_clip) % hw->mix_buf->size;
692         len -= samples_to_clip;
693         clipped += samples_to_clip;
694     }
695 }
696 
697 /*
698  * Soft voice (playback)
699  */
700 static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
701 {
702     size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck;
703     size_t hw_free;
704     size_t ret = 0, pos = 0, total = 0;
705 
706     if (!sw) {
707         return size;
708     }
709 
710     hwsamples = sw->hw->mix_buf->size;
711 
712     live = sw->total_hw_samples_mixed;
713     if (audio_bug(__func__, live > hwsamples)) {
714         dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
715         return 0;
716     }
717 
718     if (live == hwsamples) {
719 #ifdef DEBUG_OUT
720         dolog ("%s is full %zu\n", sw->name, live);
721 #endif
722         return 0;
723     }
724 
725     wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
726 
727     dead = hwsamples - live;
728     hw_free = audio_pcm_hw_get_free(sw->hw);
729     hw_free = hw_free > live ? hw_free - live : 0;
730     samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
731     samples = MIN(samples, size / sw->info.bytes_per_frame);
732     if (samples) {
733         sw->conv(sw->buf, buf, samples);
734 
735         if (!sw->hw->pcm_ops->volume_out) {
736             mixeng_volume(sw->buf, samples, &sw->vol);
737         }
738     }
739 
740     while (samples) {
741         dead = hwsamples - live;
742         left = hwsamples - wpos;
743         blck = MIN (dead, left);
744         if (!blck) {
745             break;
746         }
747         isamp = samples;
748         osamp = blck;
749         st_rate_flow_mix (
750             sw->rate,
751             sw->buf + pos,
752             sw->hw->mix_buf->samples + wpos,
753             &isamp,
754             &osamp
755             );
756         ret += isamp;
757         samples -= isamp;
758         pos += isamp;
759         live += osamp;
760         wpos = (wpos + osamp) % hwsamples;
761         total += osamp;
762     }
763 
764     sw->total_hw_samples_mixed += total;
765     sw->empty = sw->total_hw_samples_mixed == 0;
766 
767 #ifdef DEBUG_OUT
768     dolog (
769         "%s: write size %zu ret %zu total sw %zu\n",
770         SW_NAME (sw),
771         size / sw->info.bytes_per_frame,
772         ret,
773         sw->total_hw_samples_mixed
774         );
775 #endif
776 
777     return ret * sw->info.bytes_per_frame;
778 }
779 
780 #ifdef DEBUG_AUDIO
781 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
782 {
783     dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
784           cap, info->bits, info->is_signed, info->is_float, info->freq,
785           info->nchannels);
786 }
787 #endif
788 
789 #define DAC
790 #include "audio_template.h"
791 #undef DAC
792 #include "audio_template.h"
793 
794 /*
795  * Timer
796  */
797 static int audio_is_timer_needed(AudioState *s)
798 {
799     HWVoiceIn *hwi = NULL;
800     HWVoiceOut *hwo = NULL;
801 
802     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
803         if (!hwo->poll_mode) {
804             return 1;
805         }
806     }
807     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
808         if (!hwi->poll_mode) {
809             return 1;
810         }
811     }
812     return 0;
813 }
814 
815 static void audio_reset_timer (AudioState *s)
816 {
817     if (audio_is_timer_needed(s)) {
818         timer_mod_anticipate_ns(s->ts,
819             qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
820         if (!s->timer_running) {
821             s->timer_running = true;
822             s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
823             trace_audio_timer_start(s->period_ticks / SCALE_MS);
824         }
825     } else {
826         timer_del(s->ts);
827         if (s->timer_running) {
828             s->timer_running = false;
829             trace_audio_timer_stop();
830         }
831     }
832 }
833 
834 static void audio_timer (void *opaque)
835 {
836     int64_t now, diff;
837     AudioState *s = opaque;
838 
839     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
840     diff = now - s->timer_last;
841     if (diff > s->period_ticks * 3 / 2) {
842         trace_audio_timer_delayed(diff / SCALE_MS);
843     }
844     s->timer_last = now;
845 
846     audio_run(s, "timer");
847     audio_reset_timer(s);
848 }
849 
850 /*
851  * Public API
852  */
853 size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
854 {
855     HWVoiceOut *hw;
856 
857     if (!sw) {
858         /* XXX: Consider options */
859         return size;
860     }
861     hw = sw->hw;
862 
863     if (!hw->enabled) {
864         dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
865         return 0;
866     }
867 
868     if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
869         return audio_pcm_sw_write(sw, buf, size);
870     } else {
871         return hw->pcm_ops->write(hw, buf, size);
872     }
873 }
874 
875 size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
876 {
877     HWVoiceIn *hw;
878 
879     if (!sw) {
880         /* XXX: Consider options */
881         return size;
882     }
883     hw = sw->hw;
884 
885     if (!hw->enabled) {
886         dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
887         return 0;
888     }
889 
890     if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
891         return audio_pcm_sw_read(sw, buf, size);
892     } else {
893         return hw->pcm_ops->read(hw, buf, size);
894     }
895 }
896 
897 int AUD_get_buffer_size_out(SWVoiceOut *sw)
898 {
899     return sw->hw->samples * sw->hw->info.bytes_per_frame;
900 }
901 
902 void AUD_set_active_out (SWVoiceOut *sw, int on)
903 {
904     HWVoiceOut *hw;
905 
906     if (!sw) {
907         return;
908     }
909 
910     hw = sw->hw;
911     if (sw->active != on) {
912         AudioState *s = sw->s;
913         SWVoiceOut *temp_sw;
914         SWVoiceCap *sc;
915 
916         if (on) {
917             hw->pending_disable = 0;
918             if (!hw->enabled) {
919                 hw->enabled = 1;
920                 if (s->vm_running) {
921                     if (hw->pcm_ops->enable_out) {
922                         hw->pcm_ops->enable_out(hw, true);
923                     }
924                     audio_reset_timer (s);
925                 }
926             }
927         } else {
928             if (hw->enabled) {
929                 int nb_active = 0;
930 
931                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
932                      temp_sw = temp_sw->entries.le_next) {
933                     nb_active += temp_sw->active != 0;
934                 }
935 
936                 hw->pending_disable = nb_active == 1;
937             }
938         }
939 
940         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
941             sc->sw.active = hw->enabled;
942             if (hw->enabled) {
943                 audio_capture_maybe_changed (sc->cap, 1);
944             }
945         }
946         sw->active = on;
947     }
948 }
949 
950 void AUD_set_active_in (SWVoiceIn *sw, int on)
951 {
952     HWVoiceIn *hw;
953 
954     if (!sw) {
955         return;
956     }
957 
958     hw = sw->hw;
959     if (sw->active != on) {
960         AudioState *s = sw->s;
961         SWVoiceIn *temp_sw;
962 
963         if (on) {
964             if (!hw->enabled) {
965                 hw->enabled = 1;
966                 if (s->vm_running) {
967                     if (hw->pcm_ops->enable_in) {
968                         hw->pcm_ops->enable_in(hw, true);
969                     }
970                     audio_reset_timer (s);
971                 }
972             }
973             sw->total_hw_samples_acquired = hw->total_samples_captured;
974         } else {
975             if (hw->enabled) {
976                 int nb_active = 0;
977 
978                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
979                      temp_sw = temp_sw->entries.le_next) {
980                     nb_active += temp_sw->active != 0;
981                 }
982 
983                 if (nb_active == 1) {
984                     hw->enabled = 0;
985                     if (hw->pcm_ops->enable_in) {
986                         hw->pcm_ops->enable_in(hw, false);
987                     }
988                 }
989             }
990         }
991         sw->active = on;
992     }
993 }
994 
995 /**
996  * audio_frontend_frames_in() - returns the number of frames the resampling
997  * code generates from frames_in frames
998  *
999  * @sw: audio recording frontend
1000  * @frames_in: number of frames
1001  */
1002 static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in)
1003 {
1004     return (int64_t)frames_in * sw->ratio >> 32;
1005 }
1006 
1007 static size_t audio_get_avail (SWVoiceIn *sw)
1008 {
1009     size_t live;
1010 
1011     if (!sw) {
1012         return 0;
1013     }
1014 
1015     live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
1016     if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
1017         dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
1018               sw->hw->conv_buf->size);
1019         return 0;
1020     }
1021 
1022     ldebug (
1023         "%s: get_avail live %zu frontend frames %zu\n",
1024         SW_NAME (sw),
1025         live, audio_frontend_frames_in(sw, live)
1026         );
1027 
1028     return live;
1029 }
1030 
1031 /**
1032  * audio_frontend_frames_out() - returns the number of frames needed to
1033  * get frames_out frames after resampling
1034  *
1035  * @sw: audio playback frontend
1036  * @frames_out: number of frames
1037  */
1038 static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out)
1039 {
1040     return ((int64_t)frames_out << 32) / sw->ratio;
1041 }
1042 
1043 static size_t audio_get_free(SWVoiceOut *sw)
1044 {
1045     size_t live, dead;
1046 
1047     if (!sw) {
1048         return 0;
1049     }
1050 
1051     live = sw->total_hw_samples_mixed;
1052 
1053     if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
1054         dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
1055               sw->hw->mix_buf->size);
1056         return 0;
1057     }
1058 
1059     dead = sw->hw->mix_buf->size - live;
1060 
1061 #ifdef DEBUG_OUT
1062     dolog("%s: get_free live %zu dead %zu frontend frames %zu\n",
1063           SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead));
1064 #endif
1065 
1066     return dead;
1067 }
1068 
1069 static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
1070                                         size_t samples)
1071 {
1072     size_t n;
1073 
1074     if (hw->enabled) {
1075         SWVoiceCap *sc;
1076 
1077         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1078             SWVoiceOut *sw = &sc->sw;
1079             int rpos2 = rpos;
1080 
1081             n = samples;
1082             while (n) {
1083                 size_t till_end_of_hw = hw->mix_buf->size - rpos2;
1084                 size_t to_write = MIN(till_end_of_hw, n);
1085                 size_t bytes = to_write * hw->info.bytes_per_frame;
1086                 size_t written;
1087 
1088                 sw->buf = hw->mix_buf->samples + rpos2;
1089                 written = audio_pcm_sw_write (sw, NULL, bytes);
1090                 if (written - bytes) {
1091                     dolog("Could not mix %zu bytes into a capture "
1092                           "buffer, mixed %zu\n",
1093                           bytes, written);
1094                     break;
1095                 }
1096                 n -= to_write;
1097                 rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
1098             }
1099         }
1100     }
1101 
1102     n = MIN(samples, hw->mix_buf->size - rpos);
1103     mixeng_clear(hw->mix_buf->samples + rpos, n);
1104     mixeng_clear(hw->mix_buf->samples, samples - n);
1105 }
1106 
1107 static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
1108 {
1109     size_t clipped = 0;
1110 
1111     while (live) {
1112         size_t size = live * hw->info.bytes_per_frame;
1113         size_t decr, proc;
1114         void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
1115 
1116         if (size == 0) {
1117             break;
1118         }
1119 
1120         decr = MIN(size / hw->info.bytes_per_frame, live);
1121         if (buf) {
1122             audio_pcm_hw_clip_out(hw, buf, decr);
1123         }
1124         proc = hw->pcm_ops->put_buffer_out(hw, buf,
1125                                            decr * hw->info.bytes_per_frame) /
1126             hw->info.bytes_per_frame;
1127 
1128         live -= proc;
1129         clipped += proc;
1130         hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
1131 
1132         if (proc == 0 || proc < decr) {
1133             break;
1134         }
1135     }
1136 
1137     if (hw->pcm_ops->run_buffer_out) {
1138         hw->pcm_ops->run_buffer_out(hw);
1139     }
1140 
1141     return clipped;
1142 }
1143 
1144 static void audio_run_out (AudioState *s)
1145 {
1146     HWVoiceOut *hw = NULL;
1147     SWVoiceOut *sw;
1148 
1149     while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
1150         size_t played, live, prev_rpos;
1151         size_t hw_free = audio_pcm_hw_get_free(hw);
1152         int nb_live;
1153 
1154         if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1155             /* there is exactly 1 sw for each hw with no mixeng */
1156             sw = hw->sw_head.lh_first;
1157 
1158             if (hw->pending_disable) {
1159                 hw->enabled = 0;
1160                 hw->pending_disable = 0;
1161                 if (hw->pcm_ops->enable_out) {
1162                     hw->pcm_ops->enable_out(hw, false);
1163                 }
1164             }
1165 
1166             if (sw->active) {
1167                 sw->callback.fn(sw->callback.opaque,
1168                                 hw_free * sw->info.bytes_per_frame);
1169             }
1170 
1171             if (hw->pcm_ops->run_buffer_out) {
1172                 hw->pcm_ops->run_buffer_out(hw);
1173             }
1174 
1175             continue;
1176         }
1177 
1178         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1179             if (sw->active) {
1180                 size_t sw_free = audio_get_free(sw);
1181                 size_t free;
1182 
1183                 if (hw_free > sw->total_hw_samples_mixed) {
1184                     free = audio_frontend_frames_out(sw,
1185                         MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
1186                 } else {
1187                     free = 0;
1188                 }
1189                 if (free > 0) {
1190                     sw->callback.fn(sw->callback.opaque,
1191                                     free * sw->info.bytes_per_frame);
1192                 }
1193             }
1194         }
1195 
1196         live = audio_pcm_hw_get_live_out (hw, &nb_live);
1197         if (!nb_live) {
1198             live = 0;
1199         }
1200 
1201         if (audio_bug(__func__, live > hw->mix_buf->size)) {
1202             dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
1203             continue;
1204         }
1205 
1206         if (hw->pending_disable && !nb_live) {
1207             SWVoiceCap *sc;
1208 #ifdef DEBUG_OUT
1209             dolog ("Disabling voice\n");
1210 #endif
1211             hw->enabled = 0;
1212             hw->pending_disable = 0;
1213             if (hw->pcm_ops->enable_out) {
1214                 hw->pcm_ops->enable_out(hw, false);
1215             }
1216             for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1217                 sc->sw.active = 0;
1218                 audio_recalc_and_notify_capture (sc->cap);
1219             }
1220             continue;
1221         }
1222 
1223         if (!live) {
1224             if (hw->pcm_ops->run_buffer_out) {
1225                 hw->pcm_ops->run_buffer_out(hw);
1226             }
1227             continue;
1228         }
1229 
1230         prev_rpos = hw->mix_buf->pos;
1231         played = audio_pcm_hw_run_out(hw, live);
1232         replay_audio_out(&played);
1233         if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
1234             dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
1235                   hw->mix_buf->pos, hw->mix_buf->size, played);
1236             hw->mix_buf->pos = 0;
1237         }
1238 
1239 #ifdef DEBUG_OUT
1240         dolog("played=%zu\n", played);
1241 #endif
1242 
1243         if (played) {
1244             hw->ts_helper += played;
1245             audio_capture_mix_and_clear (hw, prev_rpos, played);
1246         }
1247 
1248         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1249             if (!sw->active && sw->empty) {
1250                 continue;
1251             }
1252 
1253             if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
1254                 dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
1255                       played, sw->total_hw_samples_mixed);
1256                 played = sw->total_hw_samples_mixed;
1257             }
1258 
1259             sw->total_hw_samples_mixed -= played;
1260 
1261             if (!sw->total_hw_samples_mixed) {
1262                 sw->empty = 1;
1263             }
1264         }
1265     }
1266 }
1267 
1268 static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
1269 {
1270     size_t conv = 0;
1271 
1272     if (hw->pcm_ops->run_buffer_in) {
1273         hw->pcm_ops->run_buffer_in(hw);
1274     }
1275 
1276     while (samples) {
1277         size_t proc;
1278         size_t size = samples * hw->info.bytes_per_frame;
1279         void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
1280 
1281         assert(size % hw->info.bytes_per_frame == 0);
1282         if (size == 0) {
1283             break;
1284         }
1285 
1286         proc = audio_pcm_hw_conv_in(hw, buf, size / hw->info.bytes_per_frame);
1287 
1288         samples -= proc;
1289         conv += proc;
1290         hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
1291     }
1292 
1293     return conv;
1294 }
1295 
1296 static void audio_run_in (AudioState *s)
1297 {
1298     HWVoiceIn *hw = NULL;
1299 
1300     if (!audio_get_pdo_in(s->dev)->mixing_engine) {
1301         while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1302             /* there is exactly 1 sw for each hw with no mixeng */
1303             SWVoiceIn *sw = hw->sw_head.lh_first;
1304             if (sw->active) {
1305                 sw->callback.fn(sw->callback.opaque, INT_MAX);
1306             }
1307         }
1308         return;
1309     }
1310 
1311     while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1312         SWVoiceIn *sw;
1313         size_t captured = 0, min;
1314 
1315         if (replay_mode != REPLAY_MODE_PLAY) {
1316             captured = audio_pcm_hw_run_in(
1317                 hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
1318         }
1319         replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
1320                         hw->conv_buf->size);
1321 
1322         min = audio_pcm_hw_find_min_in (hw);
1323         hw->total_samples_captured += captured - min;
1324         hw->ts_helper += captured;
1325 
1326         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1327             sw->total_hw_samples_acquired -= min;
1328 
1329             if (sw->active) {
1330                 size_t sw_avail = audio_get_avail(sw);
1331                 size_t avail;
1332 
1333                 avail = audio_frontend_frames_in(sw, sw_avail);
1334                 if (avail > 0) {
1335                     sw->callback.fn(sw->callback.opaque,
1336                                     avail * sw->info.bytes_per_frame);
1337                 }
1338             }
1339         }
1340     }
1341 }
1342 
1343 static void audio_run_capture (AudioState *s)
1344 {
1345     CaptureVoiceOut *cap;
1346 
1347     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
1348         size_t live, rpos, captured;
1349         HWVoiceOut *hw = &cap->hw;
1350         SWVoiceOut *sw;
1351 
1352         captured = live = audio_pcm_hw_get_live_out (hw, NULL);
1353         rpos = hw->mix_buf->pos;
1354         while (live) {
1355             size_t left = hw->mix_buf->size - rpos;
1356             size_t to_capture = MIN(live, left);
1357             struct st_sample *src;
1358             struct capture_callback *cb;
1359 
1360             src = hw->mix_buf->samples + rpos;
1361             hw->clip (cap->buf, src, to_capture);
1362             mixeng_clear (src, to_capture);
1363 
1364             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1365                 cb->ops.capture (cb->opaque, cap->buf,
1366                                  to_capture * hw->info.bytes_per_frame);
1367             }
1368             rpos = (rpos + to_capture) % hw->mix_buf->size;
1369             live -= to_capture;
1370         }
1371         hw->mix_buf->pos = rpos;
1372 
1373         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1374             if (!sw->active && sw->empty) {
1375                 continue;
1376             }
1377 
1378             if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
1379                 dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
1380                       captured, sw->total_hw_samples_mixed);
1381                 captured = sw->total_hw_samples_mixed;
1382             }
1383 
1384             sw->total_hw_samples_mixed -= captured;
1385             sw->empty = sw->total_hw_samples_mixed == 0;
1386         }
1387     }
1388 }
1389 
1390 void audio_run(AudioState *s, const char *msg)
1391 {
1392     audio_run_out(s);
1393     audio_run_in(s);
1394     audio_run_capture(s);
1395 
1396 #ifdef DEBUG_POLL
1397     {
1398         static double prevtime;
1399         double currtime;
1400         struct timeval tv;
1401 
1402         if (gettimeofday (&tv, NULL)) {
1403             perror ("audio_run: gettimeofday");
1404             return;
1405         }
1406 
1407         currtime = tv.tv_sec + tv.tv_usec * 1e-6;
1408         dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
1409         prevtime = currtime;
1410     }
1411 #endif
1412 }
1413 
1414 void audio_generic_run_buffer_in(HWVoiceIn *hw)
1415 {
1416     if (unlikely(!hw->buf_emul)) {
1417         hw->size_emul = hw->samples * hw->info.bytes_per_frame;
1418         hw->buf_emul = g_malloc(hw->size_emul);
1419         hw->pos_emul = hw->pending_emul = 0;
1420     }
1421 
1422     while (hw->pending_emul < hw->size_emul) {
1423         size_t read_len = MIN(hw->size_emul - hw->pos_emul,
1424                               hw->size_emul - hw->pending_emul);
1425         size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
1426                                         read_len);
1427         hw->pending_emul += read;
1428         hw->pos_emul = (hw->pos_emul + read) % hw->size_emul;
1429         if (read < read_len) {
1430             break;
1431         }
1432     }
1433 }
1434 
1435 void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
1436 {
1437     size_t start;
1438 
1439     start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
1440     assert(start < hw->size_emul);
1441 
1442     *size = MIN(*size, hw->pending_emul);
1443     *size = MIN(*size, hw->size_emul - start);
1444     return hw->buf_emul + start;
1445 }
1446 
1447 void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
1448 {
1449     assert(size <= hw->pending_emul);
1450     hw->pending_emul -= size;
1451 }
1452 
1453 size_t audio_generic_buffer_get_free(HWVoiceOut *hw)
1454 {
1455     if (hw->buf_emul) {
1456         return hw->size_emul - hw->pending_emul;
1457     } else {
1458         return hw->samples * hw->info.bytes_per_frame;
1459     }
1460 }
1461 
1462 void audio_generic_run_buffer_out(HWVoiceOut *hw)
1463 {
1464     while (hw->pending_emul) {
1465         size_t write_len, written, start;
1466 
1467         start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
1468         assert(start < hw->size_emul);
1469 
1470         write_len = MIN(hw->pending_emul, hw->size_emul - start);
1471 
1472         written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
1473         hw->pending_emul -= written;
1474 
1475         if (written < write_len) {
1476             break;
1477         }
1478     }
1479 }
1480 
1481 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
1482 {
1483     if (unlikely(!hw->buf_emul)) {
1484         hw->size_emul = hw->samples * hw->info.bytes_per_frame;
1485         hw->buf_emul = g_malloc(hw->size_emul);
1486         hw->pos_emul = hw->pending_emul = 0;
1487     }
1488 
1489     *size = MIN(hw->size_emul - hw->pending_emul,
1490                 hw->size_emul - hw->pos_emul);
1491     return hw->buf_emul + hw->pos_emul;
1492 }
1493 
1494 size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
1495 {
1496     assert(buf == hw->buf_emul + hw->pos_emul &&
1497            size + hw->pending_emul <= hw->size_emul);
1498 
1499     hw->pending_emul += size;
1500     hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
1501 
1502     return size;
1503 }
1504 
1505 size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
1506 {
1507     size_t total = 0;
1508 
1509     if (hw->pcm_ops->buffer_get_free) {
1510         size_t free = hw->pcm_ops->buffer_get_free(hw);
1511 
1512         size = MIN(size, free);
1513     }
1514 
1515     while (total < size) {
1516         size_t dst_size = size - total;
1517         size_t copy_size, proc;
1518         void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
1519 
1520         if (dst_size == 0) {
1521             break;
1522         }
1523 
1524         copy_size = MIN(size - total, dst_size);
1525         if (dst) {
1526             memcpy(dst, (char *)buf + total, copy_size);
1527         }
1528         proc = hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
1529         total += proc;
1530 
1531         if (proc == 0 || proc < copy_size) {
1532             break;
1533         }
1534     }
1535 
1536     return total;
1537 }
1538 
1539 size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
1540 {
1541     size_t total = 0;
1542 
1543     if (hw->pcm_ops->run_buffer_in) {
1544         hw->pcm_ops->run_buffer_in(hw);
1545     }
1546 
1547     while (total < size) {
1548         size_t src_size = size - total;
1549         void *src = hw->pcm_ops->get_buffer_in(hw, &src_size);
1550 
1551         if (src_size == 0) {
1552             break;
1553         }
1554 
1555         memcpy((char *)buf + total, src, src_size);
1556         hw->pcm_ops->put_buffer_in(hw, src, src_size);
1557         total += src_size;
1558     }
1559 
1560     return total;
1561 }
1562 
1563 static int audio_driver_init(AudioState *s, struct audio_driver *drv,
1564                              bool msg, Audiodev *dev)
1565 {
1566     s->drv_opaque = drv->init(dev);
1567 
1568     if (s->drv_opaque) {
1569         if (!drv->pcm_ops->get_buffer_in) {
1570             drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
1571             drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
1572         }
1573         if (!drv->pcm_ops->get_buffer_out) {
1574             drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
1575             drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
1576         }
1577 
1578         audio_init_nb_voices_out(s, drv);
1579         audio_init_nb_voices_in(s, drv);
1580         s->drv = drv;
1581         return 0;
1582     } else {
1583         if (msg) {
1584             dolog("Could not init `%s' audio driver\n", drv->name);
1585         }
1586         return -1;
1587     }
1588 }
1589 
1590 static void audio_vm_change_state_handler (void *opaque, bool running,
1591                                            RunState state)
1592 {
1593     AudioState *s = opaque;
1594     HWVoiceOut *hwo = NULL;
1595     HWVoiceIn *hwi = NULL;
1596 
1597     s->vm_running = running;
1598     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
1599         if (hwo->pcm_ops->enable_out) {
1600             hwo->pcm_ops->enable_out(hwo, running);
1601         }
1602     }
1603 
1604     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
1605         if (hwi->pcm_ops->enable_in) {
1606             hwi->pcm_ops->enable_in(hwi, running);
1607         }
1608     }
1609     audio_reset_timer (s);
1610 }
1611 
1612 static void free_audio_state(AudioState *s)
1613 {
1614     HWVoiceOut *hwo, *hwon;
1615     HWVoiceIn *hwi, *hwin;
1616 
1617     QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
1618         SWVoiceCap *sc;
1619 
1620         if (hwo->enabled && hwo->pcm_ops->enable_out) {
1621             hwo->pcm_ops->enable_out(hwo, false);
1622         }
1623         hwo->pcm_ops->fini_out (hwo);
1624 
1625         for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1626             CaptureVoiceOut *cap = sc->cap;
1627             struct capture_callback *cb;
1628 
1629             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1630                 cb->ops.destroy (cb->opaque);
1631             }
1632         }
1633         QLIST_REMOVE(hwo, entries);
1634     }
1635 
1636     QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
1637         if (hwi->enabled && hwi->pcm_ops->enable_in) {
1638             hwi->pcm_ops->enable_in(hwi, false);
1639         }
1640         hwi->pcm_ops->fini_in (hwi);
1641         QLIST_REMOVE(hwi, entries);
1642     }
1643 
1644     if (s->drv) {
1645         s->drv->fini (s->drv_opaque);
1646         s->drv = NULL;
1647     }
1648 
1649     if (s->dev) {
1650         qapi_free_Audiodev(s->dev);
1651         s->dev = NULL;
1652     }
1653 
1654     if (s->ts) {
1655         timer_free(s->ts);
1656         s->ts = NULL;
1657     }
1658 
1659     g_free(s);
1660 }
1661 
1662 void audio_cleanup(void)
1663 {
1664     while (!QTAILQ_EMPTY(&audio_states)) {
1665         AudioState *s = QTAILQ_FIRST(&audio_states);
1666         QTAILQ_REMOVE(&audio_states, s, list);
1667         free_audio_state(s);
1668     }
1669 }
1670 
1671 static bool vmstate_audio_needed(void *opaque)
1672 {
1673     /*
1674      * Never needed, this vmstate only exists in case
1675      * an old qemu sends it to us.
1676      */
1677     return false;
1678 }
1679 
1680 static const VMStateDescription vmstate_audio = {
1681     .name = "audio",
1682     .version_id = 1,
1683     .minimum_version_id = 1,
1684     .needed = vmstate_audio_needed,
1685     .fields = (VMStateField[]) {
1686         VMSTATE_END_OF_LIST()
1687     }
1688 };
1689 
1690 static void audio_validate_opts(Audiodev *dev, Error **errp);
1691 
1692 static AudiodevListEntry *audiodev_find(
1693     AudiodevListHead *head, const char *drvname)
1694 {
1695     AudiodevListEntry *e;
1696     QSIMPLEQ_FOREACH(e, head, next) {
1697         if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) {
1698             return e;
1699         }
1700     }
1701 
1702     return NULL;
1703 }
1704 
1705 /*
1706  * if we have dev, this function was called because of an -audiodev argument =>
1707  *   initialize a new state with it
1708  * if dev == NULL => legacy implicit initialization, return the already created
1709  *   state or create a new one
1710  */
1711 static AudioState *audio_init(Audiodev *dev, const char *name)
1712 {
1713     static bool atexit_registered;
1714     size_t i;
1715     int done = 0;
1716     const char *drvname = NULL;
1717     VMChangeStateEntry *e;
1718     AudioState *s;
1719     struct audio_driver *driver;
1720     /* silence gcc warning about uninitialized variable */
1721     AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
1722 
1723     if (using_spice) {
1724         /*
1725          * When using spice allow the spice audio driver being picked
1726          * as default.
1727          *
1728          * Temporary hack.  Using audio devices without explicit
1729          * audiodev= property is already deprecated.  Same goes for
1730          * the -soundhw switch.  Once this support gets finally
1731          * removed we can also drop the concept of a default audio
1732          * backend and this can go away.
1733          */
1734         driver = audio_driver_lookup("spice");
1735         if (driver) {
1736             driver->can_be_default = 1;
1737         }
1738     }
1739 
1740     if (dev) {
1741         /* -audiodev option */
1742         legacy_config = false;
1743         drvname = AudiodevDriver_str(dev->driver);
1744     } else if (!QTAILQ_EMPTY(&audio_states)) {
1745         if (!legacy_config) {
1746             dolog("Device %s: audiodev default parameter is deprecated, please "
1747                   "specify audiodev=%s\n", name,
1748                   QTAILQ_FIRST(&audio_states)->dev->id);
1749         }
1750         return QTAILQ_FIRST(&audio_states);
1751     } else {
1752         /* legacy implicit initialization */
1753         head = audio_handle_legacy_opts();
1754         /*
1755          * In case of legacy initialization, all Audiodevs in the list will have
1756          * the same configuration (except the driver), so it doesn't matter which
1757          * one we chose.  We need an Audiodev to set up AudioState before we can
1758          * init a driver.  Also note that dev at this point is still in the
1759          * list.
1760          */
1761         dev = QSIMPLEQ_FIRST(&head)->dev;
1762         audio_validate_opts(dev, &error_abort);
1763     }
1764 
1765     s = g_new0(AudioState, 1);
1766     s->dev = dev;
1767 
1768     QLIST_INIT (&s->hw_head_out);
1769     QLIST_INIT (&s->hw_head_in);
1770     QLIST_INIT (&s->cap_head);
1771     if (!atexit_registered) {
1772         atexit(audio_cleanup);
1773         atexit_registered = true;
1774     }
1775 
1776     s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
1777 
1778     s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices;
1779     s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices;
1780 
1781     if (s->nb_hw_voices_out < 1) {
1782         dolog ("Bogus number of playback voices %d, setting to 1\n",
1783                s->nb_hw_voices_out);
1784         s->nb_hw_voices_out = 1;
1785     }
1786 
1787     if (s->nb_hw_voices_in < 0) {
1788         dolog ("Bogus number of capture voices %d, setting to 0\n",
1789                s->nb_hw_voices_in);
1790         s->nb_hw_voices_in = 0;
1791     }
1792 
1793     if (drvname) {
1794         driver = audio_driver_lookup(drvname);
1795         if (driver) {
1796             done = !audio_driver_init(s, driver, true, dev);
1797         } else {
1798             dolog ("Unknown audio driver `%s'\n", drvname);
1799         }
1800         if (!done) {
1801             free_audio_state(s);
1802             return NULL;
1803         }
1804     } else {
1805         for (i = 0; audio_prio_list[i]; i++) {
1806             AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]);
1807             driver = audio_driver_lookup(audio_prio_list[i]);
1808 
1809             if (e && driver) {
1810                 s->dev = dev = e->dev;
1811                 audio_validate_opts(dev, &error_abort);
1812                 done = !audio_driver_init(s, driver, false, dev);
1813                 if (done) {
1814                     e->dev = NULL;
1815                     break;
1816                 }
1817             }
1818         }
1819     }
1820     audio_free_audiodev_list(&head);
1821 
1822     if (!done) {
1823         driver = audio_driver_lookup("none");
1824         done = !audio_driver_init(s, driver, false, dev);
1825         assert(done);
1826         dolog("warning: Using timer based audio emulation\n");
1827     }
1828 
1829     if (dev->timer_period <= 0) {
1830         s->period_ticks = 1;
1831     } else {
1832         s->period_ticks = dev->timer_period * (int64_t)SCALE_US;
1833     }
1834 
1835     e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
1836     if (!e) {
1837         dolog ("warning: Could not register change state handler\n"
1838                "(Audio can continue looping even after stopping the VM)\n");
1839     }
1840 
1841     QTAILQ_INSERT_TAIL(&audio_states, s, list);
1842     QLIST_INIT (&s->card_head);
1843     vmstate_register (NULL, 0, &vmstate_audio, s);
1844     return s;
1845 }
1846 
1847 void audio_free_audiodev_list(AudiodevListHead *head)
1848 {
1849     AudiodevListEntry *e;
1850     while ((e = QSIMPLEQ_FIRST(head))) {
1851         QSIMPLEQ_REMOVE_HEAD(head, next);
1852         qapi_free_Audiodev(e->dev);
1853         g_free(e);
1854     }
1855 }
1856 
1857 void AUD_register_card (const char *name, QEMUSoundCard *card)
1858 {
1859     if (!card->state) {
1860         card->state = audio_init(NULL, name);
1861     }
1862 
1863     card->name = g_strdup (name);
1864     memset (&card->entries, 0, sizeof (card->entries));
1865     QLIST_INSERT_HEAD(&card->state->card_head, card, entries);
1866 }
1867 
1868 void AUD_remove_card (QEMUSoundCard *card)
1869 {
1870     QLIST_REMOVE (card, entries);
1871     g_free (card->name);
1872 }
1873 
1874 static struct audio_pcm_ops capture_pcm_ops;
1875 
1876 CaptureVoiceOut *AUD_add_capture(
1877     AudioState *s,
1878     struct audsettings *as,
1879     struct audio_capture_ops *ops,
1880     void *cb_opaque
1881     )
1882 {
1883     CaptureVoiceOut *cap;
1884     struct capture_callback *cb;
1885 
1886     if (!s) {
1887         if (!legacy_config) {
1888             dolog("Capturing without setting an audiodev is deprecated\n");
1889         }
1890         s = audio_init(NULL, NULL);
1891     }
1892 
1893     if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1894         dolog("Can't capture with mixeng disabled\n");
1895         return NULL;
1896     }
1897 
1898     if (audio_validate_settings (as)) {
1899         dolog ("Invalid settings were passed when trying to add capture\n");
1900         audio_print_settings (as);
1901         return NULL;
1902     }
1903 
1904     cb = g_malloc0(sizeof(*cb));
1905     cb->ops = *ops;
1906     cb->opaque = cb_opaque;
1907 
1908     cap = audio_pcm_capture_find_specific(s, as);
1909     if (cap) {
1910         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1911         return cap;
1912     } else {
1913         HWVoiceOut *hw;
1914         CaptureVoiceOut *cap;
1915 
1916         cap = g_malloc0(sizeof(*cap));
1917 
1918         hw = &cap->hw;
1919         hw->s = s;
1920         hw->pcm_ops = &capture_pcm_ops;
1921         QLIST_INIT (&hw->sw_head);
1922         QLIST_INIT (&cap->cb_head);
1923 
1924         /* XXX find a more elegant way */
1925         hw->samples = 4096 * 4;
1926         audio_pcm_hw_alloc_resources_out(hw);
1927 
1928         audio_pcm_init_info (&hw->info, as);
1929 
1930         cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
1931 
1932         if (hw->info.is_float) {
1933             hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
1934         } else {
1935             hw->clip = mixeng_clip
1936                 [hw->info.nchannels == 2]
1937                 [hw->info.is_signed]
1938                 [hw->info.swap_endianness]
1939                 [audio_bits_to_index(hw->info.bits)];
1940         }
1941 
1942         QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
1943         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1944 
1945         QLIST_FOREACH(hw, &s->hw_head_out, entries) {
1946             audio_attach_capture (hw);
1947         }
1948         return cap;
1949     }
1950 }
1951 
1952 void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
1953 {
1954     struct capture_callback *cb;
1955 
1956     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1957         if (cb->opaque == cb_opaque) {
1958             cb->ops.destroy (cb_opaque);
1959             QLIST_REMOVE (cb, entries);
1960             g_free (cb);
1961 
1962             if (!cap->cb_head.lh_first) {
1963                 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
1964 
1965                 while (sw) {
1966                     SWVoiceCap *sc = (SWVoiceCap *) sw;
1967 #ifdef DEBUG_CAPTURE
1968                     dolog ("freeing %s\n", sw->name);
1969 #endif
1970 
1971                     sw1 = sw->entries.le_next;
1972                     if (sw->rate) {
1973                         st_rate_stop (sw->rate);
1974                         sw->rate = NULL;
1975                     }
1976                     QLIST_REMOVE (sw, entries);
1977                     QLIST_REMOVE (sc, entries);
1978                     g_free (sc);
1979                     sw = sw1;
1980                 }
1981                 QLIST_REMOVE (cap, entries);
1982                 g_free (cap->hw.mix_buf);
1983                 g_free (cap->buf);
1984                 g_free (cap);
1985             }
1986             return;
1987         }
1988     }
1989 }
1990 
1991 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
1992 {
1993     Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1994     audio_set_volume_out(sw, &vol);
1995 }
1996 
1997 void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
1998 {
1999     if (sw) {
2000         HWVoiceOut *hw = sw->hw;
2001 
2002         sw->vol.mute = vol->mute;
2003         sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
2004         sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
2005             255;
2006 
2007         if (hw->pcm_ops->volume_out) {
2008             hw->pcm_ops->volume_out(hw, vol);
2009         }
2010     }
2011 }
2012 
2013 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
2014 {
2015     Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
2016     audio_set_volume_in(sw, &vol);
2017 }
2018 
2019 void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
2020 {
2021     if (sw) {
2022         HWVoiceIn *hw = sw->hw;
2023 
2024         sw->vol.mute = vol->mute;
2025         sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
2026         sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
2027             255;
2028 
2029         if (hw->pcm_ops->volume_in) {
2030             hw->pcm_ops->volume_in(hw, vol);
2031         }
2032     }
2033 }
2034 
2035 void audio_create_pdos(Audiodev *dev)
2036 {
2037     switch (dev->driver) {
2038 #define CASE(DRIVER, driver, pdo_name)                              \
2039     case AUDIODEV_DRIVER_##DRIVER:                                  \
2040         if (!dev->u.driver.in) {                                    \
2041             dev->u.driver.in = g_malloc0(                           \
2042                 sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
2043         }                                                           \
2044         if (!dev->u.driver.out) {                                   \
2045             dev->u.driver.out = g_malloc0(                          \
2046                 sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
2047         }                                                           \
2048         break
2049 
2050         CASE(NONE, none, );
2051 #ifdef CONFIG_AUDIO_ALSA
2052         CASE(ALSA, alsa, Alsa);
2053 #endif
2054 #ifdef CONFIG_AUDIO_COREAUDIO
2055         CASE(COREAUDIO, coreaudio, Coreaudio);
2056 #endif
2057 #ifdef CONFIG_DBUS_DISPLAY
2058         CASE(DBUS, dbus, );
2059 #endif
2060 #ifdef CONFIG_AUDIO_DSOUND
2061         CASE(DSOUND, dsound, );
2062 #endif
2063 #ifdef CONFIG_AUDIO_JACK
2064         CASE(JACK, jack, Jack);
2065 #endif
2066 #ifdef CONFIG_AUDIO_OSS
2067         CASE(OSS, oss, Oss);
2068 #endif
2069 #ifdef CONFIG_AUDIO_PA
2070         CASE(PA, pa, Pa);
2071 #endif
2072 #ifdef CONFIG_AUDIO_SDL
2073         CASE(SDL, sdl, Sdl);
2074 #endif
2075 #ifdef CONFIG_AUDIO_SNDIO
2076         CASE(SNDIO, sndio, );
2077 #endif
2078 #ifdef CONFIG_SPICE
2079         CASE(SPICE, spice, );
2080 #endif
2081         CASE(WAV, wav, );
2082 
2083     case AUDIODEV_DRIVER__MAX:
2084         abort();
2085     };
2086 }
2087 
2088 static void audio_validate_per_direction_opts(
2089     AudiodevPerDirectionOptions *pdo, Error **errp)
2090 {
2091     if (!pdo->has_mixing_engine) {
2092         pdo->has_mixing_engine = true;
2093         pdo->mixing_engine = true;
2094     }
2095     if (!pdo->has_fixed_settings) {
2096         pdo->has_fixed_settings = true;
2097         pdo->fixed_settings = pdo->mixing_engine;
2098     }
2099     if (!pdo->fixed_settings &&
2100         (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
2101         error_setg(errp,
2102                    "You can't use frequency, channels or format with fixed-settings=off");
2103         return;
2104     }
2105     if (!pdo->mixing_engine && pdo->fixed_settings) {
2106         error_setg(errp, "You can't use fixed-settings without mixeng");
2107         return;
2108     }
2109 
2110     if (!pdo->has_frequency) {
2111         pdo->has_frequency = true;
2112         pdo->frequency = 44100;
2113     }
2114     if (!pdo->has_channels) {
2115         pdo->has_channels = true;
2116         pdo->channels = 2;
2117     }
2118     if (!pdo->has_voices) {
2119         pdo->has_voices = true;
2120         pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
2121     }
2122     if (!pdo->has_format) {
2123         pdo->has_format = true;
2124         pdo->format = AUDIO_FORMAT_S16;
2125     }
2126 }
2127 
2128 static void audio_validate_opts(Audiodev *dev, Error **errp)
2129 {
2130     Error *err = NULL;
2131 
2132     audio_create_pdos(dev);
2133 
2134     audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err);
2135     if (err) {
2136         error_propagate(errp, err);
2137         return;
2138     }
2139 
2140     audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err);
2141     if (err) {
2142         error_propagate(errp, err);
2143         return;
2144     }
2145 
2146     if (!dev->has_timer_period) {
2147         dev->has_timer_period = true;
2148         dev->timer_period = 10000; /* 100Hz -> 10ms */
2149     }
2150 }
2151 
2152 void audio_help(void)
2153 {
2154     int i;
2155 
2156     printf("Available audio drivers:\n");
2157 
2158     for (i = 0; i < AUDIODEV_DRIVER__MAX; i++) {
2159         audio_driver *driver = audio_driver_lookup(AudiodevDriver_str(i));
2160         if (driver) {
2161             printf("%s\n", driver->name);
2162         }
2163     }
2164 }
2165 
2166 void audio_parse_option(const char *opt)
2167 {
2168     Audiodev *dev = NULL;
2169 
2170     if (is_help_option(opt)) {
2171         audio_help();
2172         exit(EXIT_SUCCESS);
2173     }
2174     Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
2175     visit_type_Audiodev(v, NULL, &dev, &error_fatal);
2176     visit_free(v);
2177 
2178     audio_define(dev);
2179 }
2180 
2181 void audio_define(Audiodev *dev)
2182 {
2183     AudiodevListEntry *e;
2184 
2185     audio_validate_opts(dev, &error_fatal);
2186 
2187     e = g_new0(AudiodevListEntry, 1);
2188     e->dev = dev;
2189     QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
2190 }
2191 
2192 bool audio_init_audiodevs(void)
2193 {
2194     AudiodevListEntry *e;
2195 
2196     QSIMPLEQ_FOREACH(e, &audiodevs, next) {
2197         if (!audio_init(e->dev, NULL)) {
2198             return false;
2199         }
2200     }
2201 
2202     return true;
2203 }
2204 
2205 audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
2206 {
2207     return (audsettings) {
2208         .freq = pdo->frequency,
2209         .nchannels = pdo->channels,
2210         .fmt = pdo->format,
2211         .endianness = AUDIO_HOST_ENDIANNESS,
2212     };
2213 }
2214 
2215 int audioformat_bytes_per_sample(AudioFormat fmt)
2216 {
2217     switch (fmt) {
2218     case AUDIO_FORMAT_U8:
2219     case AUDIO_FORMAT_S8:
2220         return 1;
2221 
2222     case AUDIO_FORMAT_U16:
2223     case AUDIO_FORMAT_S16:
2224         return 2;
2225 
2226     case AUDIO_FORMAT_U32:
2227     case AUDIO_FORMAT_S32:
2228     case AUDIO_FORMAT_F32:
2229         return 4;
2230 
2231     case AUDIO_FORMAT__MAX:
2232         ;
2233     }
2234     abort();
2235 }
2236 
2237 
2238 /* frames = freq * usec / 1e6 */
2239 int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
2240                         audsettings *as, int def_usecs)
2241 {
2242     uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs;
2243     return (as->freq * usecs + 500000) / 1000000;
2244 }
2245 
2246 /* samples = channels * frames = channels * freq * usec / 1e6 */
2247 int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
2248                          audsettings *as, int def_usecs)
2249 {
2250     return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
2251 }
2252 
2253 /*
2254  * bytes = bytes_per_sample * samples =
2255  *     bytes_per_sample * channels * freq * usec / 1e6
2256  */
2257 int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
2258                        audsettings *as, int def_usecs)
2259 {
2260     return audio_buffer_samples(pdo, as, def_usecs) *
2261         audioformat_bytes_per_sample(as->fmt);
2262 }
2263 
2264 AudioState *audio_state_by_name(const char *name)
2265 {
2266     AudioState *s;
2267     QTAILQ_FOREACH(s, &audio_states, list) {
2268         assert(s->dev);
2269         if (strcmp(name, s->dev->id) == 0) {
2270             return s;
2271         }
2272     }
2273     return NULL;
2274 }
2275 
2276 const char *audio_get_id(QEMUSoundCard *card)
2277 {
2278     if (card->state) {
2279         assert(card->state->dev);
2280         return card->state->dev->id;
2281     } else {
2282         return "";
2283     }
2284 }
2285 
2286 const char *audio_application_name(void)
2287 {
2288     const char *vm_name;
2289 
2290     vm_name = qemu_get_vm_name();
2291     return vm_name ? vm_name : "qemu";
2292 }
2293 
2294 void audio_rate_start(RateCtl *rate)
2295 {
2296     memset(rate, 0, sizeof(RateCtl));
2297     rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2298 }
2299 
2300 size_t audio_rate_peek_bytes(RateCtl *rate, struct audio_pcm_info *info)
2301 {
2302     int64_t now;
2303     int64_t ticks;
2304     int64_t bytes;
2305     int64_t frames;
2306 
2307     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2308     ticks = now - rate->start_ticks;
2309     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
2310     frames = (bytes - rate->bytes_sent) / info->bytes_per_frame;
2311     if (frames < 0 || frames > 65536) {
2312         AUD_log(NULL, "Resetting rate control (%" PRId64 " frames)\n", frames);
2313         audio_rate_start(rate);
2314         frames = 0;
2315     }
2316 
2317     return frames * info->bytes_per_frame;
2318 }
2319 
2320 void audio_rate_add_bytes(RateCtl *rate, size_t bytes_used)
2321 {
2322     rate->bytes_sent += bytes_used;
2323 }
2324 
2325 size_t audio_rate_get_bytes(RateCtl *rate, struct audio_pcm_info *info,
2326                             size_t bytes_avail)
2327 {
2328     size_t bytes;
2329 
2330     bytes = audio_rate_peek_bytes(rate, info);
2331     bytes = MIN(bytes, bytes_avail);
2332     audio_rate_add_bytes(rate, bytes);
2333 
2334     return bytes;
2335 }
2336 
2337 AudiodevList *qmp_query_audiodevs(Error **errp)
2338 {
2339     AudiodevList *ret = NULL;
2340     AudiodevListEntry *e;
2341     QSIMPLEQ_FOREACH(e, &audiodevs, next) {
2342         QAPI_LIST_PREPEND(ret, QAPI_CLONE(Audiodev, e->dev));
2343     }
2344     return ret;
2345 }
2346