USB: use opaque parameter passing for monitor handle
[qemu/aliguori-queue.git] / audio / audio.c
blobc067afb2838c1169d08b10b59419652327e6885c
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.
24 #include "hw/hw.h"
25 #include "audio.h"
26 #include "monitor.h"
27 #include "qemu-timer.h"
28 #include "sysemu.h"
30 #define AUDIO_CAP "audio"
31 #include "audio_int.h"
33 /* #define DEBUG_PLIVE */
34 /* #define DEBUG_LIVE */
35 /* #define DEBUG_OUT */
36 /* #define DEBUG_CAPTURE */
38 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
41 /* Order of CONFIG_AUDIO_DRIVERS is import.
42 The 1st one is the one used by default, that is the reason
43 that we generate the list.
45 static struct audio_driver *drvtab[] = {
46 CONFIG_AUDIO_DRIVERS
47 &no_audio_driver,
48 &wav_audio_driver
51 struct fixed_settings {
52 int enabled;
53 int nb_voices;
54 int greedy;
55 struct audsettings settings;
58 static struct {
59 struct fixed_settings fixed_out;
60 struct fixed_settings fixed_in;
61 union {
62 int hertz;
63 int64_t ticks;
64 } period;
65 int plive;
66 int log_to_monitor;
67 } conf = {
68 .fixed_out = { /* DAC fixed settings */
69 .enabled = 1,
70 .nb_voices = 1,
71 .greedy = 1,
72 .settings = {
73 .freq = 44100,
74 .nchannels = 2,
75 .fmt = AUD_FMT_S16,
76 .endianness = AUDIO_HOST_ENDIANNESS,
80 .fixed_in = { /* ADC fixed settings */
81 .enabled = 1,
82 .nb_voices = 1,
83 .greedy = 1,
84 .settings = {
85 .freq = 44100,
86 .nchannels = 2,
87 .fmt = AUD_FMT_S16,
88 .endianness = AUDIO_HOST_ENDIANNESS,
92 .period = { .hertz = 250 },
93 .plive = 0,
94 .log_to_monitor = 0,
97 static AudioState glob_audio_state;
99 struct mixeng_volume nominal_volume = {
100 .mute = 0,
101 #ifdef FLOAT_MIXENG
102 .r = 1.0,
103 .l = 1.0,
104 #else
105 .r = 1ULL << 32,
106 .l = 1ULL << 32,
107 #endif
110 /* http://www.df.lth.se/~john_e/gems/gem002d.html */
111 /* http://www.multi-platforms.com/Tips/PopCount.htm */
112 uint32_t popcount (uint32_t u)
114 u = ((u&0x55555555) + ((u>>1)&0x55555555));
115 u = ((u&0x33333333) + ((u>>2)&0x33333333));
116 u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
117 u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
118 u = ( u&0x0000ffff) + (u>>16);
119 return u;
122 inline uint32_t lsbindex (uint32_t u)
124 return popcount ((u&-u)-1);
127 #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
128 #error No its not
129 #else
130 int audio_bug (const char *funcname, int cond)
132 if (cond) {
133 static int shown;
135 AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
136 if (!shown) {
137 shown = 1;
138 AUD_log (NULL, "Save all your work and restart without audio\n");
139 AUD_log (NULL, "Please send bug report to malc@pulsesoft.com\n");
140 AUD_log (NULL, "I am sorry\n");
142 AUD_log (NULL, "Context:\n");
144 #if defined AUDIO_BREAKPOINT_ON_BUG
145 # if defined HOST_I386
146 # if defined __GNUC__
147 __asm__ ("int3");
148 # elif defined _MSC_VER
149 _asm _emit 0xcc;
150 # else
151 abort ();
152 # endif
153 # else
154 abort ();
155 # endif
156 #endif
159 return cond;
161 #endif
163 static inline int audio_bits_to_index (int bits)
165 switch (bits) {
166 case 8:
167 return 0;
169 case 16:
170 return 1;
172 case 32:
173 return 2;
175 default:
176 audio_bug ("bits_to_index", 1);
177 AUD_log (NULL, "invalid bits %d\n", bits);
178 return 0;
182 void *audio_calloc (const char *funcname, int nmemb, size_t size)
184 int cond;
185 size_t len;
187 len = nmemb * size;
188 cond = !nmemb || !size;
189 cond |= nmemb < 0;
190 cond |= len < size;
192 if (audio_bug ("audio_calloc", cond)) {
193 AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
194 funcname);
195 AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
196 return NULL;
199 return qemu_mallocz (len);
202 static char *audio_alloc_prefix (const char *s)
204 const char qemu_prefix[] = "QEMU_";
205 size_t len, i;
206 char *r, *u;
208 if (!s) {
209 return NULL;
212 len = strlen (s);
213 r = qemu_malloc (len + sizeof (qemu_prefix));
215 u = r + sizeof (qemu_prefix) - 1;
217 pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
218 pstrcat (r, len + sizeof (qemu_prefix), s);
220 for (i = 0; i < len; ++i) {
221 u[i] = qemu_toupper(u[i]);
224 return r;
227 static const char *audio_audfmt_to_string (audfmt_e fmt)
229 switch (fmt) {
230 case AUD_FMT_U8:
231 return "U8";
233 case AUD_FMT_U16:
234 return "U16";
236 case AUD_FMT_S8:
237 return "S8";
239 case AUD_FMT_S16:
240 return "S16";
242 case AUD_FMT_U32:
243 return "U32";
245 case AUD_FMT_S32:
246 return "S32";
249 dolog ("Bogus audfmt %d returning S16\n", fmt);
250 return "S16";
253 static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
254 int *defaultp)
256 if (!strcasecmp (s, "u8")) {
257 *defaultp = 0;
258 return AUD_FMT_U8;
260 else if (!strcasecmp (s, "u16")) {
261 *defaultp = 0;
262 return AUD_FMT_U16;
264 else if (!strcasecmp (s, "u32")) {
265 *defaultp = 0;
266 return AUD_FMT_U32;
268 else if (!strcasecmp (s, "s8")) {
269 *defaultp = 0;
270 return AUD_FMT_S8;
272 else if (!strcasecmp (s, "s16")) {
273 *defaultp = 0;
274 return AUD_FMT_S16;
276 else if (!strcasecmp (s, "s32")) {
277 *defaultp = 0;
278 return AUD_FMT_S32;
280 else {
281 dolog ("Bogus audio format `%s' using %s\n",
282 s, audio_audfmt_to_string (defval));
283 *defaultp = 1;
284 return defval;
288 static audfmt_e audio_get_conf_fmt (const char *envname,
289 audfmt_e defval,
290 int *defaultp)
292 const char *var = getenv (envname);
293 if (!var) {
294 *defaultp = 1;
295 return defval;
297 return audio_string_to_audfmt (var, defval, defaultp);
300 static int audio_get_conf_int (const char *key, int defval, int *defaultp)
302 int val;
303 char *strval;
305 strval = getenv (key);
306 if (strval) {
307 *defaultp = 0;
308 val = atoi (strval);
309 return val;
311 else {
312 *defaultp = 1;
313 return defval;
317 static const char *audio_get_conf_str (const char *key,
318 const char *defval,
319 int *defaultp)
321 const char *val = getenv (key);
322 if (!val) {
323 *defaultp = 1;
324 return defval;
326 else {
327 *defaultp = 0;
328 return val;
332 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
334 if (conf.log_to_monitor) {
335 if (cap) {
336 monitor_printf(cur_mon, "%s: ", cap);
339 monitor_vprintf(cur_mon, fmt, ap);
341 else {
342 if (cap) {
343 fprintf (stderr, "%s: ", cap);
346 vfprintf (stderr, fmt, ap);
350 void AUD_log (const char *cap, const char *fmt, ...)
352 va_list ap;
354 va_start (ap, fmt);
355 AUD_vlog (cap, fmt, ap);
356 va_end (ap);
359 static void audio_print_options (const char *prefix,
360 struct audio_option *opt)
362 char *uprefix;
364 if (!prefix) {
365 dolog ("No prefix specified\n");
366 return;
369 if (!opt) {
370 dolog ("No options\n");
371 return;
374 uprefix = audio_alloc_prefix (prefix);
376 for (; opt->name; opt++) {
377 const char *state = "default";
378 printf (" %s_%s: ", uprefix, opt->name);
380 if (opt->overriddenp && *opt->overriddenp) {
381 state = "current";
384 switch (opt->tag) {
385 case AUD_OPT_BOOL:
387 int *intp = opt->valp;
388 printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
390 break;
392 case AUD_OPT_INT:
394 int *intp = opt->valp;
395 printf ("integer, %s = %d\n", state, *intp);
397 break;
399 case AUD_OPT_FMT:
401 audfmt_e *fmtp = opt->valp;
402 printf (
403 "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
404 state,
405 audio_audfmt_to_string (*fmtp)
408 break;
410 case AUD_OPT_STR:
412 const char **strp = opt->valp;
413 printf ("string, %s = %s\n",
414 state,
415 *strp ? *strp : "(not set)");
417 break;
419 default:
420 printf ("???\n");
421 dolog ("Bad value tag for option %s_%s %d\n",
422 uprefix, opt->name, opt->tag);
423 break;
425 printf (" %s\n", opt->descr);
428 qemu_free (uprefix);
431 static void audio_process_options (const char *prefix,
432 struct audio_option *opt)
434 char *optname;
435 const char qemu_prefix[] = "QEMU_";
436 size_t preflen, optlen;
438 if (audio_bug (AUDIO_FUNC, !prefix)) {
439 dolog ("prefix = NULL\n");
440 return;
443 if (audio_bug (AUDIO_FUNC, !opt)) {
444 dolog ("opt = NULL\n");
445 return;
448 preflen = strlen (prefix);
450 for (; opt->name; opt++) {
451 size_t len, i;
452 int def;
454 if (!opt->valp) {
455 dolog ("Option value pointer for `%s' is not set\n",
456 opt->name);
457 continue;
460 len = strlen (opt->name);
461 /* len of opt->name + len of prefix + size of qemu_prefix
462 * (includes trailing zero) + zero + underscore (on behalf of
463 * sizeof) */
464 optlen = len + preflen + sizeof (qemu_prefix) + 1;
465 optname = qemu_malloc (optlen);
467 pstrcpy (optname, optlen, qemu_prefix);
469 /* copy while upper-casing, including trailing zero */
470 for (i = 0; i <= preflen; ++i) {
471 optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
473 pstrcat (optname, optlen, "_");
474 pstrcat (optname, optlen, opt->name);
476 def = 1;
477 switch (opt->tag) {
478 case AUD_OPT_BOOL:
479 case AUD_OPT_INT:
481 int *intp = opt->valp;
482 *intp = audio_get_conf_int (optname, *intp, &def);
484 break;
486 case AUD_OPT_FMT:
488 audfmt_e *fmtp = opt->valp;
489 *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
491 break;
493 case AUD_OPT_STR:
495 const char **strp = opt->valp;
496 *strp = audio_get_conf_str (optname, *strp, &def);
498 break;
500 default:
501 dolog ("Bad value tag for option `%s' - %d\n",
502 optname, opt->tag);
503 break;
506 if (!opt->overriddenp) {
507 opt->overriddenp = &opt->overridden;
509 *opt->overriddenp = !def;
510 qemu_free (optname);
514 static void audio_print_settings (struct audsettings *as)
516 dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
518 switch (as->fmt) {
519 case AUD_FMT_S8:
520 AUD_log (NULL, "S8");
521 break;
522 case AUD_FMT_U8:
523 AUD_log (NULL, "U8");
524 break;
525 case AUD_FMT_S16:
526 AUD_log (NULL, "S16");
527 break;
528 case AUD_FMT_U16:
529 AUD_log (NULL, "U16");
530 break;
531 case AUD_FMT_S32:
532 AUD_log (NULL, "S32");
533 break;
534 case AUD_FMT_U32:
535 AUD_log (NULL, "U32");
536 break;
537 default:
538 AUD_log (NULL, "invalid(%d)", as->fmt);
539 break;
542 AUD_log (NULL, " endianness=");
543 switch (as->endianness) {
544 case 0:
545 AUD_log (NULL, "little");
546 break;
547 case 1:
548 AUD_log (NULL, "big");
549 break;
550 default:
551 AUD_log (NULL, "invalid");
552 break;
554 AUD_log (NULL, "\n");
557 static int audio_validate_settings (struct audsettings *as)
559 int invalid;
561 invalid = as->nchannels != 1 && as->nchannels != 2;
562 invalid |= as->endianness != 0 && as->endianness != 1;
564 switch (as->fmt) {
565 case AUD_FMT_S8:
566 case AUD_FMT_U8:
567 case AUD_FMT_S16:
568 case AUD_FMT_U16:
569 case AUD_FMT_S32:
570 case AUD_FMT_U32:
571 break;
572 default:
573 invalid = 1;
574 break;
577 invalid |= as->freq <= 0;
578 return invalid ? -1 : 0;
581 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
583 int bits = 8, sign = 0;
585 switch (as->fmt) {
586 case AUD_FMT_S8:
587 sign = 1;
588 case AUD_FMT_U8:
589 break;
591 case AUD_FMT_S16:
592 sign = 1;
593 case AUD_FMT_U16:
594 bits = 16;
595 break;
597 case AUD_FMT_S32:
598 sign = 1;
599 case AUD_FMT_U32:
600 bits = 32;
601 break;
603 return info->freq == as->freq
604 && info->nchannels == as->nchannels
605 && info->sign == sign
606 && info->bits == bits
607 && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
610 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
612 int bits = 8, sign = 0, shift = 0;
614 switch (as->fmt) {
615 case AUD_FMT_S8:
616 sign = 1;
617 case AUD_FMT_U8:
618 break;
620 case AUD_FMT_S16:
621 sign = 1;
622 case AUD_FMT_U16:
623 bits = 16;
624 shift = 1;
625 break;
627 case AUD_FMT_S32:
628 sign = 1;
629 case AUD_FMT_U32:
630 bits = 32;
631 shift = 2;
632 break;
635 info->freq = as->freq;
636 info->bits = bits;
637 info->sign = sign;
638 info->nchannels = as->nchannels;
639 info->shift = (as->nchannels == 2) + shift;
640 info->align = (1 << info->shift) - 1;
641 info->bytes_per_second = info->freq << info->shift;
642 info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
645 void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
647 if (!len) {
648 return;
651 if (info->sign) {
652 memset (buf, 0x00, len << info->shift);
654 else {
655 switch (info->bits) {
656 case 8:
657 memset (buf, 0x80, len << info->shift);
658 break;
660 case 16:
662 int i;
663 uint16_t *p = buf;
664 int shift = info->nchannels - 1;
665 short s = INT16_MAX;
667 if (info->swap_endianness) {
668 s = bswap16 (s);
671 for (i = 0; i < len << shift; i++) {
672 p[i] = s;
675 break;
677 case 32:
679 int i;
680 uint32_t *p = buf;
681 int shift = info->nchannels - 1;
682 int32_t s = INT32_MAX;
684 if (info->swap_endianness) {
685 s = bswap32 (s);
688 for (i = 0; i < len << shift; i++) {
689 p[i] = s;
692 break;
694 default:
695 AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
696 info->bits);
697 break;
703 * Capture
705 static void noop_conv (struct st_sample *dst, const void *src,
706 int samples, struct mixeng_volume *vol)
708 (void) src;
709 (void) dst;
710 (void) samples;
711 (void) vol;
714 static CaptureVoiceOut *audio_pcm_capture_find_specific (
715 struct audsettings *as
718 CaptureVoiceOut *cap;
719 AudioState *s = &glob_audio_state;
721 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
722 if (audio_pcm_info_eq (&cap->hw.info, as)) {
723 return cap;
726 return NULL;
729 static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
731 struct capture_callback *cb;
733 #ifdef DEBUG_CAPTURE
734 dolog ("notification %d sent\n", cmd);
735 #endif
736 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
737 cb->ops.notify (cb->opaque, cmd);
741 static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
743 if (cap->hw.enabled != enabled) {
744 audcnotification_e cmd;
745 cap->hw.enabled = enabled;
746 cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
747 audio_notify_capture (cap, cmd);
751 static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
753 HWVoiceOut *hw = &cap->hw;
754 SWVoiceOut *sw;
755 int enabled = 0;
757 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
758 if (sw->active) {
759 enabled = 1;
760 break;
763 audio_capture_maybe_changed (cap, enabled);
766 static void audio_detach_capture (HWVoiceOut *hw)
768 SWVoiceCap *sc = hw->cap_head.lh_first;
770 while (sc) {
771 SWVoiceCap *sc1 = sc->entries.le_next;
772 SWVoiceOut *sw = &sc->sw;
773 CaptureVoiceOut *cap = sc->cap;
774 int was_active = sw->active;
776 if (sw->rate) {
777 st_rate_stop (sw->rate);
778 sw->rate = NULL;
781 LIST_REMOVE (sw, entries);
782 LIST_REMOVE (sc, entries);
783 qemu_free (sc);
784 if (was_active) {
785 /* We have removed soft voice from the capture:
786 this might have changed the overall status of the capture
787 since this might have been the only active voice */
788 audio_recalc_and_notify_capture (cap);
790 sc = sc1;
794 static int audio_attach_capture (HWVoiceOut *hw)
796 AudioState *s = &glob_audio_state;
797 CaptureVoiceOut *cap;
799 audio_detach_capture (hw);
800 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
801 SWVoiceCap *sc;
802 SWVoiceOut *sw;
803 HWVoiceOut *hw_cap = &cap->hw;
805 sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
806 if (!sc) {
807 dolog ("Could not allocate soft capture voice (%zu bytes)\n",
808 sizeof (*sc));
809 return -1;
812 sc->cap = cap;
813 sw = &sc->sw;
814 sw->hw = hw_cap;
815 sw->info = hw->info;
816 sw->empty = 1;
817 sw->active = hw->enabled;
818 sw->conv = noop_conv;
819 sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
820 sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
821 if (!sw->rate) {
822 dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
823 qemu_free (sw);
824 return -1;
826 LIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
827 LIST_INSERT_HEAD (&hw->cap_head, sc, entries);
828 #ifdef DEBUG_CAPTURE
829 asprintf (&sw->name, "for %p %d,%d,%d",
830 hw, sw->info.freq, sw->info.bits, sw->info.nchannels);
831 dolog ("Added %s active = %d\n", sw->name, sw->active);
832 #endif
833 if (sw->active) {
834 audio_capture_maybe_changed (cap, 1);
837 return 0;
841 * Hard voice (capture)
843 static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
845 SWVoiceIn *sw;
846 int m = hw->total_samples_captured;
848 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
849 if (sw->active) {
850 m = audio_MIN (m, sw->total_hw_samples_acquired);
853 return m;
856 int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
858 int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
859 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
860 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
861 return 0;
863 return live;
867 * Soft voice (capture)
869 static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
871 HWVoiceIn *hw = sw->hw;
872 int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
873 int rpos;
875 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
876 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
877 return 0;
880 rpos = hw->wpos - live;
881 if (rpos >= 0) {
882 return rpos;
884 else {
885 return hw->samples + rpos;
889 int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
891 HWVoiceIn *hw = sw->hw;
892 int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
893 struct st_sample *src, *dst = sw->buf;
895 rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
897 live = hw->total_samples_captured - sw->total_hw_samples_acquired;
898 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
899 dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
900 return 0;
903 samples = size >> sw->info.shift;
904 if (!live) {
905 return 0;
908 swlim = (live * sw->ratio) >> 32;
909 swlim = audio_MIN (swlim, samples);
911 while (swlim) {
912 src = hw->conv_buf + rpos;
913 isamp = hw->wpos - rpos;
914 /* XXX: <= ? */
915 if (isamp <= 0) {
916 isamp = hw->samples - rpos;
919 if (!isamp) {
920 break;
922 osamp = swlim;
924 if (audio_bug (AUDIO_FUNC, osamp < 0)) {
925 dolog ("osamp=%d\n", osamp);
926 return 0;
929 st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
930 swlim -= osamp;
931 rpos = (rpos + isamp) % hw->samples;
932 dst += osamp;
933 ret += osamp;
934 total += isamp;
937 sw->clip (buf, sw->buf, ret);
938 sw->total_hw_samples_acquired += total;
939 return ret << sw->info.shift;
943 * Hard voice (playback)
945 static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
947 SWVoiceOut *sw;
948 int m = INT_MAX;
949 int nb_live = 0;
951 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
952 if (sw->active || !sw->empty) {
953 m = audio_MIN (m, sw->total_hw_samples_mixed);
954 nb_live += 1;
958 *nb_livep = nb_live;
959 return m;
962 int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
964 int smin;
966 smin = audio_pcm_hw_find_min_out (hw, nb_live);
968 if (!*nb_live) {
969 return 0;
971 else {
972 int live = smin;
974 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
975 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
976 return 0;
978 return live;
982 int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
984 int nb_live;
985 int live;
987 live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
988 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
989 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
990 return 0;
992 return live;
996 * Soft voice (playback)
998 int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
1000 int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
1001 int ret = 0, pos = 0, total = 0;
1003 if (!sw) {
1004 return size;
1007 hwsamples = sw->hw->samples;
1009 live = sw->total_hw_samples_mixed;
1010 if (audio_bug (AUDIO_FUNC, live < 0 || live > hwsamples)){
1011 dolog ("live=%d hw->samples=%d\n", live, hwsamples);
1012 return 0;
1015 if (live == hwsamples) {
1016 #ifdef DEBUG_OUT
1017 dolog ("%s is full %d\n", sw->name, live);
1018 #endif
1019 return 0;
1022 wpos = (sw->hw->rpos + live) % hwsamples;
1023 samples = size >> sw->info.shift;
1025 dead = hwsamples - live;
1026 swlim = ((int64_t) dead << 32) / sw->ratio;
1027 swlim = audio_MIN (swlim, samples);
1028 if (swlim) {
1029 sw->conv (sw->buf, buf, swlim, &sw->vol);
1032 while (swlim) {
1033 dead = hwsamples - live;
1034 left = hwsamples - wpos;
1035 blck = audio_MIN (dead, left);
1036 if (!blck) {
1037 break;
1039 isamp = swlim;
1040 osamp = blck;
1041 st_rate_flow_mix (
1042 sw->rate,
1043 sw->buf + pos,
1044 sw->hw->mix_buf + wpos,
1045 &isamp,
1046 &osamp
1048 ret += isamp;
1049 swlim -= isamp;
1050 pos += isamp;
1051 live += osamp;
1052 wpos = (wpos + osamp) % hwsamples;
1053 total += osamp;
1056 sw->total_hw_samples_mixed += total;
1057 sw->empty = sw->total_hw_samples_mixed == 0;
1059 #ifdef DEBUG_OUT
1060 dolog (
1061 "%s: write size %d ret %d total sw %d\n",
1062 SW_NAME (sw),
1063 size >> sw->info.shift,
1064 ret,
1065 sw->total_hw_samples_mixed
1067 #endif
1069 return ret << sw->info.shift;
1072 #ifdef DEBUG_AUDIO
1073 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
1075 dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
1076 cap, info->bits, info->sign, info->freq, info->nchannels);
1078 #endif
1080 #define DAC
1081 #include "audio_template.h"
1082 #undef DAC
1083 #include "audio_template.h"
1085 int AUD_write (SWVoiceOut *sw, void *buf, int size)
1087 int bytes;
1089 if (!sw) {
1090 /* XXX: Consider options */
1091 return size;
1094 if (!sw->hw->enabled) {
1095 dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
1096 return 0;
1099 bytes = sw->hw->pcm_ops->write (sw, buf, size);
1100 return bytes;
1103 int AUD_read (SWVoiceIn *sw, void *buf, int size)
1105 int bytes;
1107 if (!sw) {
1108 /* XXX: Consider options */
1109 return size;
1112 if (!sw->hw->enabled) {
1113 dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
1114 return 0;
1117 bytes = sw->hw->pcm_ops->read (sw, buf, size);
1118 return bytes;
1121 int AUD_get_buffer_size_out (SWVoiceOut *sw)
1123 return sw->hw->samples << sw->hw->info.shift;
1126 void AUD_set_active_out (SWVoiceOut *sw, int on)
1128 HWVoiceOut *hw;
1130 if (!sw) {
1131 return;
1134 hw = sw->hw;
1135 if (sw->active != on) {
1136 AudioState *s = &glob_audio_state;
1137 SWVoiceOut *temp_sw;
1138 SWVoiceCap *sc;
1140 if (on) {
1141 hw->pending_disable = 0;
1142 if (!hw->enabled) {
1143 hw->enabled = 1;
1144 if (s->vm_running) {
1145 hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
1149 else {
1150 if (hw->enabled) {
1151 int nb_active = 0;
1153 for (temp_sw = hw->sw_head.lh_first; temp_sw;
1154 temp_sw = temp_sw->entries.le_next) {
1155 nb_active += temp_sw->active != 0;
1158 hw->pending_disable = nb_active == 1;
1162 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1163 sc->sw.active = hw->enabled;
1164 if (hw->enabled) {
1165 audio_capture_maybe_changed (sc->cap, 1);
1168 sw->active = on;
1172 void AUD_set_active_in (SWVoiceIn *sw, int on)
1174 HWVoiceIn *hw;
1176 if (!sw) {
1177 return;
1180 hw = sw->hw;
1181 if (sw->active != on) {
1182 AudioState *s = &glob_audio_state;
1183 SWVoiceIn *temp_sw;
1185 if (on) {
1186 if (!hw->enabled) {
1187 hw->enabled = 1;
1188 if (s->vm_running) {
1189 hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
1192 sw->total_hw_samples_acquired = hw->total_samples_captured;
1194 else {
1195 if (hw->enabled) {
1196 int nb_active = 0;
1198 for (temp_sw = hw->sw_head.lh_first; temp_sw;
1199 temp_sw = temp_sw->entries.le_next) {
1200 nb_active += temp_sw->active != 0;
1203 if (nb_active == 1) {
1204 hw->enabled = 0;
1205 hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
1209 sw->active = on;
1213 static int audio_get_avail (SWVoiceIn *sw)
1215 int live;
1217 if (!sw) {
1218 return 0;
1221 live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
1222 if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
1223 dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
1224 return 0;
1227 ldebug (
1228 "%s: get_avail live %d ret %" PRId64 "\n",
1229 SW_NAME (sw),
1230 live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
1233 return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
1236 static int audio_get_free (SWVoiceOut *sw)
1238 int live, dead;
1240 if (!sw) {
1241 return 0;
1244 live = sw->total_hw_samples_mixed;
1246 if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
1247 dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
1248 return 0;
1251 dead = sw->hw->samples - live;
1253 #ifdef DEBUG_OUT
1254 dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
1255 SW_NAME (sw),
1256 live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
1257 #endif
1259 return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
1262 static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
1264 int n;
1266 if (hw->enabled) {
1267 SWVoiceCap *sc;
1269 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1270 SWVoiceOut *sw = &sc->sw;
1271 int rpos2 = rpos;
1273 n = samples;
1274 while (n) {
1275 int till_end_of_hw = hw->samples - rpos2;
1276 int to_write = audio_MIN (till_end_of_hw, n);
1277 int bytes = to_write << hw->info.shift;
1278 int written;
1280 sw->buf = hw->mix_buf + rpos2;
1281 written = audio_pcm_sw_write (sw, NULL, bytes);
1282 if (written - bytes) {
1283 dolog ("Could not mix %d bytes into a capture "
1284 "buffer, mixed %d\n",
1285 bytes, written);
1286 break;
1288 n -= to_write;
1289 rpos2 = (rpos2 + to_write) % hw->samples;
1294 n = audio_MIN (samples, hw->samples - rpos);
1295 mixeng_clear (hw->mix_buf + rpos, n);
1296 mixeng_clear (hw->mix_buf, samples - n);
1299 static void audio_run_out (AudioState *s)
1301 HWVoiceOut *hw = NULL;
1302 SWVoiceOut *sw;
1304 while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) {
1305 int played;
1306 int live, free, nb_live, cleanup_required, prev_rpos;
1308 live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
1309 if (!nb_live) {
1310 live = 0;
1313 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
1314 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
1315 continue;
1318 if (hw->pending_disable && !nb_live) {
1319 SWVoiceCap *sc;
1320 #ifdef DEBUG_OUT
1321 dolog ("Disabling voice\n");
1322 #endif
1323 hw->enabled = 0;
1324 hw->pending_disable = 0;
1325 hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
1326 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1327 sc->sw.active = 0;
1328 audio_recalc_and_notify_capture (sc->cap);
1330 continue;
1333 if (!live) {
1334 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1335 if (sw->active) {
1336 free = audio_get_free (sw);
1337 if (free > 0) {
1338 sw->callback.fn (sw->callback.opaque, free);
1342 continue;
1345 prev_rpos = hw->rpos;
1346 played = hw->pcm_ops->run_out (hw);
1347 if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
1348 dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
1349 hw->rpos, hw->samples, played);
1350 hw->rpos = 0;
1353 #ifdef DEBUG_OUT
1354 dolog ("played=%d\n", played);
1355 #endif
1357 if (played) {
1358 hw->ts_helper += played;
1359 audio_capture_mix_and_clear (hw, prev_rpos, played);
1362 cleanup_required = 0;
1363 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1364 if (!sw->active && sw->empty) {
1365 continue;
1368 if (audio_bug (AUDIO_FUNC, played > sw->total_hw_samples_mixed)) {
1369 dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
1370 played, sw->total_hw_samples_mixed);
1371 played = sw->total_hw_samples_mixed;
1374 sw->total_hw_samples_mixed -= played;
1376 if (!sw->total_hw_samples_mixed) {
1377 sw->empty = 1;
1378 cleanup_required |= !sw->active && !sw->callback.fn;
1381 if (sw->active) {
1382 free = audio_get_free (sw);
1383 if (free > 0) {
1384 sw->callback.fn (sw->callback.opaque, free);
1389 if (cleanup_required) {
1390 SWVoiceOut *sw1;
1392 sw = hw->sw_head.lh_first;
1393 while (sw) {
1394 sw1 = sw->entries.le_next;
1395 if (!sw->active && !sw->callback.fn) {
1396 #ifdef DEBUG_PLIVE
1397 dolog ("Finishing with old voice\n");
1398 #endif
1399 audio_close_out (sw);
1401 sw = sw1;
1407 static void audio_run_in (AudioState *s)
1409 HWVoiceIn *hw = NULL;
1411 while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
1412 SWVoiceIn *sw;
1413 int captured, min;
1415 captured = hw->pcm_ops->run_in (hw);
1417 min = audio_pcm_hw_find_min_in (hw);
1418 hw->total_samples_captured += captured - min;
1419 hw->ts_helper += captured;
1421 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1422 sw->total_hw_samples_acquired -= min;
1424 if (sw->active) {
1425 int avail;
1427 avail = audio_get_avail (sw);
1428 if (avail > 0) {
1429 sw->callback.fn (sw->callback.opaque, avail);
1436 static void audio_run_capture (AudioState *s)
1438 CaptureVoiceOut *cap;
1440 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
1441 int live, rpos, captured;
1442 HWVoiceOut *hw = &cap->hw;
1443 SWVoiceOut *sw;
1445 captured = live = audio_pcm_hw_get_live_out (hw);
1446 rpos = hw->rpos;
1447 while (live) {
1448 int left = hw->samples - rpos;
1449 int to_capture = audio_MIN (live, left);
1450 struct st_sample *src;
1451 struct capture_callback *cb;
1453 src = hw->mix_buf + rpos;
1454 hw->clip (cap->buf, src, to_capture);
1455 mixeng_clear (src, to_capture);
1457 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1458 cb->ops.capture (cb->opaque, cap->buf,
1459 to_capture << hw->info.shift);
1461 rpos = (rpos + to_capture) % hw->samples;
1462 live -= to_capture;
1464 hw->rpos = rpos;
1466 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1467 if (!sw->active && sw->empty) {
1468 continue;
1471 if (audio_bug (AUDIO_FUNC, captured > sw->total_hw_samples_mixed)) {
1472 dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
1473 captured, sw->total_hw_samples_mixed);
1474 captured = sw->total_hw_samples_mixed;
1477 sw->total_hw_samples_mixed -= captured;
1478 sw->empty = sw->total_hw_samples_mixed == 0;
1483 static void audio_timer (void *opaque)
1485 AudioState *s = opaque;
1487 audio_run_out (s);
1488 audio_run_in (s);
1489 audio_run_capture (s);
1491 qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
1494 static struct audio_option audio_options[] = {
1495 /* DAC */
1497 .name = "DAC_FIXED_SETTINGS",
1498 .tag = AUD_OPT_BOOL,
1499 .valp = &conf.fixed_out.enabled,
1500 .descr = "Use fixed settings for host DAC"
1503 .name = "DAC_FIXED_FREQ",
1504 .tag = AUD_OPT_INT,
1505 .valp = &conf.fixed_out.settings.freq,
1506 .descr = "Frequency for fixed host DAC"
1509 .name = "DAC_FIXED_FMT",
1510 .tag = AUD_OPT_FMT,
1511 .valp = &conf.fixed_out.settings.fmt,
1512 .descr = "Format for fixed host DAC"
1515 .name = "DAC_FIXED_CHANNELS",
1516 .tag = AUD_OPT_INT,
1517 .valp = &conf.fixed_out.settings.nchannels,
1518 .descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)"
1521 .name = "DAC_VOICES",
1522 .tag = AUD_OPT_INT,
1523 .valp = &conf.fixed_out.nb_voices,
1524 .descr = "Number of voices for DAC"
1526 /* ADC */
1528 .name = "ADC_FIXED_SETTINGS",
1529 .tag = AUD_OPT_BOOL,
1530 .valp = &conf.fixed_in.enabled,
1531 .descr = "Use fixed settings for host ADC"
1534 .name = "ADC_FIXED_FREQ",
1535 .tag = AUD_OPT_INT,
1536 .valp = &conf.fixed_in.settings.freq,
1537 .descr = "Frequency for fixed host ADC"
1540 .name = "ADC_FIXED_FMT",
1541 .tag = AUD_OPT_FMT,
1542 .valp = &conf.fixed_in.settings.fmt,
1543 .descr = "Format for fixed host ADC"
1546 .name = "ADC_FIXED_CHANNELS",
1547 .tag = AUD_OPT_INT,
1548 .valp = &conf.fixed_in.settings.nchannels,
1549 .descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)"
1552 .name = "ADC_VOICES",
1553 .tag = AUD_OPT_INT,
1554 .valp = &conf.fixed_in.nb_voices,
1555 .descr = "Number of voices for ADC"
1557 /* Misc */
1559 .name = "TIMER_PERIOD",
1560 .tag = AUD_OPT_INT,
1561 .valp = &conf.period.hertz,
1562 .descr = "Timer period in HZ (0 - use lowest possible)"
1565 .name = "PLIVE",
1566 .tag = AUD_OPT_BOOL,
1567 .valp = &conf.plive,
1568 .descr = "(undocumented)"
1571 .name = "LOG_TO_MONITOR",
1572 .tag = AUD_OPT_BOOL,
1573 .valp = &conf.log_to_monitor,
1574 .descr = "print logging messages to monitor instead of stderr"
1576 { /* End of list */ }
1579 static void audio_pp_nb_voices (const char *typ, int nb)
1581 switch (nb) {
1582 case 0:
1583 printf ("Does not support %s\n", typ);
1584 break;
1585 case 1:
1586 printf ("One %s voice\n", typ);
1587 break;
1588 case INT_MAX:
1589 printf ("Theoretically supports many %s voices\n", typ);
1590 break;
1591 default:
1592 printf ("Theoretically supports upto %d %s voices\n", nb, typ);
1593 break;
1598 void AUD_help (void)
1600 size_t i;
1602 audio_process_options ("AUDIO", audio_options);
1603 for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1604 struct audio_driver *d = drvtab[i];
1605 if (d->options) {
1606 audio_process_options (d->name, d->options);
1610 printf ("Audio options:\n");
1611 audio_print_options ("AUDIO", audio_options);
1612 printf ("\n");
1614 printf ("Available drivers:\n");
1616 for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1617 struct audio_driver *d = drvtab[i];
1619 printf ("Name: %s\n", d->name);
1620 printf ("Description: %s\n", d->descr);
1622 audio_pp_nb_voices ("playback", d->max_voices_out);
1623 audio_pp_nb_voices ("capture", d->max_voices_in);
1625 if (d->options) {
1626 printf ("Options:\n");
1627 audio_print_options (d->name, d->options);
1629 else {
1630 printf ("No options\n");
1632 printf ("\n");
1635 printf (
1636 "Options are settable through environment variables.\n"
1637 "Example:\n"
1638 #ifdef _WIN32
1639 " set QEMU_AUDIO_DRV=wav\n"
1640 " set QEMU_WAV_PATH=c:\\tune.wav\n"
1641 #else
1642 " export QEMU_AUDIO_DRV=wav\n"
1643 " export QEMU_WAV_PATH=$HOME/tune.wav\n"
1644 "(for csh replace export with setenv in the above)\n"
1645 #endif
1646 " qemu ...\n\n"
1650 static int audio_driver_init (AudioState *s, struct audio_driver *drv)
1652 if (drv->options) {
1653 audio_process_options (drv->name, drv->options);
1655 s->drv_opaque = drv->init ();
1657 if (s->drv_opaque) {
1658 audio_init_nb_voices_out (drv);
1659 audio_init_nb_voices_in (drv);
1660 s->drv = drv;
1661 return 0;
1663 else {
1664 dolog ("Could not init `%s' audio driver\n", drv->name);
1665 return -1;
1669 static void audio_vm_change_state_handler (void *opaque, int running,
1670 int reason)
1672 AudioState *s = opaque;
1673 HWVoiceOut *hwo = NULL;
1674 HWVoiceIn *hwi = NULL;
1675 int op = running ? VOICE_ENABLE : VOICE_DISABLE;
1677 s->vm_running = running;
1678 while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
1679 hwo->pcm_ops->ctl_out (hwo, op);
1682 while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
1683 hwi->pcm_ops->ctl_in (hwi, op);
1687 static void audio_atexit (void)
1689 AudioState *s = &glob_audio_state;
1690 HWVoiceOut *hwo = NULL;
1691 HWVoiceIn *hwi = NULL;
1693 while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
1694 SWVoiceCap *sc;
1696 hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
1697 hwo->pcm_ops->fini_out (hwo);
1699 for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1700 CaptureVoiceOut *cap = sc->cap;
1701 struct capture_callback *cb;
1703 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1704 cb->ops.destroy (cb->opaque);
1709 while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
1710 hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
1711 hwi->pcm_ops->fini_in (hwi);
1714 if (s->drv) {
1715 s->drv->fini (s->drv_opaque);
1719 static void audio_save (QEMUFile *f, void *opaque)
1721 (void) f;
1722 (void) opaque;
1725 static int audio_load (QEMUFile *f, void *opaque, int version_id)
1727 (void) f;
1728 (void) opaque;
1730 if (version_id != 1) {
1731 return -EINVAL;
1734 return 0;
1737 static void audio_init (void)
1739 size_t i;
1740 int done = 0;
1741 const char *drvname;
1742 AudioState *s = &glob_audio_state;
1744 if (s->drv) {
1745 return;
1748 LIST_INIT (&s->hw_head_out);
1749 LIST_INIT (&s->hw_head_in);
1750 LIST_INIT (&s->cap_head);
1751 atexit (audio_atexit);
1753 s->ts = qemu_new_timer (vm_clock, audio_timer, s);
1754 if (!s->ts) {
1755 hw_error("Could not create audio timer\n");
1758 audio_process_options ("AUDIO", audio_options);
1760 s->nb_hw_voices_out = conf.fixed_out.nb_voices;
1761 s->nb_hw_voices_in = conf.fixed_in.nb_voices;
1763 if (s->nb_hw_voices_out <= 0) {
1764 dolog ("Bogus number of playback voices %d, setting to 1\n",
1765 s->nb_hw_voices_out);
1766 s->nb_hw_voices_out = 1;
1769 if (s->nb_hw_voices_in <= 0) {
1770 dolog ("Bogus number of capture voices %d, setting to 0\n",
1771 s->nb_hw_voices_in);
1772 s->nb_hw_voices_in = 0;
1776 int def;
1777 drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
1780 if (drvname) {
1781 int found = 0;
1783 for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1784 if (!strcmp (drvname, drvtab[i]->name)) {
1785 done = !audio_driver_init (s, drvtab[i]);
1786 found = 1;
1787 break;
1791 if (!found) {
1792 dolog ("Unknown audio driver `%s'\n", drvname);
1793 dolog ("Run with -audio-help to list available drivers\n");
1797 if (!done) {
1798 for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
1799 if (drvtab[i]->can_be_default) {
1800 done = !audio_driver_init (s, drvtab[i]);
1805 if (!done) {
1806 done = !audio_driver_init (s, &no_audio_driver);
1807 if (!done) {
1808 hw_error("Could not initialize audio subsystem\n");
1810 else {
1811 dolog ("warning: Using timer based audio emulation\n");
1815 VMChangeStateEntry *e;
1817 if (conf.period.hertz <= 0) {
1818 if (conf.period.hertz < 0) {
1819 dolog ("warning: Timer period is negative - %d "
1820 "treating as zero\n",
1821 conf.period.hertz);
1823 conf.period.ticks = 1;
1824 } else {
1825 conf.period.ticks = ticks_per_sec / conf.period.hertz;
1828 e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
1829 if (!e) {
1830 dolog ("warning: Could not register change state handler\n"
1831 "(Audio can continue looping even after stopping the VM)\n");
1834 LIST_INIT (&s->card_head);
1835 register_savevm ("audio", 0, 1, audio_save, audio_load, s);
1836 qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
1839 void AUD_register_card (const char *name, QEMUSoundCard *card)
1841 audio_init ();
1842 card->name = qemu_strdup (name);
1843 memset (&card->entries, 0, sizeof (card->entries));
1844 LIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
1847 void AUD_remove_card (QEMUSoundCard *card)
1849 LIST_REMOVE (card, entries);
1850 qemu_free (card->name);
1854 CaptureVoiceOut *AUD_add_capture (
1855 struct audsettings *as,
1856 struct audio_capture_ops *ops,
1857 void *cb_opaque
1860 AudioState *s = &glob_audio_state;
1861 CaptureVoiceOut *cap;
1862 struct capture_callback *cb;
1864 if (audio_validate_settings (as)) {
1865 dolog ("Invalid settings were passed when trying to add capture\n");
1866 audio_print_settings (as);
1867 goto err0;
1870 cb = audio_calloc (AUDIO_FUNC, 1, sizeof (*cb));
1871 if (!cb) {
1872 dolog ("Could not allocate capture callback information, size %zu\n",
1873 sizeof (*cb));
1874 goto err0;
1876 cb->ops = *ops;
1877 cb->opaque = cb_opaque;
1879 cap = audio_pcm_capture_find_specific (as);
1880 if (cap) {
1881 LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1882 return cap;
1884 else {
1885 HWVoiceOut *hw;
1886 CaptureVoiceOut *cap;
1888 cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap));
1889 if (!cap) {
1890 dolog ("Could not allocate capture voice, size %zu\n",
1891 sizeof (*cap));
1892 goto err1;
1895 hw = &cap->hw;
1896 LIST_INIT (&hw->sw_head);
1897 LIST_INIT (&cap->cb_head);
1899 /* XXX find a more elegant way */
1900 hw->samples = 4096 * 4;
1901 hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
1902 sizeof (struct st_sample));
1903 if (!hw->mix_buf) {
1904 dolog ("Could not allocate capture mix buffer (%d samples)\n",
1905 hw->samples);
1906 goto err2;
1909 audio_pcm_init_info (&hw->info, as);
1911 cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
1912 if (!cap->buf) {
1913 dolog ("Could not allocate capture buffer "
1914 "(%d samples, each %d bytes)\n",
1915 hw->samples, 1 << hw->info.shift);
1916 goto err3;
1919 hw->clip = mixeng_clip
1920 [hw->info.nchannels == 2]
1921 [hw->info.sign]
1922 [hw->info.swap_endianness]
1923 [audio_bits_to_index (hw->info.bits)];
1925 LIST_INSERT_HEAD (&s->cap_head, cap, entries);
1926 LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1928 hw = NULL;
1929 while ((hw = audio_pcm_hw_find_any_out (hw))) {
1930 audio_attach_capture (hw);
1932 return cap;
1934 err3:
1935 qemu_free (cap->hw.mix_buf);
1936 err2:
1937 qemu_free (cap);
1938 err1:
1939 qemu_free (cb);
1940 err0:
1941 return NULL;
1945 void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
1947 struct capture_callback *cb;
1949 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1950 if (cb->opaque == cb_opaque) {
1951 cb->ops.destroy (cb_opaque);
1952 LIST_REMOVE (cb, entries);
1953 qemu_free (cb);
1955 if (!cap->cb_head.lh_first) {
1956 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
1958 while (sw) {
1959 SWVoiceCap *sc = (SWVoiceCap *) sw;
1960 #ifdef DEBUG_CAPTURE
1961 dolog ("freeing %s\n", sw->name);
1962 #endif
1964 sw1 = sw->entries.le_next;
1965 if (sw->rate) {
1966 st_rate_stop (sw->rate);
1967 sw->rate = NULL;
1969 LIST_REMOVE (sw, entries);
1970 LIST_REMOVE (sc, entries);
1971 qemu_free (sc);
1972 sw = sw1;
1974 LIST_REMOVE (cap, entries);
1975 qemu_free (cap);
1977 return;
1982 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
1984 if (sw) {
1985 sw->vol.mute = mute;
1986 sw->vol.l = nominal_volume.l * lvol / 255;
1987 sw->vol.r = nominal_volume.r * rvol / 255;
1991 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
1993 if (sw) {
1994 sw->vol.mute = mute;
1995 sw->vol.l = nominal_volume.l * lvol / 255;
1996 sw->vol.r = nominal_volume.r * rvol / 255;