Merge tag 'pull-request-2024-06-24' of https://gitlab.com/thuth/qemu into staging
[qemu/armbru.git] / audio / audio.c
blobaf0ae33fedb1e5407e66c2a51bb2f0881976de35
1 /*
2 * QEMU Audio subsystem
4 * Copyright (c) 2003-2005 Vassili Karpov (malc)
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:
13 * The above copyright notice and this permission notice shall be included in
14 * all copies or substantial portions of the Software.
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.
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 "qapi/qmp/qdict.h"
36 #include "qemu/cutils.h"
37 #include "qemu/error-report.h"
38 #include "qemu/log.h"
39 #include "qemu/module.h"
40 #include "qemu/help_option.h"
41 #include "sysemu/sysemu.h"
42 #include "sysemu/replay.h"
43 #include "sysemu/runstate.h"
44 #include "ui/qemu-spice.h"
45 #include "trace.h"
47 #define AUDIO_CAP "audio"
48 #include "audio_int.h"
50 /* #define DEBUG_LIVE */
51 /* #define DEBUG_OUT */
52 /* #define DEBUG_CAPTURE */
53 /* #define DEBUG_POLL */
55 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
58 /* Order of CONFIG_AUDIO_DRIVERS is import.
59 The 1st one is the one used by default, that is the reason
60 that we generate the list.
62 const char *audio_prio_list[] = {
63 "spice",
64 CONFIG_AUDIO_DRIVERS
65 "none",
66 NULL
69 static QLIST_HEAD(, audio_driver) audio_drivers;
70 static AudiodevListHead audiodevs =
71 QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
72 static AudiodevListHead default_audiodevs =
73 QSIMPLEQ_HEAD_INITIALIZER(default_audiodevs);
76 void audio_driver_register(audio_driver *drv)
78 QLIST_INSERT_HEAD(&audio_drivers, drv, next);
81 static audio_driver *audio_driver_lookup(const char *name)
83 struct audio_driver *d;
84 Error *local_err = NULL;
85 int rv;
87 QLIST_FOREACH(d, &audio_drivers, next) {
88 if (strcmp(name, d->name) == 0) {
89 return d;
92 rv = audio_module_load(name, &local_err);
93 if (rv > 0) {
94 QLIST_FOREACH(d, &audio_drivers, next) {
95 if (strcmp(name, d->name) == 0) {
96 return d;
99 } else if (rv < 0) {
100 error_report_err(local_err);
102 return NULL;
105 static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
106 QTAILQ_HEAD_INITIALIZER(audio_states);
107 static AudioState *default_audio_state;
109 const struct mixeng_volume nominal_volume = {
110 .mute = 0,
111 #ifdef FLOAT_MIXENG
112 .r = 1.0,
113 .l = 1.0,
114 #else
115 .r = 1ULL << 32,
116 .l = 1ULL << 32,
117 #endif
120 int audio_bug (const char *funcname, int cond)
122 if (cond) {
123 static int shown;
125 AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
126 if (!shown) {
127 shown = 1;
128 AUD_log (NULL, "Save all your work and restart without audio\n");
129 AUD_log (NULL, "I am sorry\n");
131 AUD_log (NULL, "Context:\n");
134 return cond;
137 static inline int audio_bits_to_index (int bits)
139 switch (bits) {
140 case 8:
141 return 0;
143 case 16:
144 return 1;
146 case 32:
147 return 2;
149 default:
150 audio_bug ("bits_to_index", 1);
151 AUD_log (NULL, "invalid bits %d\n", bits);
152 return 0;
156 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
158 if (cap) {
159 fprintf(stderr, "%s: ", cap);
162 vfprintf(stderr, fmt, ap);
165 void AUD_log (const char *cap, const char *fmt, ...)
167 va_list ap;
169 va_start (ap, fmt);
170 AUD_vlog (cap, fmt, ap);
171 va_end (ap);
174 static void audio_print_settings (struct audsettings *as)
176 dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
178 switch (as->fmt) {
179 case AUDIO_FORMAT_S8:
180 AUD_log (NULL, "S8");
181 break;
182 case AUDIO_FORMAT_U8:
183 AUD_log (NULL, "U8");
184 break;
185 case AUDIO_FORMAT_S16:
186 AUD_log (NULL, "S16");
187 break;
188 case AUDIO_FORMAT_U16:
189 AUD_log (NULL, "U16");
190 break;
191 case AUDIO_FORMAT_S32:
192 AUD_log (NULL, "S32");
193 break;
194 case AUDIO_FORMAT_U32:
195 AUD_log (NULL, "U32");
196 break;
197 case AUDIO_FORMAT_F32:
198 AUD_log (NULL, "F32");
199 break;
200 default:
201 AUD_log (NULL, "invalid(%d)", as->fmt);
202 break;
205 AUD_log (NULL, " endianness=");
206 switch (as->endianness) {
207 case 0:
208 AUD_log (NULL, "little");
209 break;
210 case 1:
211 AUD_log (NULL, "big");
212 break;
213 default:
214 AUD_log (NULL, "invalid");
215 break;
217 AUD_log (NULL, "\n");
220 static int audio_validate_settings (struct audsettings *as)
222 int invalid;
224 invalid = as->nchannels < 1;
225 invalid |= as->endianness != 0 && as->endianness != 1;
227 switch (as->fmt) {
228 case AUDIO_FORMAT_S8:
229 case AUDIO_FORMAT_U8:
230 case AUDIO_FORMAT_S16:
231 case AUDIO_FORMAT_U16:
232 case AUDIO_FORMAT_S32:
233 case AUDIO_FORMAT_U32:
234 case AUDIO_FORMAT_F32:
235 break;
236 default:
237 invalid = 1;
238 break;
241 invalid |= as->freq <= 0;
242 return invalid ? -1 : 0;
245 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
247 int bits = 8;
248 bool is_signed = false, is_float = false;
250 switch (as->fmt) {
251 case AUDIO_FORMAT_S8:
252 is_signed = true;
253 /* fall through */
254 case AUDIO_FORMAT_U8:
255 break;
257 case AUDIO_FORMAT_S16:
258 is_signed = true;
259 /* fall through */
260 case AUDIO_FORMAT_U16:
261 bits = 16;
262 break;
264 case AUDIO_FORMAT_F32:
265 is_float = true;
266 /* fall through */
267 case AUDIO_FORMAT_S32:
268 is_signed = true;
269 /* fall through */
270 case AUDIO_FORMAT_U32:
271 bits = 32;
272 break;
274 default:
275 abort();
277 return info->freq == as->freq
278 && info->nchannels == as->nchannels
279 && info->is_signed == is_signed
280 && info->is_float == is_float
281 && info->bits == bits
282 && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
285 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
287 int bits = 8, mul;
288 bool is_signed = false, is_float = false;
290 switch (as->fmt) {
291 case AUDIO_FORMAT_S8:
292 is_signed = true;
293 /* fall through */
294 case AUDIO_FORMAT_U8:
295 mul = 1;
296 break;
298 case AUDIO_FORMAT_S16:
299 is_signed = true;
300 /* fall through */
301 case AUDIO_FORMAT_U16:
302 bits = 16;
303 mul = 2;
304 break;
306 case AUDIO_FORMAT_F32:
307 is_float = true;
308 /* fall through */
309 case AUDIO_FORMAT_S32:
310 is_signed = true;
311 /* fall through */
312 case AUDIO_FORMAT_U32:
313 bits = 32;
314 mul = 4;
315 break;
317 default:
318 abort();
321 info->freq = as->freq;
322 info->bits = bits;
323 info->is_signed = is_signed;
324 info->is_float = is_float;
325 info->nchannels = as->nchannels;
326 info->bytes_per_frame = as->nchannels * mul;
327 info->bytes_per_second = info->freq * info->bytes_per_frame;
328 info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
331 void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
333 if (!len) {
334 return;
337 if (info->is_signed || info->is_float) {
338 memset(buf, 0x00, len * info->bytes_per_frame);
339 } else {
340 switch (info->bits) {
341 case 8:
342 memset(buf, 0x80, len * info->bytes_per_frame);
343 break;
345 case 16:
347 int i;
348 uint16_t *p = buf;
349 short s = INT16_MAX;
351 if (info->swap_endianness) {
352 s = bswap16 (s);
355 for (i = 0; i < len * info->nchannels; i++) {
356 p[i] = s;
359 break;
361 case 32:
363 int i;
364 uint32_t *p = buf;
365 int32_t s = INT32_MAX;
367 if (info->swap_endianness) {
368 s = bswap32 (s);
371 for (i = 0; i < len * info->nchannels; i++) {
372 p[i] = s;
375 break;
377 default:
378 AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
379 info->bits);
380 break;
386 * Capture
388 static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
389 struct audsettings *as)
391 CaptureVoiceOut *cap;
393 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
394 if (audio_pcm_info_eq (&cap->hw.info, as)) {
395 return cap;
398 return NULL;
401 static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
403 struct capture_callback *cb;
405 #ifdef DEBUG_CAPTURE
406 dolog ("notification %d sent\n", cmd);
407 #endif
408 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
409 cb->ops.notify (cb->opaque, cmd);
413 static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
415 if (cap->hw.enabled != enabled) {
416 audcnotification_e cmd;
417 cap->hw.enabled = enabled;
418 cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
419 audio_notify_capture (cap, cmd);
423 static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
425 HWVoiceOut *hw = &cap->hw;
426 SWVoiceOut *sw;
427 int enabled = 0;
429 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
430 if (sw->active) {
431 enabled = 1;
432 break;
435 audio_capture_maybe_changed (cap, enabled);
438 static void audio_detach_capture (HWVoiceOut *hw)
440 SWVoiceCap *sc = hw->cap_head.lh_first;
442 while (sc) {
443 SWVoiceCap *sc1 = sc->entries.le_next;
444 SWVoiceOut *sw = &sc->sw;
445 CaptureVoiceOut *cap = sc->cap;
446 int was_active = sw->active;
448 if (sw->rate) {
449 st_rate_stop (sw->rate);
450 sw->rate = NULL;
453 QLIST_REMOVE (sw, entries);
454 QLIST_REMOVE (sc, entries);
455 g_free (sc);
456 if (was_active) {
457 /* We have removed soft voice from the capture:
458 this might have changed the overall status of the capture
459 since this might have been the only active voice */
460 audio_recalc_and_notify_capture (cap);
462 sc = sc1;
466 static int audio_attach_capture (HWVoiceOut *hw)
468 AudioState *s = hw->s;
469 CaptureVoiceOut *cap;
471 audio_detach_capture (hw);
472 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
473 SWVoiceCap *sc;
474 SWVoiceOut *sw;
475 HWVoiceOut *hw_cap = &cap->hw;
477 sc = g_malloc0(sizeof(*sc));
479 sc->cap = cap;
480 sw = &sc->sw;
481 sw->hw = hw_cap;
482 sw->info = hw->info;
483 sw->empty = 1;
484 sw->active = hw->enabled;
485 sw->vol = nominal_volume;
486 sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
487 QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
488 QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
489 #ifdef DEBUG_CAPTURE
490 sw->name = g_strdup_printf ("for %p %d,%d,%d",
491 hw, sw->info.freq, sw->info.bits,
492 sw->info.nchannels);
493 dolog ("Added %s active = %d\n", sw->name, sw->active);
494 #endif
495 if (sw->active) {
496 audio_capture_maybe_changed (cap, 1);
499 return 0;
503 * Hard voice (capture)
505 static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
507 SWVoiceIn *sw;
508 size_t m = hw->total_samples_captured;
510 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
511 if (sw->active) {
512 m = MIN (m, sw->total_hw_samples_acquired);
515 return m;
518 static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
520 size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
521 if (audio_bug(__func__, live > hw->conv_buf.size)) {
522 dolog("live=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size);
523 return 0;
525 return live;
528 static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
530 size_t conv = 0;
531 STSampleBuffer *conv_buf = &hw->conv_buf;
533 while (samples) {
534 uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame);
535 size_t proc = MIN(samples, conv_buf->size - conv_buf->pos);
537 hw->conv(conv_buf->buffer + conv_buf->pos, src, proc);
538 conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
539 samples -= proc;
540 conv += proc;
543 return conv;
547 * Soft voice (capture)
549 static void audio_pcm_sw_resample_in(SWVoiceIn *sw,
550 size_t frames_in_max, size_t frames_out_max,
551 size_t *total_in, size_t *total_out)
553 HWVoiceIn *hw = sw->hw;
554 struct st_sample *src, *dst;
555 size_t live, rpos, frames_in, frames_out;
557 live = hw->total_samples_captured - sw->total_hw_samples_acquired;
558 rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
560 /* resample conv_buf from rpos to end of buffer */
561 src = hw->conv_buf.buffer + rpos;
562 frames_in = MIN(frames_in_max, hw->conv_buf.size - rpos);
563 dst = sw->resample_buf.buffer;
564 frames_out = frames_out_max;
565 st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
566 rpos += frames_in;
567 *total_in = frames_in;
568 *total_out = frames_out;
570 /* resample conv_buf from start of buffer if there are input frames left */
571 if (frames_in_max - frames_in && rpos == hw->conv_buf.size) {
572 src = hw->conv_buf.buffer;
573 frames_in = frames_in_max - frames_in;
574 dst += frames_out;
575 frames_out = frames_out_max - frames_out;
576 st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
577 *total_in += frames_in;
578 *total_out += frames_out;
582 static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len)
584 HWVoiceIn *hw = sw->hw;
585 size_t live, frames_out_max, total_in, total_out;
587 live = hw->total_samples_captured - sw->total_hw_samples_acquired;
588 if (!live) {
589 return 0;
591 if (audio_bug(__func__, live > hw->conv_buf.size)) {
592 dolog("live_in=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size);
593 return 0;
596 frames_out_max = MIN(buf_len / sw->info.bytes_per_frame,
597 sw->resample_buf.size);
599 audio_pcm_sw_resample_in(sw, live, frames_out_max, &total_in, &total_out);
601 if (!hw->pcm_ops->volume_in) {
602 mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol);
604 sw->clip(buf, sw->resample_buf.buffer, total_out);
606 sw->total_hw_samples_acquired += total_in;
607 return total_out * sw->info.bytes_per_frame;
611 * Hard voice (playback)
613 static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
615 SWVoiceOut *sw;
616 size_t m = SIZE_MAX;
617 int nb_live = 0;
619 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
620 if (sw->active || !sw->empty) {
621 m = MIN (m, sw->total_hw_samples_mixed);
622 nb_live += 1;
626 *nb_livep = nb_live;
627 return m;
630 static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
632 size_t smin;
633 int nb_live1;
635 smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
636 if (nb_live) {
637 *nb_live = nb_live1;
640 if (nb_live1) {
641 size_t live = smin;
643 if (audio_bug(__func__, live > hw->mix_buf.size)) {
644 dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
645 return 0;
647 return live;
649 return 0;
652 static size_t audio_pcm_hw_get_free(HWVoiceOut *hw)
654 return (hw->pcm_ops->buffer_get_free ? hw->pcm_ops->buffer_get_free(hw) :
655 INT_MAX) / hw->info.bytes_per_frame;
658 static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
660 size_t clipped = 0;
661 size_t pos = hw->mix_buf.pos;
663 while (len) {
664 st_sample *src = hw->mix_buf.buffer + pos;
665 uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
666 size_t samples_till_end_of_buf = hw->mix_buf.size - pos;
667 size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
669 hw->clip(dst, src, samples_to_clip);
671 pos = (pos + samples_to_clip) % hw->mix_buf.size;
672 len -= samples_to_clip;
673 clipped += samples_to_clip;
678 * Soft voice (playback)
680 static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
681 size_t frames_in_max, size_t frames_out_max,
682 size_t *total_in, size_t *total_out)
684 HWVoiceOut *hw = sw->hw;
685 struct st_sample *src, *dst;
686 size_t live, wpos, frames_in, frames_out;
688 live = sw->total_hw_samples_mixed;
689 wpos = (hw->mix_buf.pos + live) % hw->mix_buf.size;
691 /* write to mix_buf from wpos to end of buffer */
692 src = sw->resample_buf.buffer;
693 frames_in = frames_in_max;
694 dst = hw->mix_buf.buffer + wpos;
695 frames_out = MIN(frames_out_max, hw->mix_buf.size - wpos);
696 st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
697 wpos += frames_out;
698 *total_in = frames_in;
699 *total_out = frames_out;
701 /* write to mix_buf from start of buffer if there are input frames left */
702 if (frames_in_max - frames_in > 0 && wpos == hw->mix_buf.size) {
703 src += frames_in;
704 frames_in = frames_in_max - frames_in;
705 dst = hw->mix_buf.buffer;
706 frames_out = frames_out_max - frames_out;
707 st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
708 *total_in += frames_in;
709 *total_out += frames_out;
713 static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
715 HWVoiceOut *hw = sw->hw;
716 size_t live, dead, hw_free, sw_max, fe_max;
717 size_t frames_in_max, frames_out_max, total_in, total_out;
719 live = sw->total_hw_samples_mixed;
720 if (audio_bug(__func__, live > hw->mix_buf.size)) {
721 dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
722 return 0;
725 if (live == hw->mix_buf.size) {
726 #ifdef DEBUG_OUT
727 dolog ("%s is full %zu\n", sw->name, live);
728 #endif
729 return 0;
732 dead = hw->mix_buf.size - live;
733 hw_free = audio_pcm_hw_get_free(hw);
734 hw_free = hw_free > live ? hw_free - live : 0;
735 frames_out_max = MIN(dead, hw_free);
736 sw_max = st_rate_frames_in(sw->rate, frames_out_max);
737 fe_max = MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.pos,
738 sw->resample_buf.size);
739 frames_in_max = MIN(sw_max, fe_max);
741 if (!frames_in_max) {
742 return 0;
745 if (frames_in_max > sw->resample_buf.pos) {
746 sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos,
747 buf, frames_in_max - sw->resample_buf.pos);
748 if (!sw->hw->pcm_ops->volume_out) {
749 mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos,
750 frames_in_max - sw->resample_buf.pos, &sw->vol);
754 audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max,
755 &total_in, &total_out);
757 sw->total_hw_samples_mixed += total_out;
758 sw->empty = sw->total_hw_samples_mixed == 0;
761 * Upsampling may leave one audio frame in the resample buffer. Decrement
762 * total_in by one if there was a leftover frame from the previous resample
763 * pass in the resample buffer. Increment total_in by one if the current
764 * resample pass left one frame in the resample buffer.
766 if (frames_in_max - total_in == 1) {
767 /* copy one leftover audio frame to the beginning of the buffer */
768 *sw->resample_buf.buffer = *(sw->resample_buf.buffer + total_in);
769 total_in += 1 - sw->resample_buf.pos;
770 sw->resample_buf.pos = 1;
771 } else if (total_in >= sw->resample_buf.pos) {
772 total_in -= sw->resample_buf.pos;
773 sw->resample_buf.pos = 0;
776 #ifdef DEBUG_OUT
777 dolog (
778 "%s: write size %zu written %zu total mixed %zu\n",
779 SW_NAME(sw),
780 buf_len / sw->info.bytes_per_frame,
781 total_in,
782 sw->total_hw_samples_mixed
784 #endif
786 return total_in * sw->info.bytes_per_frame;
789 #ifdef DEBUG_AUDIO
790 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
792 dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
793 cap, info->bits, info->is_signed, info->is_float, info->freq,
794 info->nchannels);
796 #endif
798 #define DAC
799 #include "audio_template.h"
800 #undef DAC
801 #include "audio_template.h"
804 * Timer
806 static int audio_is_timer_needed(AudioState *s)
808 HWVoiceIn *hwi = NULL;
809 HWVoiceOut *hwo = NULL;
811 while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
812 if (!hwo->poll_mode) {
813 return 1;
816 while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
817 if (!hwi->poll_mode) {
818 return 1;
821 return 0;
824 static void audio_reset_timer (AudioState *s)
826 if (audio_is_timer_needed(s)) {
827 timer_mod_anticipate_ns(s->ts,
828 qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
829 if (!s->timer_running) {
830 s->timer_running = true;
831 s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
832 trace_audio_timer_start(s->period_ticks / SCALE_MS);
834 } else {
835 timer_del(s->ts);
836 if (s->timer_running) {
837 s->timer_running = false;
838 trace_audio_timer_stop();
843 static void audio_timer (void *opaque)
845 int64_t now, diff;
846 AudioState *s = opaque;
848 now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
849 diff = now - s->timer_last;
850 if (diff > s->period_ticks * 3 / 2) {
851 trace_audio_timer_delayed(diff / SCALE_MS);
853 s->timer_last = now;
855 audio_run(s, "timer");
856 audio_reset_timer(s);
860 * Public API
862 size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
864 HWVoiceOut *hw;
866 if (!sw) {
867 /* XXX: Consider options */
868 return size;
870 hw = sw->hw;
872 if (!hw->enabled) {
873 dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
874 return 0;
877 if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
878 return audio_pcm_sw_write(sw, buf, size);
879 } else {
880 return hw->pcm_ops->write(hw, buf, size);
884 size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
886 HWVoiceIn *hw;
888 if (!sw) {
889 /* XXX: Consider options */
890 return size;
892 hw = sw->hw;
894 if (!hw->enabled) {
895 dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
896 return 0;
899 if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
900 return audio_pcm_sw_read(sw, buf, size);
901 } else {
902 return hw->pcm_ops->read(hw, buf, size);
906 int AUD_get_buffer_size_out(SWVoiceOut *sw)
908 return sw->hw->samples * sw->hw->info.bytes_per_frame;
911 void AUD_set_active_out (SWVoiceOut *sw, int on)
913 HWVoiceOut *hw;
915 if (!sw) {
916 return;
919 hw = sw->hw;
920 if (sw->active != on) {
921 AudioState *s = sw->s;
922 SWVoiceOut *temp_sw;
923 SWVoiceCap *sc;
925 if (on) {
926 hw->pending_disable = 0;
927 if (!hw->enabled) {
928 hw->enabled = 1;
929 if (s->vm_running) {
930 if (hw->pcm_ops->enable_out) {
931 hw->pcm_ops->enable_out(hw, true);
933 audio_reset_timer (s);
936 } else {
937 if (hw->enabled) {
938 int nb_active = 0;
940 for (temp_sw = hw->sw_head.lh_first; temp_sw;
941 temp_sw = temp_sw->entries.le_next) {
942 nb_active += temp_sw->active != 0;
945 hw->pending_disable = nb_active == 1;
949 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
950 sc->sw.active = hw->enabled;
951 if (hw->enabled) {
952 audio_capture_maybe_changed (sc->cap, 1);
955 sw->active = on;
959 void AUD_set_active_in (SWVoiceIn *sw, int on)
961 HWVoiceIn *hw;
963 if (!sw) {
964 return;
967 hw = sw->hw;
968 if (sw->active != on) {
969 AudioState *s = sw->s;
970 SWVoiceIn *temp_sw;
972 if (on) {
973 if (!hw->enabled) {
974 hw->enabled = 1;
975 if (s->vm_running) {
976 if (hw->pcm_ops->enable_in) {
977 hw->pcm_ops->enable_in(hw, true);
979 audio_reset_timer (s);
982 sw->total_hw_samples_acquired = hw->total_samples_captured;
983 } else {
984 if (hw->enabled) {
985 int nb_active = 0;
987 for (temp_sw = hw->sw_head.lh_first; temp_sw;
988 temp_sw = temp_sw->entries.le_next) {
989 nb_active += temp_sw->active != 0;
992 if (nb_active == 1) {
993 hw->enabled = 0;
994 if (hw->pcm_ops->enable_in) {
995 hw->pcm_ops->enable_in(hw, false);
1000 sw->active = on;
1004 static size_t audio_get_avail (SWVoiceIn *sw)
1006 size_t live;
1008 if (!sw) {
1009 return 0;
1012 live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
1013 if (audio_bug(__func__, live > sw->hw->conv_buf.size)) {
1014 dolog("live=%zu sw->hw->conv_buf.size=%zu\n", live,
1015 sw->hw->conv_buf.size);
1016 return 0;
1019 ldebug (
1020 "%s: get_avail live %zu frontend frames %u\n",
1021 SW_NAME (sw),
1022 live, st_rate_frames_out(sw->rate, live)
1025 return live;
1028 static size_t audio_get_free(SWVoiceOut *sw)
1030 size_t live, dead;
1032 if (!sw) {
1033 return 0;
1036 live = sw->total_hw_samples_mixed;
1038 if (audio_bug(__func__, live > sw->hw->mix_buf.size)) {
1039 dolog("live=%zu sw->hw->mix_buf.size=%zu\n", live,
1040 sw->hw->mix_buf.size);
1041 return 0;
1044 dead = sw->hw->mix_buf.size - live;
1046 #ifdef DEBUG_OUT
1047 dolog("%s: get_free live %zu dead %zu frontend frames %u\n",
1048 SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead));
1049 #endif
1051 return dead;
1054 static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
1055 size_t samples)
1057 size_t n;
1059 if (hw->enabled) {
1060 SWVoiceCap *sc;
1062 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1063 SWVoiceOut *sw = &sc->sw;
1064 size_t rpos2 = rpos;
1066 n = samples;
1067 while (n) {
1068 size_t till_end_of_hw = hw->mix_buf.size - rpos2;
1069 size_t to_read = MIN(till_end_of_hw, n);
1070 size_t live, frames_in, frames_out;
1072 sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
1073 sw->resample_buf.size = to_read;
1074 live = sw->total_hw_samples_mixed;
1076 audio_pcm_sw_resample_out(sw,
1077 to_read, sw->hw->mix_buf.size - live,
1078 &frames_in, &frames_out);
1080 sw->total_hw_samples_mixed += frames_out;
1081 sw->empty = sw->total_hw_samples_mixed == 0;
1083 if (to_read - frames_in) {
1084 dolog("Could not mix %zu frames into a capture "
1085 "buffer, mixed %zu\n",
1086 to_read, frames_in);
1087 break;
1089 n -= to_read;
1090 rpos2 = (rpos2 + to_read) % hw->mix_buf.size;
1095 n = MIN(samples, hw->mix_buf.size - rpos);
1096 mixeng_clear(hw->mix_buf.buffer + rpos, n);
1097 mixeng_clear(hw->mix_buf.buffer, samples - n);
1100 static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
1102 size_t clipped = 0;
1104 while (live) {
1105 size_t size = live * hw->info.bytes_per_frame;
1106 size_t decr, proc;
1107 void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
1109 if (size == 0) {
1110 break;
1113 decr = MIN(size / hw->info.bytes_per_frame, live);
1114 if (buf) {
1115 audio_pcm_hw_clip_out(hw, buf, decr);
1117 proc = hw->pcm_ops->put_buffer_out(hw, buf,
1118 decr * hw->info.bytes_per_frame) /
1119 hw->info.bytes_per_frame;
1121 live -= proc;
1122 clipped += proc;
1123 hw->mix_buf.pos = (hw->mix_buf.pos + proc) % hw->mix_buf.size;
1125 if (proc == 0 || proc < decr) {
1126 break;
1130 if (hw->pcm_ops->run_buffer_out) {
1131 hw->pcm_ops->run_buffer_out(hw);
1134 return clipped;
1137 static void audio_run_out (AudioState *s)
1139 HWVoiceOut *hw = NULL;
1140 SWVoiceOut *sw;
1142 while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
1143 size_t played, live, prev_rpos;
1144 size_t hw_free = audio_pcm_hw_get_free(hw);
1145 int nb_live;
1147 if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1148 /* there is exactly 1 sw for each hw with no mixeng */
1149 sw = hw->sw_head.lh_first;
1151 if (hw->pending_disable) {
1152 hw->enabled = 0;
1153 hw->pending_disable = 0;
1154 if (hw->pcm_ops->enable_out) {
1155 hw->pcm_ops->enable_out(hw, false);
1159 if (sw->active) {
1160 sw->callback.fn(sw->callback.opaque,
1161 hw_free * sw->info.bytes_per_frame);
1164 if (hw->pcm_ops->run_buffer_out) {
1165 hw->pcm_ops->run_buffer_out(hw);
1168 continue;
1171 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1172 if (sw->active) {
1173 size_t sw_free = audio_get_free(sw);
1174 size_t free;
1176 if (hw_free > sw->total_hw_samples_mixed) {
1177 free = st_rate_frames_in(sw->rate,
1178 MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
1179 } else {
1180 free = 0;
1182 if (free > sw->resample_buf.pos) {
1183 free = MIN(free, sw->resample_buf.size)
1184 - sw->resample_buf.pos;
1185 sw->callback.fn(sw->callback.opaque,
1186 free * sw->info.bytes_per_frame);
1191 live = audio_pcm_hw_get_live_out (hw, &nb_live);
1192 if (!nb_live) {
1193 live = 0;
1196 if (audio_bug(__func__, live > hw->mix_buf.size)) {
1197 dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
1198 continue;
1201 if (hw->pending_disable && !nb_live) {
1202 SWVoiceCap *sc;
1203 #ifdef DEBUG_OUT
1204 dolog ("Disabling voice\n");
1205 #endif
1206 hw->enabled = 0;
1207 hw->pending_disable = 0;
1208 if (hw->pcm_ops->enable_out) {
1209 hw->pcm_ops->enable_out(hw, false);
1211 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1212 sc->sw.active = 0;
1213 audio_recalc_and_notify_capture (sc->cap);
1215 continue;
1218 if (!live) {
1219 if (hw->pcm_ops->run_buffer_out) {
1220 hw->pcm_ops->run_buffer_out(hw);
1222 continue;
1225 prev_rpos = hw->mix_buf.pos;
1226 played = audio_pcm_hw_run_out(hw, live);
1227 replay_audio_out(&played);
1228 if (audio_bug(__func__, hw->mix_buf.pos >= hw->mix_buf.size)) {
1229 dolog("hw->mix_buf.pos=%zu hw->mix_buf.size=%zu played=%zu\n",
1230 hw->mix_buf.pos, hw->mix_buf.size, played);
1231 hw->mix_buf.pos = 0;
1234 #ifdef DEBUG_OUT
1235 dolog("played=%zu\n", played);
1236 #endif
1238 if (played) {
1239 hw->ts_helper += played;
1240 audio_capture_mix_and_clear (hw, prev_rpos, played);
1243 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1244 if (!sw->active && sw->empty) {
1245 continue;
1248 if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
1249 dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
1250 played, sw->total_hw_samples_mixed);
1251 played = sw->total_hw_samples_mixed;
1254 sw->total_hw_samples_mixed -= played;
1256 if (!sw->total_hw_samples_mixed) {
1257 sw->empty = 1;
1263 static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
1265 size_t conv = 0;
1267 if (hw->pcm_ops->run_buffer_in) {
1268 hw->pcm_ops->run_buffer_in(hw);
1271 while (samples) {
1272 size_t proc;
1273 size_t size = samples * hw->info.bytes_per_frame;
1274 void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
1276 assert(size % hw->info.bytes_per_frame == 0);
1277 if (size == 0) {
1278 break;
1281 proc = audio_pcm_hw_conv_in(hw, buf, size / hw->info.bytes_per_frame);
1283 samples -= proc;
1284 conv += proc;
1285 hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
1288 return conv;
1291 static void audio_run_in (AudioState *s)
1293 HWVoiceIn *hw = NULL;
1295 if (!audio_get_pdo_in(s->dev)->mixing_engine) {
1296 while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1297 /* there is exactly 1 sw for each hw with no mixeng */
1298 SWVoiceIn *sw = hw->sw_head.lh_first;
1299 if (sw->active) {
1300 sw->callback.fn(sw->callback.opaque, INT_MAX);
1303 return;
1306 while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1307 SWVoiceIn *sw;
1308 size_t captured = 0, min;
1310 if (replay_mode != REPLAY_MODE_PLAY) {
1311 captured = audio_pcm_hw_run_in(
1312 hw, hw->conv_buf.size - audio_pcm_hw_get_live_in(hw));
1314 replay_audio_in(&captured, hw->conv_buf.buffer, &hw->conv_buf.pos,
1315 hw->conv_buf.size);
1317 min = audio_pcm_hw_find_min_in (hw);
1318 hw->total_samples_captured += captured - min;
1319 hw->ts_helper += captured;
1321 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1322 sw->total_hw_samples_acquired -= min;
1324 if (sw->active) {
1325 size_t sw_avail = audio_get_avail(sw);
1326 size_t avail;
1328 avail = st_rate_frames_out(sw->rate, sw_avail);
1329 if (avail > 0) {
1330 avail = MIN(avail, sw->resample_buf.size);
1331 sw->callback.fn(sw->callback.opaque,
1332 avail * sw->info.bytes_per_frame);
1339 static void audio_run_capture (AudioState *s)
1341 CaptureVoiceOut *cap;
1343 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
1344 size_t live, rpos, captured;
1345 HWVoiceOut *hw = &cap->hw;
1346 SWVoiceOut *sw;
1348 captured = live = audio_pcm_hw_get_live_out (hw, NULL);
1349 rpos = hw->mix_buf.pos;
1350 while (live) {
1351 size_t left = hw->mix_buf.size - rpos;
1352 size_t to_capture = MIN(live, left);
1353 struct st_sample *src;
1354 struct capture_callback *cb;
1356 src = hw->mix_buf.buffer + rpos;
1357 hw->clip (cap->buf, src, to_capture);
1358 mixeng_clear (src, to_capture);
1360 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1361 cb->ops.capture (cb->opaque, cap->buf,
1362 to_capture * hw->info.bytes_per_frame);
1364 rpos = (rpos + to_capture) % hw->mix_buf.size;
1365 live -= to_capture;
1367 hw->mix_buf.pos = rpos;
1369 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1370 if (!sw->active && sw->empty) {
1371 continue;
1374 if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
1375 dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
1376 captured, sw->total_hw_samples_mixed);
1377 captured = sw->total_hw_samples_mixed;
1380 sw->total_hw_samples_mixed -= captured;
1381 sw->empty = sw->total_hw_samples_mixed == 0;
1386 void audio_run(AudioState *s, const char *msg)
1388 audio_run_out(s);
1389 audio_run_in(s);
1390 audio_run_capture(s);
1392 #ifdef DEBUG_POLL
1394 static double prevtime;
1395 double currtime;
1396 struct timeval tv;
1398 if (gettimeofday (&tv, NULL)) {
1399 perror ("audio_run: gettimeofday");
1400 return;
1403 currtime = tv.tv_sec + tv.tv_usec * 1e-6;
1404 dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
1405 prevtime = currtime;
1407 #endif
1410 void audio_generic_run_buffer_in(HWVoiceIn *hw)
1412 if (unlikely(!hw->buf_emul)) {
1413 hw->size_emul = hw->samples * hw->info.bytes_per_frame;
1414 hw->buf_emul = g_malloc(hw->size_emul);
1415 hw->pos_emul = hw->pending_emul = 0;
1418 while (hw->pending_emul < hw->size_emul) {
1419 size_t read_len = MIN(hw->size_emul - hw->pos_emul,
1420 hw->size_emul - hw->pending_emul);
1421 size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
1422 read_len);
1423 hw->pending_emul += read;
1424 hw->pos_emul = (hw->pos_emul + read) % hw->size_emul;
1425 if (read < read_len) {
1426 break;
1431 void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
1433 size_t start;
1435 start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
1436 assert(start < hw->size_emul);
1438 *size = MIN(*size, hw->pending_emul);
1439 *size = MIN(*size, hw->size_emul - start);
1440 return hw->buf_emul + start;
1443 void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
1445 assert(size <= hw->pending_emul);
1446 hw->pending_emul -= size;
1449 size_t audio_generic_buffer_get_free(HWVoiceOut *hw)
1451 if (hw->buf_emul) {
1452 return hw->size_emul - hw->pending_emul;
1453 } else {
1454 return hw->samples * hw->info.bytes_per_frame;
1458 void audio_generic_run_buffer_out(HWVoiceOut *hw)
1460 while (hw->pending_emul) {
1461 size_t write_len, written, start;
1463 start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
1464 assert(start < hw->size_emul);
1466 write_len = MIN(hw->pending_emul, hw->size_emul - start);
1468 written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
1469 hw->pending_emul -= written;
1471 if (written < write_len) {
1472 break;
1477 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
1479 if (unlikely(!hw->buf_emul)) {
1480 hw->size_emul = hw->samples * hw->info.bytes_per_frame;
1481 hw->buf_emul = g_malloc(hw->size_emul);
1482 hw->pos_emul = hw->pending_emul = 0;
1485 *size = MIN(hw->size_emul - hw->pending_emul,
1486 hw->size_emul - hw->pos_emul);
1487 return hw->buf_emul + hw->pos_emul;
1490 size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
1492 assert(buf == hw->buf_emul + hw->pos_emul &&
1493 size + hw->pending_emul <= hw->size_emul);
1495 hw->pending_emul += size;
1496 hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
1498 return size;
1501 size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
1503 size_t total = 0;
1505 if (hw->pcm_ops->buffer_get_free) {
1506 size_t free = hw->pcm_ops->buffer_get_free(hw);
1508 size = MIN(size, free);
1511 while (total < size) {
1512 size_t dst_size = size - total;
1513 size_t copy_size, proc;
1514 void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
1516 if (dst_size == 0) {
1517 break;
1520 copy_size = MIN(size - total, dst_size);
1521 if (dst) {
1522 memcpy(dst, (char *)buf + total, copy_size);
1524 proc = hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
1525 total += proc;
1527 if (proc == 0 || proc < copy_size) {
1528 break;
1532 return total;
1535 size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
1537 size_t total = 0;
1539 if (hw->pcm_ops->run_buffer_in) {
1540 hw->pcm_ops->run_buffer_in(hw);
1543 while (total < size) {
1544 size_t src_size = size - total;
1545 void *src = hw->pcm_ops->get_buffer_in(hw, &src_size);
1547 if (src_size == 0) {
1548 break;
1551 memcpy((char *)buf + total, src, src_size);
1552 hw->pcm_ops->put_buffer_in(hw, src, src_size);
1553 total += src_size;
1556 return total;
1559 static int audio_driver_init(AudioState *s, struct audio_driver *drv,
1560 Audiodev *dev, Error **errp)
1562 Error *local_err = NULL;
1564 s->drv_opaque = drv->init(dev, &local_err);
1566 if (s->drv_opaque) {
1567 if (!drv->pcm_ops->get_buffer_in) {
1568 drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
1569 drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
1571 if (!drv->pcm_ops->get_buffer_out) {
1572 drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
1573 drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
1576 audio_init_nb_voices_out(s, drv, 1);
1577 audio_init_nb_voices_in(s, drv, 0);
1578 s->drv = drv;
1579 return 0;
1580 } else {
1581 if (local_err) {
1582 error_propagate(errp, local_err);
1583 } else {
1584 error_setg(errp, "Could not init `%s' audio driver", drv->name);
1586 return -1;
1590 static void audio_vm_change_state_handler (void *opaque, bool running,
1591 RunState state)
1593 AudioState *s = opaque;
1594 HWVoiceOut *hwo = NULL;
1595 HWVoiceIn *hwi = NULL;
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);
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);
1609 audio_reset_timer (s);
1612 static void free_audio_state(AudioState *s)
1614 HWVoiceOut *hwo, *hwon;
1615 HWVoiceIn *hwi, *hwin;
1617 QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
1618 SWVoiceCap *sc;
1620 if (hwo->enabled && hwo->pcm_ops->enable_out) {
1621 hwo->pcm_ops->enable_out(hwo, false);
1623 hwo->pcm_ops->fini_out (hwo);
1625 for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1626 CaptureVoiceOut *cap = sc->cap;
1627 struct capture_callback *cb;
1629 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1630 cb->ops.destroy (cb->opaque);
1633 QLIST_REMOVE(hwo, entries);
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);
1640 hwi->pcm_ops->fini_in (hwi);
1641 QLIST_REMOVE(hwi, entries);
1644 if (s->drv) {
1645 s->drv->fini (s->drv_opaque);
1646 s->drv = NULL;
1649 if (s->dev) {
1650 qapi_free_Audiodev(s->dev);
1651 s->dev = NULL;
1654 if (s->ts) {
1655 timer_free(s->ts);
1656 s->ts = NULL;
1659 g_free(s);
1662 void audio_cleanup(void)
1664 default_audio_state = NULL;
1665 while (!QTAILQ_EMPTY(&audio_states)) {
1666 AudioState *s = QTAILQ_FIRST(&audio_states);
1667 QTAILQ_REMOVE(&audio_states, s, list);
1668 free_audio_state(s);
1672 static bool vmstate_audio_needed(void *opaque)
1675 * Never needed, this vmstate only exists in case
1676 * an old qemu sends it to us.
1678 return false;
1681 static const VMStateDescription vmstate_audio = {
1682 .name = "audio",
1683 .version_id = 1,
1684 .minimum_version_id = 1,
1685 .needed = vmstate_audio_needed,
1686 .fields = (const VMStateField[]) {
1687 VMSTATE_END_OF_LIST()
1691 void audio_create_default_audiodevs(void)
1693 for (int i = 0; audio_prio_list[i]; i++) {
1694 if (audio_driver_lookup(audio_prio_list[i])) {
1695 QDict *dict = qdict_new();
1696 Audiodev *dev = NULL;
1697 Visitor *v;
1699 qdict_put_str(dict, "driver", audio_prio_list[i]);
1700 qdict_put_str(dict, "id", "#default");
1702 v = qobject_input_visitor_new_keyval(QOBJECT(dict));
1703 qobject_unref(dict);
1704 visit_type_Audiodev(v, NULL, &dev, &error_fatal);
1705 visit_free(v);
1707 audio_define_default(dev, &error_abort);
1713 * if we have dev, this function was called because of an -audiodev argument =>
1714 * initialize a new state with it
1715 * if dev == NULL => legacy implicit initialization, return the already created
1716 * state or create a new one
1718 static AudioState *audio_init(Audiodev *dev, Error **errp)
1720 static bool atexit_registered;
1721 int done = 0;
1722 const char *drvname;
1723 VMChangeStateEntry *vmse;
1724 AudioState *s;
1725 struct audio_driver *driver;
1727 s = g_new0(AudioState, 1);
1729 QLIST_INIT (&s->hw_head_out);
1730 QLIST_INIT (&s->hw_head_in);
1731 QLIST_INIT (&s->cap_head);
1732 if (!atexit_registered) {
1733 atexit(audio_cleanup);
1734 atexit_registered = true;
1737 s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
1739 if (dev) {
1740 /* -audiodev option */
1741 s->dev = dev;
1742 drvname = AudiodevDriver_str(dev->driver);
1743 driver = audio_driver_lookup(drvname);
1744 if (driver) {
1745 done = !audio_driver_init(s, driver, dev, errp);
1746 } else {
1747 error_setg(errp, "Unknown audio driver `%s'", drvname);
1749 if (!done) {
1750 goto out;
1752 } else {
1753 assert(!default_audio_state);
1754 for (;;) {
1755 AudiodevListEntry *e = QSIMPLEQ_FIRST(&default_audiodevs);
1756 if (!e) {
1757 error_setg(errp, "no default audio driver available");
1758 goto out;
1760 s->dev = dev = e->dev;
1761 QSIMPLEQ_REMOVE_HEAD(&default_audiodevs, next);
1762 g_free(e);
1763 drvname = AudiodevDriver_str(dev->driver);
1764 driver = audio_driver_lookup(drvname);
1765 if (!audio_driver_init(s, driver, dev, NULL)) {
1766 break;
1768 qapi_free_Audiodev(dev);
1769 s->dev = NULL;
1773 if (dev->timer_period <= 0) {
1774 s->period_ticks = 1;
1775 } else {
1776 s->period_ticks = dev->timer_period * (int64_t)SCALE_US;
1779 vmse = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
1780 if (!vmse) {
1781 dolog ("warning: Could not register change state handler\n"
1782 "(Audio can continue looping even after stopping the VM)\n");
1785 QTAILQ_INSERT_TAIL(&audio_states, s, list);
1786 QLIST_INIT (&s->card_head);
1787 vmstate_register_any(NULL, &vmstate_audio, s);
1788 return s;
1790 out:
1791 free_audio_state(s);
1792 return NULL;
1795 AudioState *audio_get_default_audio_state(Error **errp)
1797 if (!default_audio_state) {
1798 default_audio_state = audio_init(NULL, errp);
1799 if (!default_audio_state) {
1800 if (!QSIMPLEQ_EMPTY(&audiodevs)) {
1801 error_append_hint(errp, "Perhaps you wanted to use -audio or set audiodev=%s?\n",
1802 QSIMPLEQ_FIRST(&audiodevs)->dev->id);
1807 return default_audio_state;
1810 bool AUD_register_card (const char *name, QEMUSoundCard *card, Error **errp)
1812 if (!card->state) {
1813 card->state = audio_get_default_audio_state(errp);
1814 if (!card->state) {
1815 return false;
1819 card->name = g_strdup (name);
1820 memset (&card->entries, 0, sizeof (card->entries));
1821 QLIST_INSERT_HEAD(&card->state->card_head, card, entries);
1823 return true;
1826 void AUD_remove_card (QEMUSoundCard *card)
1828 QLIST_REMOVE (card, entries);
1829 g_free (card->name);
1832 static struct audio_pcm_ops capture_pcm_ops;
1834 CaptureVoiceOut *AUD_add_capture(
1835 AudioState *s,
1836 struct audsettings *as,
1837 struct audio_capture_ops *ops,
1838 void *cb_opaque
1841 CaptureVoiceOut *cap;
1842 struct capture_callback *cb;
1844 if (!s) {
1845 error_report("Capturing without setting an audiodev is not supported");
1846 abort();
1849 if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1850 dolog("Can't capture with mixeng disabled\n");
1851 return NULL;
1854 if (audio_validate_settings (as)) {
1855 dolog ("Invalid settings were passed when trying to add capture\n");
1856 audio_print_settings (as);
1857 return NULL;
1860 cb = g_malloc0(sizeof(*cb));
1861 cb->ops = *ops;
1862 cb->opaque = cb_opaque;
1864 cap = audio_pcm_capture_find_specific(s, as);
1865 if (cap) {
1866 QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1867 } else {
1868 HWVoiceOut *hw;
1870 cap = g_malloc0(sizeof(*cap));
1872 hw = &cap->hw;
1873 hw->s = s;
1874 hw->pcm_ops = &capture_pcm_ops;
1875 QLIST_INIT (&hw->sw_head);
1876 QLIST_INIT (&cap->cb_head);
1878 /* XXX find a more elegant way */
1879 hw->samples = 4096 * 4;
1880 audio_pcm_hw_alloc_resources_out(hw);
1882 audio_pcm_init_info (&hw->info, as);
1884 cap->buf = g_malloc0_n(hw->mix_buf.size, hw->info.bytes_per_frame);
1886 if (hw->info.is_float) {
1887 hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
1888 } else {
1889 hw->clip = mixeng_clip
1890 [hw->info.nchannels == 2]
1891 [hw->info.is_signed]
1892 [hw->info.swap_endianness]
1893 [audio_bits_to_index(hw->info.bits)];
1896 QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
1897 QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1899 QLIST_FOREACH(hw, &s->hw_head_out, entries) {
1900 audio_attach_capture (hw);
1904 return cap;
1907 void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
1909 struct capture_callback *cb;
1911 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1912 if (cb->opaque == cb_opaque) {
1913 cb->ops.destroy (cb_opaque);
1914 QLIST_REMOVE (cb, entries);
1915 g_free (cb);
1917 if (!cap->cb_head.lh_first) {
1918 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
1920 while (sw) {
1921 SWVoiceCap *sc = (SWVoiceCap *) sw;
1922 #ifdef DEBUG_CAPTURE
1923 dolog ("freeing %s\n", sw->name);
1924 #endif
1926 sw1 = sw->entries.le_next;
1927 if (sw->rate) {
1928 st_rate_stop (sw->rate);
1929 sw->rate = NULL;
1931 QLIST_REMOVE (sw, entries);
1932 QLIST_REMOVE (sc, entries);
1933 g_free (sc);
1934 sw = sw1;
1936 QLIST_REMOVE (cap, entries);
1937 g_free(cap->hw.mix_buf.buffer);
1938 g_free (cap->buf);
1939 g_free (cap);
1941 return;
1946 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
1948 Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1949 audio_set_volume_out(sw, &vol);
1952 void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
1954 if (sw) {
1955 HWVoiceOut *hw = sw->hw;
1957 sw->vol.mute = vol->mute;
1958 sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
1959 sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
1960 255;
1962 if (hw->pcm_ops->volume_out) {
1963 hw->pcm_ops->volume_out(hw, vol);
1968 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
1970 Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1971 audio_set_volume_in(sw, &vol);
1974 void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
1976 if (sw) {
1977 HWVoiceIn *hw = sw->hw;
1979 sw->vol.mute = vol->mute;
1980 sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
1981 sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
1982 255;
1984 if (hw->pcm_ops->volume_in) {
1985 hw->pcm_ops->volume_in(hw, vol);
1990 void audio_create_pdos(Audiodev *dev)
1992 switch (dev->driver) {
1993 #define CASE(DRIVER, driver, pdo_name) \
1994 case AUDIODEV_DRIVER_##DRIVER: \
1995 if (!dev->u.driver.in) { \
1996 dev->u.driver.in = g_malloc0( \
1997 sizeof(Audiodev##pdo_name##PerDirectionOptions)); \
1999 if (!dev->u.driver.out) { \
2000 dev->u.driver.out = g_malloc0( \
2001 sizeof(Audiodev##pdo_name##PerDirectionOptions)); \
2003 break
2005 CASE(NONE, none, );
2006 #ifdef CONFIG_AUDIO_ALSA
2007 CASE(ALSA, alsa, Alsa);
2008 #endif
2009 #ifdef CONFIG_AUDIO_COREAUDIO
2010 CASE(COREAUDIO, coreaudio, Coreaudio);
2011 #endif
2012 #ifdef CONFIG_DBUS_DISPLAY
2013 CASE(DBUS, dbus, );
2014 #endif
2015 #ifdef CONFIG_AUDIO_DSOUND
2016 CASE(DSOUND, dsound, );
2017 #endif
2018 #ifdef CONFIG_AUDIO_JACK
2019 CASE(JACK, jack, Jack);
2020 #endif
2021 #ifdef CONFIG_AUDIO_OSS
2022 CASE(OSS, oss, Oss);
2023 #endif
2024 #ifdef CONFIG_AUDIO_PA
2025 CASE(PA, pa, Pa);
2026 #endif
2027 #ifdef CONFIG_AUDIO_PIPEWIRE
2028 CASE(PIPEWIRE, pipewire, Pipewire);
2029 #endif
2030 #ifdef CONFIG_AUDIO_SDL
2031 CASE(SDL, sdl, Sdl);
2032 #endif
2033 #ifdef CONFIG_AUDIO_SNDIO
2034 CASE(SNDIO, sndio, );
2035 #endif
2036 #ifdef CONFIG_SPICE
2037 CASE(SPICE, spice, );
2038 #endif
2039 CASE(WAV, wav, );
2041 case AUDIODEV_DRIVER__MAX:
2042 abort();
2046 static void audio_validate_per_direction_opts(
2047 AudiodevPerDirectionOptions *pdo, Error **errp)
2049 if (!pdo->has_mixing_engine) {
2050 pdo->has_mixing_engine = true;
2051 pdo->mixing_engine = true;
2053 if (!pdo->has_fixed_settings) {
2054 pdo->has_fixed_settings = true;
2055 pdo->fixed_settings = pdo->mixing_engine;
2057 if (!pdo->fixed_settings &&
2058 (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
2059 error_setg(errp,
2060 "You can't use frequency, channels or format with fixed-settings=off");
2061 return;
2063 if (!pdo->mixing_engine && pdo->fixed_settings) {
2064 error_setg(errp, "You can't use fixed-settings without mixeng");
2065 return;
2068 if (!pdo->has_frequency) {
2069 pdo->has_frequency = true;
2070 pdo->frequency = 44100;
2072 if (!pdo->has_channels) {
2073 pdo->has_channels = true;
2074 pdo->channels = 2;
2076 if (!pdo->has_voices) {
2077 pdo->has_voices = true;
2078 pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
2080 if (!pdo->has_format) {
2081 pdo->has_format = true;
2082 pdo->format = AUDIO_FORMAT_S16;
2086 static void audio_validate_opts(Audiodev *dev, Error **errp)
2088 Error *err = NULL;
2090 audio_create_pdos(dev);
2092 audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err);
2093 if (err) {
2094 error_propagate(errp, err);
2095 return;
2098 audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err);
2099 if (err) {
2100 error_propagate(errp, err);
2101 return;
2104 if (!dev->has_timer_period) {
2105 dev->has_timer_period = true;
2106 dev->timer_period = 10000; /* 100Hz -> 10ms */
2110 void audio_help(void)
2112 int i;
2114 printf("Available audio drivers:\n");
2116 for (i = 0; i < AUDIODEV_DRIVER__MAX; i++) {
2117 audio_driver *driver = audio_driver_lookup(AudiodevDriver_str(i));
2118 if (driver) {
2119 printf("%s\n", driver->name);
2124 void audio_parse_option(const char *opt)
2126 Audiodev *dev = NULL;
2128 if (is_help_option(opt)) {
2129 audio_help();
2130 exit(EXIT_SUCCESS);
2132 Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
2133 visit_type_Audiodev(v, NULL, &dev, &error_fatal);
2134 visit_free(v);
2136 audio_define(dev);
2139 void audio_define(Audiodev *dev)
2141 AudiodevListEntry *e;
2143 audio_validate_opts(dev, &error_fatal);
2145 e = g_new0(AudiodevListEntry, 1);
2146 e->dev = dev;
2147 QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
2150 void audio_define_default(Audiodev *dev, Error **errp)
2152 AudiodevListEntry *e;
2154 audio_validate_opts(dev, errp);
2156 e = g_new0(AudiodevListEntry, 1);
2157 e->dev = dev;
2158 QSIMPLEQ_INSERT_TAIL(&default_audiodevs, e, next);
2161 void audio_init_audiodevs(void)
2163 AudiodevListEntry *e;
2165 QSIMPLEQ_FOREACH(e, &audiodevs, next) {
2166 audio_init(e->dev, &error_fatal);
2170 audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
2172 return (audsettings) {
2173 .freq = pdo->frequency,
2174 .nchannels = pdo->channels,
2175 .fmt = pdo->format,
2176 .endianness = AUDIO_HOST_ENDIANNESS,
2180 int audioformat_bytes_per_sample(AudioFormat fmt)
2182 switch (fmt) {
2183 case AUDIO_FORMAT_U8:
2184 case AUDIO_FORMAT_S8:
2185 return 1;
2187 case AUDIO_FORMAT_U16:
2188 case AUDIO_FORMAT_S16:
2189 return 2;
2191 case AUDIO_FORMAT_U32:
2192 case AUDIO_FORMAT_S32:
2193 case AUDIO_FORMAT_F32:
2194 return 4;
2196 case AUDIO_FORMAT__MAX:
2199 abort();
2203 /* frames = freq * usec / 1e6 */
2204 int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
2205 audsettings *as, int def_usecs)
2207 uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs;
2208 return (as->freq * usecs + 500000) / 1000000;
2211 /* samples = channels * frames = channels * freq * usec / 1e6 */
2212 int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
2213 audsettings *as, int def_usecs)
2215 return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
2219 * bytes = bytes_per_sample * samples =
2220 * bytes_per_sample * channels * freq * usec / 1e6
2222 int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
2223 audsettings *as, int def_usecs)
2225 return audio_buffer_samples(pdo, as, def_usecs) *
2226 audioformat_bytes_per_sample(as->fmt);
2229 AudioState *audio_state_by_name(const char *name, Error **errp)
2231 AudioState *s;
2232 QTAILQ_FOREACH(s, &audio_states, list) {
2233 assert(s->dev);
2234 if (strcmp(name, s->dev->id) == 0) {
2235 return s;
2238 error_setg(errp, "audiodev '%s' not found", name);
2239 return NULL;
2242 const char *audio_get_id(QEMUSoundCard *card)
2244 if (card->state) {
2245 assert(card->state->dev);
2246 return card->state->dev->id;
2247 } else {
2248 return "";
2252 const char *audio_application_name(void)
2254 const char *vm_name;
2256 vm_name = qemu_get_vm_name();
2257 return vm_name ? vm_name : "qemu";
2260 void audio_rate_start(RateCtl *rate)
2262 memset(rate, 0, sizeof(RateCtl));
2263 rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2266 size_t audio_rate_peek_bytes(RateCtl *rate, struct audio_pcm_info *info)
2268 int64_t now;
2269 int64_t ticks;
2270 int64_t bytes;
2271 int64_t frames;
2273 now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2274 ticks = now - rate->start_ticks;
2275 bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
2276 frames = (bytes - rate->bytes_sent) / info->bytes_per_frame;
2277 if (frames < 0 || frames > 65536) {
2278 AUD_log(NULL, "Resetting rate control (%" PRId64 " frames)\n", frames);
2279 audio_rate_start(rate);
2280 frames = 0;
2283 return frames * info->bytes_per_frame;
2286 void audio_rate_add_bytes(RateCtl *rate, size_t bytes_used)
2288 rate->bytes_sent += bytes_used;
2291 size_t audio_rate_get_bytes(RateCtl *rate, struct audio_pcm_info *info,
2292 size_t bytes_avail)
2294 size_t bytes;
2296 bytes = audio_rate_peek_bytes(rate, info);
2297 bytes = MIN(bytes, bytes_avail);
2298 audio_rate_add_bytes(rate, bytes);
2300 return bytes;
2303 AudiodevList *qmp_query_audiodevs(Error **errp)
2305 AudiodevList *ret = NULL;
2306 AudiodevListEntry *e;
2307 QSIMPLEQ_FOREACH(e, &audiodevs, next) {
2308 QAPI_LIST_PREPEND(ret, QAPI_CLONE(Audiodev, e->dev));
2310 return ret;