1 /*
2 * QEMU Audio subsystem
3 *
4 * Copyright (c) 2007-2008 The Android Open Source Project
5 * Copyright (c) 2003-2005 Vassili Karpov (malc)
6 *
7 * Permission is hereby granted, free of charge, to any person obtaining a copy
8 * of this software and associated documentation files (the "Software"), to deal
9 * in the Software without restriction, including without limitation the rights
10 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
11 * copies of the Software, and to permit persons to whom the Software is
12 * furnished to do so, subject to the following conditions:
13 *
14 * The above copyright notice and this permission notice shall be included in
15 * all copies or substantial portions of the Software.
16 *
17 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
18 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
19 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
20 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
21 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
22 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
23 * THE SOFTWARE.
24 */
25 #include "hw/hw.h"
26 #include "audio.h"
27 #include "monitor.h"
28 #include "qemu-timer.h"
29 #include "sysemu.h"
30
31 #define AUDIO_CAP "audio"
32 #include "audio_int.h"
33 #include "android/utils/system.h"
34 #include "qemu_debug.h"
35 #include "android/android.h"
36
37 /* #define DEBUG_PLIVE */
38 /* #define DEBUG_LIVE */
39 /* #define DEBUG_OUT */
40 /* #define DEBUG_CAPTURE */
41
42 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
43
44 static struct audio_driver *drvtab[] = {
45 #ifdef CONFIG_ESD
46 &esd_audio_driver,
47 #endif
48 #ifdef CONFIG_ALSA
49 &alsa_audio_driver,
50 #endif
51 #ifdef CONFIG_COREAUDIO
52 &coreaudio_audio_driver,
53 #endif
54 #ifdef CONFIG_DSOUND
55 &dsound_audio_driver,
56 #endif
57 #ifdef CONFIG_FMOD
58 &fmod_audio_driver,
59 #endif
60 #ifdef CONFIG_WINAUDIO
61 &win_audio_driver,
62 #endif
63 #ifdef CONFIG_SDL
64 &sdl_audio_driver,
65 #endif
66 #ifdef CONFIG_OSS
67 &oss_audio_driver,
68 #endif
69 &no_audio_driver,
70 #if 0 /* disabled WAV audio for now - until we find a user-friendly way to use it */
71 &wav_audio_driver
72 #endif
73 };
74
75
76 int
audio_get_backend_count(int is_input)77 audio_get_backend_count( int is_input )
78 {
79 int nn, count = 0;
80
81 for (nn = 0; nn < sizeof(drvtab)/sizeof(drvtab[0]); nn++)
82 {
83 if (is_input) {
84 if ( drvtab[nn]->max_voices_in > 0 )
85 count += 1;
86 } else {
87 if ( drvtab[nn]->max_voices_out > 0 )
88 count += 1;
89 }
90 }
91 return count;
92 }
93
94 const char*
audio_get_backend_name(int is_input,int index,const char ** pinfo)95 audio_get_backend_name( int is_input, int index, const char* *pinfo )
96 {
97 int nn;
98
99 index += 1;
100 for (nn = 0; nn < sizeof(drvtab)/sizeof(drvtab[0]); nn++)
101 {
102 if (is_input) {
103 if ( drvtab[nn]->max_voices_in > 0 ) {
104 if ( --index == 0 ) {
105 *pinfo = drvtab[nn]->descr;
106 return drvtab[nn]->name;
107 }
108 }
109 } else {
110 if ( drvtab[nn]->max_voices_out > 0 ) {
111 if ( --index == 0 ) {
112 *pinfo = drvtab[nn]->descr;
113 return drvtab[nn]->name;
114 }
115 }
116 }
117 }
118 *pinfo = NULL;
119 return NULL;
120 }
121
122
123 int
audio_check_backend_name(int is_input,const char * name)124 audio_check_backend_name( int is_input, const char* name )
125 {
126 int nn;
127
128 for (nn = 0; nn < sizeof(drvtab)/sizeof(drvtab[0]); nn++)
129 {
130 if ( !strcmp(drvtab[nn]->name, name) ) {
131 if (is_input) {
132 if (drvtab[nn]->max_voices_in > 0)
133 return 1;
134 } else {
135 if (drvtab[nn]->max_voices_out > 0)
136 return 1;
137 }
138 break;
139 }
140 }
141 return 0;
142 }
143
144
145 struct fixed_settings {
146 int enabled;
147 int nb_voices;
148 int greedy;
149 struct audsettings settings;
150 };
151
152 static struct {
153 struct fixed_settings fixed_out;
154 struct fixed_settings fixed_in;
155 union {
156 int hertz;
157 int64_t ticks;
158 } period;
159 int plive;
160 int log_to_monitor;
161 } conf = {
162 { /* DAC fixed settings */
163 1, /* enabled */
164 1, /* nb_voices */
165 1, /* greedy */
166 {
167 44100, /* freq */
168 2, /* nchannels */
169 AUD_FMT_S16, /* fmt */
170 AUDIO_HOST_ENDIANNESS
171 }
172 },
173
174 { /* ADC fixed settings */
175 1, /* enabled */
176 1, /* nb_voices */
177 1, /* greedy */
178 {
179 44100, /* freq */
180 2, /* nchannels */
181 AUD_FMT_S16, /* fmt */
182 AUDIO_HOST_ENDIANNESS
183 }
184 },
185
186 { 250 }, /* period */
187 0, /* plive */
188 0 /* log_to_monitor */
189 };
190
191 static AudioState glob_audio_state;
192
193 struct mixeng_volume nominal_volume = {
194 0,
195 #ifdef FLOAT_MIXENG
196 1.0,
197 1.0
198 #else
199 1ULL << 32,
200 1ULL << 32
201 #endif
202 };
203
204 /* http://www.df.lth.se/~john_e/gems/gem002d.html */
205 /* http://www.multi-platforms.com/Tips/PopCount.htm */
popcount(uint32_t u)206 uint32_t popcount (uint32_t u)
207 {
208 u = ((u&0x55555555) + ((u>>1)&0x55555555));
209 u = ((u&0x33333333) + ((u>>2)&0x33333333));
210 u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
211 u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
212 u = ( u&0x0000ffff) + (u>>16);
213 return u;
214 }
215
lsbindex(uint32_t u)216 inline uint32_t lsbindex (uint32_t u)
217 {
218 return popcount ((u&-u)-1);
219 }
220
221 #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
222 #error No its not
223 #else
audio_bug(const char * funcname,int cond)224 int audio_bug (const char *funcname, int cond)
225 {
226 if (cond) {
227 static int shown;
228
229 AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
230 if (!shown) {
231 shown = 1;
232 AUD_log (NULL, "Save all your work and restart without audio\n");
233 AUD_log (NULL, "Please send bug report to malc@pulsesoft.com\n");
234 AUD_log (NULL, "I am sorry\n");
235 }
236 AUD_log (NULL, "Context:\n");
237
238 #if defined AUDIO_BREAKPOINT_ON_BUG
239 # if defined HOST_I386
240 # if defined __GNUC__
241 __asm__ ("int3");
242 # elif defined _MSC_VER
243 _asm _emit 0xcc;
244 # else
245 abort ();
246 # endif
247 # else
248 abort ();
249 # endif
250 #endif
251 }
252
253 return cond;
254 }
255 #endif
256
audio_bits_to_index(int bits)257 static inline int audio_bits_to_index (int bits)
258 {
259 switch (bits) {
260 case 8:
261 return 0;
262
263 case 16:
264 return 1;
265
266 case 32:
267 return 2;
268
269 default:
270 audio_bug ("bits_to_index", 1);
271 AUD_log (NULL, "invalid bits %d\n", bits);
272 return 0;
273 }
274 }
275
audio_calloc(const char * funcname,int nmemb,size_t size)276 void *audio_calloc (const char *funcname, int nmemb, size_t size)
277 {
278 int cond;
279 size_t len;
280
281 len = nmemb * size;
282 cond = !nmemb || !size;
283 cond |= nmemb < 0;
284 cond |= len < size;
285
286 if (audio_bug ("audio_calloc", cond)) {
287 AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
288 funcname);
289 AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
290 return NULL;
291 }
292
293 return qemu_mallocz (len);
294 }
295
audio_alloc_prefix(const char * s)296 static char *audio_alloc_prefix (const char *s)
297 {
298 const char qemu_prefix[] = "QEMU_";
299 size_t len, i;
300 char *r, *u;
301
302 if (!s) {
303 return NULL;
304 }
305
306 len = strlen (s);
307 r = qemu_malloc (len + sizeof (qemu_prefix));
308
309 u = r + sizeof (qemu_prefix) - 1;
310
311 pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
312 pstrcat (r, len + sizeof (qemu_prefix), s);
313
314 for (i = 0; i < len; ++i) {
315 u[i] = qemu_toupper(u[i]);
316 }
317
318 return r;
319 }
320
audio_audfmt_to_string(audfmt_e fmt)321 static const char *audio_audfmt_to_string (audfmt_e fmt)
322 {
323 switch (fmt) {
324 case AUD_FMT_U8:
325 return "U8";
326
327 case AUD_FMT_U16:
328 return "U16";
329
330 case AUD_FMT_S8:
331 return "S8";
332
333 case AUD_FMT_S16:
334 return "S16";
335
336 case AUD_FMT_U32:
337 return "U32";
338
339 case AUD_FMT_S32:
340 return "S32";
341 }
342
343 dolog ("Bogus audfmt %d returning S16\n", fmt);
344 return "S16";
345 }
346
audio_string_to_audfmt(const char * s,audfmt_e defval,int * defaultp)347 static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
348 int *defaultp)
349 {
350 if (!strcasecmp (s, "u8")) {
351 *defaultp = 0;
352 return AUD_FMT_U8;
353 }
354 else if (!strcasecmp (s, "u16")) {
355 *defaultp = 0;
356 return AUD_FMT_U16;
357 }
358 else if (!strcasecmp (s, "u32")) {
359 *defaultp = 0;
360 return AUD_FMT_U32;
361 }
362 else if (!strcasecmp (s, "s8")) {
363 *defaultp = 0;
364 return AUD_FMT_S8;
365 }
366 else if (!strcasecmp (s, "s16")) {
367 *defaultp = 0;
368 return AUD_FMT_S16;
369 }
370 else if (!strcasecmp (s, "s32")) {
371 *defaultp = 0;
372 return AUD_FMT_S32;
373 }
374 else {
375 dolog ("Bogus audio format `%s' using %s\n",
376 s, audio_audfmt_to_string (defval));
377 *defaultp = 1;
378 return defval;
379 }
380 }
381
audio_get_conf_fmt(const char * envname,audfmt_e defval,int * defaultp)382 static audfmt_e audio_get_conf_fmt (const char *envname,
383 audfmt_e defval,
384 int *defaultp)
385 {
386 const char *var = getenv (envname);
387 if (!var) {
388 *defaultp = 1;
389 return defval;
390 }
391 return audio_string_to_audfmt (var, defval, defaultp);
392 }
393
audio_get_conf_int(const char * key,int defval,int * defaultp)394 static int audio_get_conf_int (const char *key, int defval, int *defaultp)
395 {
396 int val;
397 char *strval;
398
399 strval = getenv (key);
400 if (strval) {
401 *defaultp = 0;
402 val = atoi (strval);
403 return val;
404 }
405 else {
406 *defaultp = 1;
407 return defval;
408 }
409 }
410
audio_get_conf_str(const char * key,const char * defval,int * defaultp)411 static const char *audio_get_conf_str (const char *key,
412 const char *defval,
413 int *defaultp)
414 {
415 const char *val = getenv (key);
416 if (!val) {
417 *defaultp = 1;
418 return defval;
419 }
420 else {
421 *defaultp = 0;
422 return val;
423 }
424 }
425
426 /* defined in android_sdl.c */
427 extern void dprintn(const char* fmt, ...);
428 extern void dprintnv(const char* fmt, va_list args);
429
AUD_vlog(const char * cap,const char * fmt,va_list ap)430 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
431 {
432 if (conf.log_to_monitor) {
433 if (cap) {
434 monitor_printf(cur_mon, "%s: ", cap);
435 }
436
437 monitor_vprintf(cur_mon, fmt, ap);
438 }
439 else {
440 if (!VERBOSE_CHECK(audio))
441 return;
442
443 if (cap) {
444 dprintn("%s: ", cap);
445 }
446
447 dprintnv(fmt, ap);
448 }
449 }
450
AUD_log(const char * cap,const char * fmt,...)451 void AUD_log (const char *cap, const char *fmt, ...)
452 {
453 va_list ap;
454
455 va_start (ap, fmt);
456 AUD_vlog (cap, fmt, ap);
457 va_end (ap);
458 }
459
audio_print_options(const char * prefix,struct audio_option * opt)460 static void audio_print_options (const char *prefix,
461 struct audio_option *opt)
462 {
463 char *uprefix;
464
465 if (!prefix) {
466 dolog ("No prefix specified\n");
467 return;
468 }
469
470 if (!opt) {
471 dolog ("No options\n");
472 return;
473 }
474
475 uprefix = audio_alloc_prefix (prefix);
476
477 for (; opt->name; opt++) {
478 const char *state = "default";
479 printf (" %s_%s: ", uprefix, opt->name);
480
481 if (opt->overriddenp && *opt->overriddenp) {
482 state = "current";
483 }
484
485 switch (opt->tag) {
486 case AUD_OPT_BOOL:
487 {
488 int *intp = opt->valp;
489 printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
490 }
491 break;
492
493 case AUD_OPT_INT:
494 {
495 int *intp = opt->valp;
496 printf ("integer, %s = %d\n", state, *intp);
497 }
498 break;
499
500 case AUD_OPT_FMT:
501 {
502 audfmt_e *fmtp = opt->valp;
503 printf (
504 "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
505 state,
506 audio_audfmt_to_string (*fmtp)
507 );
508 }
509 break;
510
511 case AUD_OPT_STR:
512 {
513 const char **strp = opt->valp;
514 printf ("string, %s = %s\n",
515 state,
516 *strp ? *strp : "(not set)");
517 }
518 break;
519
520 default:
521 printf ("???\n");
522 dolog ("Bad value tag for option %s_%s %d\n",
523 uprefix, opt->name, opt->tag);
524 break;
525 }
526 printf (" %s\n", opt->descr);
527 }
528
529 qemu_free (uprefix);
530 }
531
audio_process_options(const char * prefix,struct audio_option * opt)532 static void audio_process_options (const char *prefix,
533 struct audio_option *opt)
534 {
535 char *optname;
536 const char qemu_prefix[] = "QEMU_";
537 size_t preflen, optlen;
538
539 if (audio_bug (AUDIO_FUNC, !prefix)) {
540 dolog ("prefix = NULL\n");
541 return;
542 }
543
544 if (audio_bug (AUDIO_FUNC, !opt)) {
545 dolog ("opt = NULL\n");
546 return;
547 }
548
549 preflen = strlen (prefix);
550
551 for (; opt->name; opt++) {
552 size_t len, i;
553 int def;
554
555 if (!opt->valp) {
556 dolog ("Option value pointer for `%s' is not set\n",
557 opt->name);
558 continue;
559 }
560
561 len = strlen (opt->name);
562 /* len of opt->name + len of prefix + size of qemu_prefix
563 * (includes trailing zero) + zero + underscore (on behalf of
564 * sizeof) */
565 optlen = len + preflen + sizeof (qemu_prefix) + 1;
566 optname = qemu_malloc (optlen);
567
568 pstrcpy (optname, optlen, qemu_prefix);
569
570 /* copy while upper-casing, including trailing zero */
571 for (i = 0; i <= preflen; ++i) {
572 optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
573 }
574 pstrcat (optname, optlen, "_");
575 pstrcat (optname, optlen, opt->name);
576
577 def = 1;
578 switch (opt->tag) {
579 case AUD_OPT_BOOL:
580 case AUD_OPT_INT:
581 {
582 int *intp = opt->valp;
583 *intp = audio_get_conf_int (optname, *intp, &def);
584 }
585 break;
586
587 case AUD_OPT_FMT:
588 {
589 audfmt_e *fmtp = opt->valp;
590 *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
591 }
592 break;
593
594 case AUD_OPT_STR:
595 {
596 const char **strp = opt->valp;
597 *strp = audio_get_conf_str (optname, *strp, &def);
598 }
599 break;
600
601 default:
602 dolog ("Bad value tag for option `%s' - %d\n",
603 optname, opt->tag);
604 break;
605 }
606
607 if (!opt->overriddenp) {
608 opt->overriddenp = &opt->overridden;
609 }
610 *opt->overriddenp = !def;
611 qemu_free (optname);
612 }
613 }
614
audio_print_settings(struct audsettings * as)615 static void audio_print_settings (struct audsettings *as)
616 {
617 dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
618
619 switch (as->fmt) {
620 case AUD_FMT_S8:
621 AUD_log (NULL, "S8");
622 break;
623 case AUD_FMT_U8:
624 AUD_log (NULL, "U8");
625 break;
626 case AUD_FMT_S16:
627 AUD_log (NULL, "S16");
628 break;
629 case AUD_FMT_U16:
630 AUD_log (NULL, "U16");
631 break;
632 case AUD_FMT_S32:
633 AUD_log (NULL, "S32");
634 break;
635 case AUD_FMT_U32:
636 AUD_log (NULL, "U32");
637 break;
638 default:
639 AUD_log (NULL, "invalid(%d)", as->fmt);
640 break;
641 }
642
643 AUD_log (NULL, " endianness=");
644 switch (as->endianness) {
645 case 0:
646 AUD_log (NULL, "little");
647 break;
648 case 1:
649 AUD_log (NULL, "big");
650 break;
651 default:
652 AUD_log (NULL, "invalid");
653 break;
654 }
655 AUD_log (NULL, "\n");
656 }
657
audio_validate_settings(struct audsettings * as)658 static int audio_validate_settings (struct audsettings *as)
659 {
660 int invalid;
661
662 invalid = as->nchannels != 1 && as->nchannels != 2;
663 invalid |= as->endianness != 0 && as->endianness != 1;
664
665 switch (as->fmt) {
666 case AUD_FMT_S8:
667 case AUD_FMT_U8:
668 case AUD_FMT_S16:
669 case AUD_FMT_U16:
670 case AUD_FMT_S32:
671 case AUD_FMT_U32:
672 break;
673 default:
674 invalid = 1;
675 break;
676 }
677
678 invalid |= as->freq <= 0;
679 return invalid ? -1 : 0;
680 }
681
audio_pcm_info_eq(struct audio_pcm_info * info,struct audsettings * as)682 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
683 {
684 int bits = 8, sign = 0;
685
686 switch (as->fmt) {
687 case AUD_FMT_S8:
688 sign = 1;
689 case AUD_FMT_U8:
690 break;
691
692 case AUD_FMT_S16:
693 sign = 1;
694 case AUD_FMT_U16:
695 bits = 16;
696 break;
697
698 case AUD_FMT_S32:
699 sign = 1;
700 case AUD_FMT_U32:
701 bits = 32;
702 break;
703 }
704 return info->freq == as->freq
705 && info->nchannels == as->nchannels
706 && info->sign == sign
707 && info->bits == bits
708 && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
709 }
710
audio_pcm_init_info(struct audio_pcm_info * info,struct audsettings * as)711 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
712 {
713 int bits = 8, sign = 0, shift = 0;
714
715 switch (as->fmt) {
716 case AUD_FMT_S8:
717 sign = 1;
718 case AUD_FMT_U8:
719 break;
720
721 case AUD_FMT_S16:
722 sign = 1;
723 case AUD_FMT_U16:
724 bits = 16;
725 shift = 1;
726 break;
727
728 case AUD_FMT_S32:
729 sign = 1;
730 case AUD_FMT_U32:
731 bits = 32;
732 shift = 2;
733 break;
734 }
735
736 info->freq = as->freq;
737 info->bits = bits;
738 info->sign = sign;
739 info->nchannels = as->nchannels;
740 info->shift = (as->nchannels == 2) + shift;
741 info->align = (1 << info->shift) - 1;
742 info->bytes_per_second = info->freq << info->shift;
743 info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
744 }
745
audio_pcm_info_clear_buf(struct audio_pcm_info * info,void * buf,int len)746 void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
747 {
748 if (!len) {
749 return;
750 }
751
752 if (info->sign) {
753 memset (buf, 0x00, len << info->shift);
754 }
755 else {
756 switch (info->bits) {
757 case 8:
758 memset (buf, 0x80, len << info->shift);
759 break;
760
761 case 16:
762 {
763 int i;
764 uint16_t *p = buf;
765 int shift = info->nchannels - 1;
766 short s = INT16_MAX;
767
768 if (info->swap_endianness) {
769 s = bswap16 (s);
770 }
771
772 for (i = 0; i < len << shift; i++) {
773 p[i] = s;
774 }
775 }
776 break;
777
778 case 32:
779 {
780 int i;
781 uint32_t *p = buf;
782 int shift = info->nchannels - 1;
783 int32_t s = INT32_MAX;
784
785 if (info->swap_endianness) {
786 s = bswap32 (s);
787 }
788
789 for (i = 0; i < len << shift; i++) {
790 p[i] = s;
791 }
792 }
793 break;
794
795 default:
796 AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
797 info->bits);
798 break;
799 }
800 }
801 }
802
803 /*
804 * Capture
805 */
noop_conv(struct st_sample * dst,const void * src,int samples,struct mixeng_volume * vol)806 static void noop_conv (struct st_sample *dst, const void *src,
807 int samples, struct mixeng_volume *vol)
808 {
809 (void) src;
810 (void) dst;
811 (void) samples;
812 (void) vol;
813 }
814
audio_pcm_capture_find_specific(struct audsettings * as)815 static CaptureVoiceOut *audio_pcm_capture_find_specific (
816 struct audsettings *as
817 )
818 {
819 CaptureVoiceOut *cap;
820 AudioState *s = &glob_audio_state;
821
822 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
823 if (audio_pcm_info_eq (&cap->hw.info, as)) {
824 return cap;
825 }
826 }
827 return NULL;
828 }
829
audio_notify_capture(CaptureVoiceOut * cap,audcnotification_e cmd)830 static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
831 {
832 struct capture_callback *cb;
833
834 #ifdef DEBUG_CAPTURE
835 dolog ("notification %d sent\n", cmd);
836 #endif
837 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
838 cb->ops.notify (cb->opaque, cmd);
839 }
840 }
841
audio_capture_maybe_changed(CaptureVoiceOut * cap,int enabled)842 static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
843 {
844 if (cap->hw.enabled != enabled) {
845 audcnotification_e cmd;
846 cap->hw.enabled = enabled;
847 cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
848 audio_notify_capture (cap, cmd);
849 }
850 }
851
audio_recalc_and_notify_capture(CaptureVoiceOut * cap)852 static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
853 {
854 HWVoiceOut *hw = &cap->hw;
855 SWVoiceOut *sw;
856 int enabled = 0;
857
858 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
859 if (sw->active) {
860 enabled = 1;
861 break;
862 }
863 }
864 audio_capture_maybe_changed (cap, enabled);
865 }
866
audio_detach_capture(HWVoiceOut * hw)867 static void audio_detach_capture (HWVoiceOut *hw)
868 {
869 SWVoiceCap *sc = hw->cap_head.lh_first;
870
871 while (sc) {
872 SWVoiceCap *sc1 = sc->entries.le_next;
873 SWVoiceOut *sw = &sc->sw;
874 CaptureVoiceOut *cap = sc->cap;
875 int was_active = sw->active;
876
877 if (sw->rate) {
878 st_rate_stop (sw->rate);
879 sw->rate = NULL;
880 }
881
882 LIST_REMOVE (sw, entries);
883 LIST_REMOVE (sc, entries);
884 qemu_free (sc);
885 if (was_active) {
886 /* We have removed soft voice from the capture:
887 this might have changed the overall status of the capture
888 since this might have been the only active voice */
889 audio_recalc_and_notify_capture (cap);
890 }
891 sc = sc1;
892 }
893 }
894
audio_attach_capture(HWVoiceOut * hw)895 static int audio_attach_capture (HWVoiceOut *hw)
896 {
897 AudioState *s = &glob_audio_state;
898 CaptureVoiceOut *cap;
899
900 audio_detach_capture (hw);
901 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
902 SWVoiceCap *sc;
903 SWVoiceOut *sw;
904 HWVoiceOut *hw_cap = &cap->hw;
905
906 sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
907 if (!sc) {
908 dolog ("Could not allocate soft capture voice (%zu bytes)\n",
909 sizeof (*sc));
910 return -1;
911 }
912
913 sc->cap = cap;
914 sw = &sc->sw;
915 sw->hw = hw_cap;
916 sw->info = hw->info;
917 sw->empty = 1;
918 sw->active = hw->enabled;
919 sw->conv = noop_conv;
920 sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
921 sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
922 if (!sw->rate) {
923 dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
924 qemu_free (sw);
925 return -1;
926 }
927 LIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
928 LIST_INSERT_HEAD (&hw->cap_head, sc, entries);
929 #ifdef DEBUG_CAPTURE
930 asprintf (&sw->name, "for %p %d,%d,%d",
931 hw, sw->info.freq, sw->info.bits, sw->info.nchannels);
932 dolog ("Added %s active = %d\n", sw->name, sw->active);
933 #endif
934 if (sw->active) {
935 audio_capture_maybe_changed (cap, 1);
936 }
937 }
938 return 0;
939 }
940
941 /*
942 * Hard voice (capture)
943 */
audio_pcm_hw_find_min_in(HWVoiceIn * hw)944 static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
945 {
946 SWVoiceIn *sw;
947 int m = hw->total_samples_captured;
948
949 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
950 if (sw->active) {
951 m = audio_MIN (m, sw->total_hw_samples_acquired);
952 }
953 }
954 return m;
955 }
956
audio_pcm_hw_get_live_in(HWVoiceIn * hw)957 int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
958 {
959 int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
960 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
961 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
962 return 0;
963 }
964 return live;
965 }
966
967 /*
968 * Soft voice (capture)
969 */
audio_pcm_sw_get_rpos_in(SWVoiceIn * sw)970 static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
971 {
972 HWVoiceIn *hw = sw->hw;
973 int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
974 int rpos;
975
976 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
977 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
978 return 0;
979 }
980
981 rpos = hw->wpos - live;
982 if (rpos >= 0) {
983 return rpos;
984 }
985 else {
986 return hw->samples + rpos;
987 }
988 }
989
audio_pcm_sw_read(SWVoiceIn * sw,void * buf,int size)990 int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
991 {
992 HWVoiceIn *hw = sw->hw;
993 int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
994 struct st_sample *src, *dst = sw->buf;
995
996 rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
997
998 live = hw->total_samples_captured - sw->total_hw_samples_acquired;
999 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
1000 dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
1001 return 0;
1002 }
1003
1004 samples = size >> sw->info.shift;
1005 if (!live) {
1006 return 0;
1007 }
1008
1009 swlim = (live * sw->ratio) >> 32;
1010 swlim = audio_MIN (swlim, samples);
1011
1012 while (swlim) {
1013 src = hw->conv_buf + rpos;
1014 isamp = hw->wpos - rpos;
1015 /* XXX: <= ? */
1016 if (isamp <= 0) {
1017 isamp = hw->samples - rpos;
1018 }
1019
1020 if (!isamp) {
1021 break;
1022 }
1023 osamp = swlim;
1024
1025 if (audio_bug (AUDIO_FUNC, osamp < 0)) {
1026 dolog ("osamp=%d\n", osamp);
1027 return 0;
1028 }
1029
1030 st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
1031 swlim -= osamp;
1032 rpos = (rpos + isamp) % hw->samples;
1033 dst += osamp;
1034 ret += osamp;
1035 total += isamp;
1036 }
1037
1038 sw->clip (buf, sw->buf, ret);
1039 sw->total_hw_samples_acquired += total;
1040 return ret << sw->info.shift;
1041 }
1042
1043 /*
1044 * Hard voice (playback)
1045 */
audio_pcm_hw_find_min_out(HWVoiceOut * hw,int * nb_livep)1046 static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
1047 {
1048 SWVoiceOut *sw;
1049 int m = INT_MAX;
1050 int nb_live = 0;
1051
1052 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1053 if (sw->active || !sw->empty) {
1054 m = audio_MIN (m, sw->total_hw_samples_mixed);
1055 nb_live += 1;
1056 }
1057 }
1058
1059 *nb_livep = nb_live;
1060 return m;
1061 }
1062
audio_pcm_hw_get_live_out2(HWVoiceOut * hw,int * nb_live)1063 int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
1064 {
1065 int smin;
1066
1067 smin = audio_pcm_hw_find_min_out (hw, nb_live);
1068
1069 if (!*nb_live) {
1070 return 0;
1071 }
1072 else {
1073 int live = smin;
1074
1075 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
1076 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
1077 return 0;
1078 }
1079 return live;
1080 }
1081 }
1082
audio_pcm_hw_get_live_out(HWVoiceOut * hw)1083 int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
1084 {
1085 int nb_live;
1086 int live;
1087
1088 live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
1089 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
1090 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
1091 return 0;
1092 }
1093 return live;
1094 }
1095
1096 /*
1097 * Soft voice (playback)
1098 */
audio_pcm_sw_write(SWVoiceOut * sw,void * buf,int size)1099 int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
1100 {
1101 int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
1102 int ret = 0, pos = 0, total = 0;
1103
1104 if (!sw) {
1105 return size;
1106 }
1107
1108 hwsamples = sw->hw->samples;
1109
1110 live = sw->total_hw_samples_mixed;
1111 if (audio_bug (AUDIO_FUNC, live < 0 || live > hwsamples)){
1112 dolog ("live=%d hw->samples=%d\n", live, hwsamples);
1113 return 0;
1114 }
1115
1116 if (live == hwsamples) {
1117 #ifdef DEBUG_OUT
1118 dolog ("%s is full %d\n", sw->name, live);
1119 #endif
1120 return 0;
1121 }
1122
1123 wpos = (sw->hw->rpos + live) % hwsamples;
1124 samples = size >> sw->info.shift;
1125
1126 dead = hwsamples - live;
1127 swlim = ((int64_t) dead << 32) / sw->ratio;
1128 swlim = audio_MIN (swlim, samples);
1129 if (swlim) {
1130 sw->conv (sw->buf, buf, swlim, &sw->vol);
1131 }
1132
1133 while (swlim) {
1134 dead = hwsamples - live;
1135 left = hwsamples - wpos;
1136 blck = audio_MIN (dead, left);
1137 if (!blck) {
1138 break;
1139 }
1140 isamp = swlim;
1141 osamp = blck;
1142 st_rate_flow_mix (
1143 sw->rate,
1144 sw->buf + pos,
1145 sw->hw->mix_buf + wpos,
1146 &isamp,
1147 &osamp
1148 );
1149 ret += isamp;
1150 swlim -= isamp;
1151 pos += isamp;
1152 live += osamp;
1153 wpos = (wpos + osamp) % hwsamples;
1154 total += osamp;
1155 }
1156
1157 sw->total_hw_samples_mixed += total;
1158 sw->empty = sw->total_hw_samples_mixed == 0;
1159
1160 #ifdef DEBUG_OUT
1161 dolog (
1162 "%s: write size %d ret %d total sw %d\n",
1163 SW_NAME (sw),
1164 size >> sw->info.shift,
1165 ret,
1166 sw->total_hw_samples_mixed
1167 );
1168 #endif
1169
1170 return ret << sw->info.shift;
1171 }
1172
1173 #ifdef DEBUG_AUDIO
audio_pcm_print_info(const char * cap,struct audio_pcm_info * info)1174 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
1175 {
1176 dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
1177 cap, info->bits, info->sign, info->freq, info->nchannels);
1178 }
1179 #endif
1180
1181 #define DAC
1182 #include "audio_template.h"
1183 #undef DAC
1184 #include "audio_template.h"
1185
AUD_write(SWVoiceOut * sw,void * buf,int size)1186 int AUD_write (SWVoiceOut *sw, void *buf, int size)
1187 {
1188 int bytes;
1189
1190 if (!sw) {
1191 /* XXX: Consider options */
1192 return size;
1193 }
1194
1195 if (!sw->hw->enabled) {
1196 dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
1197 return 0;
1198 }
1199
1200 BEGIN_NOSIGALRM
1201 bytes = sw->hw->pcm_ops->write (sw, buf, size);
1202 END_NOSIGALRM
1203 return bytes;
1204 }
1205
AUD_read(SWVoiceIn * sw,void * buf,int size)1206 int AUD_read (SWVoiceIn *sw, void *buf, int size)
1207 {
1208 int bytes;
1209
1210 if (!sw) {
1211 /* XXX: Consider options */
1212 return size;
1213 }
1214
1215 if (!sw->hw->enabled) {
1216 dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
1217 return 0;
1218 }
1219
1220 BEGIN_NOSIGALRM
1221 bytes = sw->hw->pcm_ops->read (sw, buf, size);
1222 END_NOSIGALRM
1223 return bytes;
1224 }
1225
AUD_get_buffer_size_out(SWVoiceOut * sw)1226 int AUD_get_buffer_size_out (SWVoiceOut *sw)
1227 {
1228 return sw->hw->samples << sw->hw->info.shift;
1229 }
1230
AUD_set_active_out(SWVoiceOut * sw,int on)1231 void AUD_set_active_out (SWVoiceOut *sw, int on)
1232 {
1233 HWVoiceOut *hw;
1234
1235 if (!sw) {
1236 return;
1237 }
1238
1239 hw = sw->hw;
1240 if (sw->active != on) {
1241 AudioState *s = &glob_audio_state;
1242 SWVoiceOut *temp_sw;
1243 SWVoiceCap *sc;
1244
1245 if (on) {
1246 hw->pending_disable = 0;
1247 if (!hw->enabled) {
1248 hw->enabled = 1;
1249 if (s->vm_running) {
1250 BEGIN_NOSIGALRM
1251 hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
1252 END_NOSIGALRM
1253 }
1254 }
1255 }
1256 else {
1257 if (hw->enabled) {
1258 int nb_active = 0;
1259
1260 for (temp_sw = hw->sw_head.lh_first; temp_sw;
1261 temp_sw = temp_sw->entries.le_next) {
1262 nb_active += temp_sw->active != 0;
1263 }
1264
1265 hw->pending_disable = nb_active == 1;
1266 }
1267 }
1268
1269 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1270 sc->sw.active = hw->enabled;
1271 if (hw->enabled) {
1272 audio_capture_maybe_changed (sc->cap, 1);
1273 }
1274 }
1275 sw->active = on;
1276 }
1277 }
1278
AUD_set_active_in(SWVoiceIn * sw,int on)1279 void AUD_set_active_in (SWVoiceIn *sw, int on)
1280 {
1281 HWVoiceIn *hw;
1282
1283 if (!sw) {
1284 return;
1285 }
1286
1287 hw = sw->hw;
1288 if (sw->active != on) {
1289 AudioState *s = &glob_audio_state;
1290 SWVoiceIn *temp_sw;
1291
1292 if (on) {
1293 if (!hw->enabled) {
1294 hw->enabled = 1;
1295 if (s->vm_running) {
1296 BEGIN_NOSIGALRM
1297 hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
1298 END_NOSIGALRM
1299 }
1300 }
1301 sw->total_hw_samples_acquired = hw->total_samples_captured;
1302 }
1303 else {
1304 if (hw->enabled) {
1305 int nb_active = 0;
1306
1307 for (temp_sw = hw->sw_head.lh_first; temp_sw;
1308 temp_sw = temp_sw->entries.le_next) {
1309 nb_active += temp_sw->active != 0;
1310 }
1311
1312 if (nb_active == 1) {
1313 hw->enabled = 0;
1314 BEGIN_NOSIGALRM
1315 hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
1316 END_NOSIGALRM
1317 }
1318 }
1319 }
1320 sw->active = on;
1321 }
1322 }
1323
audio_get_avail(SWVoiceIn * sw)1324 static int audio_get_avail (SWVoiceIn *sw)
1325 {
1326 int live;
1327
1328 if (!sw) {
1329 return 0;
1330 }
1331
1332 live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
1333 if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
1334 dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
1335 return 0;
1336 }
1337
1338 ldebug (
1339 "%s: get_avail live %d ret %" PRId64 "\n",
1340 SW_NAME (sw),
1341 live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
1342 );
1343
1344 return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
1345 }
1346
audio_get_free(SWVoiceOut * sw)1347 static int audio_get_free (SWVoiceOut *sw)
1348 {
1349 int live, dead;
1350
1351 if (!sw) {
1352 return 0;
1353 }
1354
1355 live = sw->total_hw_samples_mixed;
1356
1357 if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
1358 dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
1359 return 0;
1360 }
1361
1362 dead = sw->hw->samples - live;
1363
1364 #ifdef DEBUG_OUT
1365 dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
1366 SW_NAME (sw),
1367 live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
1368 #endif
1369
1370 return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
1371 }
1372
audio_capture_mix_and_clear(HWVoiceOut * hw,int rpos,int samples)1373 static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
1374 {
1375 int n;
1376
1377 if (hw->enabled) {
1378 SWVoiceCap *sc;
1379
1380 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1381 SWVoiceOut *sw = &sc->sw;
1382 int rpos2 = rpos;
1383
1384 n = samples;
1385 while (n) {
1386 int till_end_of_hw = hw->samples - rpos2;
1387 int to_write = audio_MIN (till_end_of_hw, n);
1388 int bytes = to_write << hw->info.shift;
1389 int written;
1390
1391 sw->buf = hw->mix_buf + rpos2;
1392 written = audio_pcm_sw_write (sw, NULL, bytes);
1393 if (written - bytes) {
1394 dolog ("Could not mix %d bytes into a capture "
1395 "buffer, mixed %d\n",
1396 bytes, written);
1397 break;
1398 }
1399 n -= to_write;
1400 rpos2 = (rpos2 + to_write) % hw->samples;
1401 }
1402 }
1403 }
1404
1405 n = audio_MIN (samples, hw->samples - rpos);
1406 mixeng_clear (hw->mix_buf + rpos, n);
1407 mixeng_clear (hw->mix_buf, samples - n);
1408 }
1409
audio_run_out(AudioState * s)1410 static void audio_run_out (AudioState *s)
1411 {
1412 HWVoiceOut *hw = NULL;
1413 SWVoiceOut *sw;
1414
1415 while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) {
1416 int played;
1417 int live, free, nb_live, cleanup_required, prev_rpos;
1418
1419 live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
1420 if (!nb_live) {
1421 live = 0;
1422 }
1423
1424 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
1425 dolog ("live=%d hw->samples=%d\n", live, hw->samples);
1426 continue;
1427 }
1428
1429 if (hw->pending_disable && !nb_live) {
1430 SWVoiceCap *sc;
1431 #ifdef DEBUG_OUT
1432 dolog ("Disabling voice\n");
1433 #endif
1434 hw->enabled = 0;
1435 hw->pending_disable = 0;
1436 BEGIN_NOSIGALRM
1437 hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
1438 END_NOSIGALRM
1439 for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1440 sc->sw.active = 0;
1441 audio_recalc_and_notify_capture (sc->cap);
1442 }
1443 continue;
1444 }
1445
1446 if (!live) {
1447 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1448 if (sw->active) {
1449 free = audio_get_free (sw);
1450 if (free > 0) {
1451 sw->callback.fn (sw->callback.opaque, free);
1452 }
1453 }
1454 }
1455 continue;
1456 }
1457
1458 prev_rpos = hw->rpos;
1459 BEGIN_NOSIGALRM
1460 played = hw->pcm_ops->run_out (hw);
1461 END_NOSIGALRM
1462 if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
1463 dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
1464 hw->rpos, hw->samples, played);
1465 hw->rpos = 0;
1466 }
1467
1468 #ifdef DEBUG_OUT
1469 dolog ("played=%d\n", played);
1470 #endif
1471
1472 if (played) {
1473 hw->ts_helper += played;
1474 audio_capture_mix_and_clear (hw, prev_rpos, played);
1475 }
1476
1477 cleanup_required = 0;
1478 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1479 if (!sw->active && sw->empty) {
1480 continue;
1481 }
1482
1483 if (audio_bug (AUDIO_FUNC, played > sw->total_hw_samples_mixed)) {
1484 dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
1485 played, sw->total_hw_samples_mixed);
1486 played = sw->total_hw_samples_mixed;
1487 }
1488
1489 sw->total_hw_samples_mixed -= played;
1490
1491 if (!sw->total_hw_samples_mixed) {
1492 sw->empty = 1;
1493 cleanup_required |= !sw->active && !sw->callback.fn;
1494 }
1495
1496 if (sw->active) {
1497 free = audio_get_free (sw);
1498 if (free > 0) {
1499 sw->callback.fn (sw->callback.opaque, free);
1500 }
1501 }
1502 }
1503
1504 if (cleanup_required) {
1505 SWVoiceOut *sw1;
1506
1507 sw = hw->sw_head.lh_first;
1508 while (sw) {
1509 sw1 = sw->entries.le_next;
1510 if (!sw->active && !sw->callback.fn) {
1511 #ifdef DEBUG_PLIVE
1512 dolog ("Finishing with old voice\n");
1513 #endif
1514 audio_close_out (sw);
1515 }
1516 sw = sw1;
1517 }
1518 }
1519 }
1520 }
1521
audio_run_in(AudioState * s)1522 static void audio_run_in (AudioState *s)
1523 {
1524 HWVoiceIn *hw = NULL;
1525
1526 while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
1527 SWVoiceIn *sw;
1528 int captured, min;
1529
1530 BEGIN_NOSIGALRM
1531 captured = hw->pcm_ops->run_in (hw);
1532 END_NOSIGALRM
1533
1534 min = audio_pcm_hw_find_min_in (hw);
1535 hw->total_samples_captured += captured - min;
1536 hw->ts_helper += captured;
1537
1538 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1539 sw->total_hw_samples_acquired -= min;
1540
1541 if (sw->active) {
1542 int avail;
1543
1544 avail = audio_get_avail (sw);
1545 if (avail > 0) {
1546 sw->callback.fn (sw->callback.opaque, avail);
1547 }
1548 }
1549 }
1550 }
1551 }
1552
audio_run_capture(AudioState * s)1553 static void audio_run_capture (AudioState *s)
1554 {
1555 CaptureVoiceOut *cap;
1556
1557 for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
1558 int live, rpos, captured;
1559 HWVoiceOut *hw = &cap->hw;
1560 SWVoiceOut *sw;
1561
1562 captured = live = audio_pcm_hw_get_live_out (hw);
1563 rpos = hw->rpos;
1564 while (live) {
1565 int left = hw->samples - rpos;
1566 int to_capture = audio_MIN (live, left);
1567 struct st_sample *src;
1568 struct capture_callback *cb;
1569
1570 src = hw->mix_buf + rpos;
1571 hw->clip (cap->buf, src, to_capture);
1572 mixeng_clear (src, to_capture);
1573
1574 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1575 cb->ops.capture (cb->opaque, cap->buf,
1576 to_capture << hw->info.shift);
1577 }
1578 rpos = (rpos + to_capture) % hw->samples;
1579 live -= to_capture;
1580 }
1581 hw->rpos = rpos;
1582
1583 for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1584 if (!sw->active && sw->empty) {
1585 continue;
1586 }
1587
1588 if (audio_bug (AUDIO_FUNC, captured > sw->total_hw_samples_mixed)) {
1589 dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
1590 captured, sw->total_hw_samples_mixed);
1591 captured = sw->total_hw_samples_mixed;
1592 }
1593
1594 sw->total_hw_samples_mixed -= captured;
1595 sw->empty = sw->total_hw_samples_mixed == 0;
1596 }
1597 }
1598 }
1599
audio_timer(void * opaque)1600 static void audio_timer (void *opaque)
1601 {
1602 AudioState *s = opaque;
1603 #if 0
1604 #define MAX_DIFFS 1000
1605 int64_t now = qemu_get_clock(vm_clock);
1606 static int64_t last = 0;
1607 static float diffs[MAX_DIFFS];
1608 static int num_diffs;
1609
1610 if (last == 0)
1611 last = now;
1612 else {
1613 diffs[num_diffs] = (float)((now-last)/1e6); /* last diff in ms */
1614 if (++num_diffs == MAX_DIFFS) {
1615 double min_diff = 1e6, max_diff = -1e6;
1616 double all_diff = 0.;
1617 int nn;
1618
1619 for (nn = 0; nn < num_diffs; nn++) {
1620 if (diffs[nn] < min_diff) min_diff = diffs[nn];
1621 if (diffs[nn] > max_diff) max_diff = diffs[nn];
1622 all_diff += diffs[nn];
1623 }
1624 all_diff *= 1.0/num_diffs;
1625 printf("audio timer: min_diff=%6.2g max_diff=%6.2g avg_diff=%6.2g samples=%d\n",
1626 min_diff, max_diff, all_diff, num_diffs);
1627 num_diffs = 0;
1628 }
1629 }
1630 last = now;
1631 #endif
1632 audio_run_out (s);
1633 audio_run_in (s);
1634 audio_run_capture (s);
1635
1636 qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
1637 }
1638
1639 static struct audio_option audio_options[] = {
1640 /* DAC */
1641 {"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_out.enabled,
1642 "Use fixed settings for host DAC", NULL, 0},
1643
1644 {"DAC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_out.settings.freq,
1645 "Frequency for fixed host DAC", NULL, 0},
1646
1647 {"DAC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_out.settings.fmt,
1648 "Format for fixed host DAC", NULL, 0},
1649
1650 {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_out.settings.nchannels,
1651 "Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0},
1652
1653 {"DAC_VOICES", AUD_OPT_INT, &conf.fixed_out.nb_voices,
1654 "Number of voices for DAC", NULL, 0},
1655
1656 /* ADC */
1657 {"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_in.enabled,
1658 "Use fixed settings for host ADC", NULL, 0},
1659
1660 {"ADC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_in.settings.freq,
1661 "Frequency for fixed host ADC", NULL, 0},
1662
1663 {"ADC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_in.settings.fmt,
1664 "Format for fixed host ADC", NULL, 0},
1665
1666 {"ADC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_in.settings.nchannels,
1667 "Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0},
1668
1669 {"ADC_VOICES", AUD_OPT_INT, &conf.fixed_in.nb_voices,
1670 "Number of voices for ADC", NULL, 0},
1671
1672 /* Misc */
1673 {"TIMER_PERIOD", AUD_OPT_INT, &conf.period.hertz,
1674 "Timer period in HZ (0 - use lowest possible)", NULL, 0},
1675
1676 {"PLIVE", AUD_OPT_BOOL, &conf.plive,
1677 "(undocumented)", NULL, 0},
1678
1679 {"LOG_TO_MONITOR", AUD_OPT_BOOL, &conf.log_to_monitor,
1680 "print logging messages to monitor instead of stderr", NULL, 0},
1681
1682 {NULL, 0, NULL, NULL, NULL, 0}
1683 };
1684
audio_pp_nb_voices(const char * typ,int nb)1685 static void audio_pp_nb_voices (const char *typ, int nb)
1686 {
1687 switch (nb) {
1688 case 0:
1689 printf ("Does not support %s\n", typ);
1690 break;
1691 case 1:
1692 printf ("One %s voice\n", typ);
1693 break;
1694 case INT_MAX:
1695 printf ("Theoretically supports many %s voices\n", typ);
1696 break;
1697 default:
1698 printf ("Theoretically supports upto %d %s voices\n", nb, typ);
1699 break;
1700 }
1701
1702 }
1703
AUD_help(void)1704 void AUD_help (void)
1705 {
1706 size_t i;
1707
1708 audio_process_options ("AUDIO", audio_options);
1709 for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1710 struct audio_driver *d = drvtab[i];
1711 if (d->options) {
1712 audio_process_options (d->name, d->options);
1713 }
1714 }
1715
1716 printf ("Audio options:\n");
1717 audio_print_options ("AUDIO", audio_options);
1718 printf ("\n");
1719
1720 printf ("Available drivers:\n");
1721
1722 for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1723 struct audio_driver *d = drvtab[i];
1724
1725 printf ("Name: %s\n", d->name);
1726 printf ("Description: %s\n", d->descr);
1727
1728 audio_pp_nb_voices ("playback", d->max_voices_out);
1729 audio_pp_nb_voices ("capture", d->max_voices_in);
1730
1731 if (d->options) {
1732 printf ("Options:\n");
1733 audio_print_options (d->name, d->options);
1734 }
1735 else {
1736 printf ("No options\n");
1737 }
1738 printf ("\n");
1739 }
1740
1741 printf (
1742 "Options are settable through environment variables.\n"
1743 "Example:\n"
1744 #ifdef _WIN32
1745 " set QEMU_AUDIO_DRV=wav\n"
1746 " set QEMU_WAV_PATH=c:\\tune.wav\n"
1747 #else
1748 " export QEMU_AUDIO_DRV=wav\n"
1749 " export QEMU_WAV_PATH=$HOME/tune.wav\n"
1750 "(for csh replace export with setenv in the above)\n"
1751 #endif
1752 " qemu ...\n\n"
1753 );
1754 }
1755
audio_driver_init(AudioState * s,struct audio_driver * drv,int out)1756 static int audio_driver_init (AudioState *s, struct audio_driver *drv, int out)
1757 {
1758 void* opaque;
1759
1760 if (drv->options) {
1761 audio_process_options (drv->name, drv->options);
1762 }
1763
1764 /* is the driver already initialized ? */
1765 if (out) {
1766 if (drv == s->drv_in) {
1767 s->drv_out = drv;
1768 s->drv_out_opaque = s->drv_in_opaque;
1769 return 0;
1770 }
1771 } else {
1772 if (drv == s->drv_out) {
1773 s->drv_in = drv;
1774 s->drv_in_opaque = s->drv_out_opaque;
1775 return 0;
1776 }
1777 }
1778
1779 BEGIN_NOSIGALRM
1780 opaque = drv->init();
1781 END_NOSIGALRM
1782
1783 if (opaque != NULL) {
1784 audio_init_nb_voices_out (drv);
1785 audio_init_nb_voices_in (drv);
1786 if (out) {
1787 s->drv_out = drv;
1788 s->drv_out_opaque = opaque;
1789 } else {
1790 s->drv_in = drv;
1791 s->drv_in_opaque = opaque;
1792 }
1793 return 0;
1794 }
1795 else {
1796 dolog ("Could not init `%s' audio driver\n", drv->name);
1797 return -1;
1798 }
1799 }
1800
audio_vm_change_state_handler(void * opaque,int running,int reason)1801 static void audio_vm_change_state_handler (void *opaque, int running,
1802 int reason)
1803 {
1804 AudioState *s = opaque;
1805 HWVoiceOut *hwo = NULL;
1806 HWVoiceIn *hwi = NULL;
1807 int op = running ? VOICE_ENABLE : VOICE_DISABLE;
1808
1809 s->vm_running = running;
1810 BEGIN_NOSIGALRM
1811 while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
1812 hwo->pcm_ops->ctl_out (hwo, op);
1813 }
1814
1815 while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
1816 hwi->pcm_ops->ctl_in (hwi, op);
1817 }
1818 END_NOSIGALRM
1819 }
1820
1821 // to make sure audio_atexit() is only called once
1822 static int initialized = 0;
1823
audio_atexit(void)1824 static void audio_atexit (void)
1825 {
1826 AudioState *s = &glob_audio_state;
1827 HWVoiceOut *hwo = NULL;
1828 HWVoiceIn *hwi = NULL;
1829
1830 if (!initialized) return;
1831 initialized = 0;
1832
1833 BEGIN_NOSIGALRM
1834 while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
1835 SWVoiceCap *sc;
1836
1837 hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
1838 hwo->pcm_ops->fini_out (hwo);
1839
1840 for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1841 CaptureVoiceOut *cap = sc->cap;
1842 struct capture_callback *cb;
1843
1844 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1845 cb->ops.destroy (cb->opaque);
1846 }
1847 }
1848 }
1849
1850 while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
1851 hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
1852 hwi->pcm_ops->fini_in (hwi);
1853 }
1854
1855 if (s->drv_in) {
1856 s->drv_in->fini (s->drv_in_opaque);
1857 }
1858 if (s->drv_out) {
1859 s->drv_out->fini (s->drv_out_opaque);
1860 }
1861 END_NOSIGALRM
1862 }
1863
audio_save(QEMUFile * f,void * opaque)1864 static void audio_save (QEMUFile *f, void *opaque)
1865 {
1866 (void) f;
1867 (void) opaque;
1868 }
1869
audio_load(QEMUFile * f,void * opaque,int version_id)1870 static int audio_load (QEMUFile *f, void *opaque, int version_id)
1871 {
1872 (void) f;
1873 (void) opaque;
1874
1875 if (version_id != 1) {
1876 return -EINVAL;
1877 }
1878
1879 return 0;
1880 }
1881
1882 static int
find_audio_driver(AudioState * s,int out)1883 find_audio_driver( AudioState* s, int out )
1884 {
1885 int i, done = 0, def;
1886 const char* envname;
1887 const char* drvname;
1888 struct audio_driver* drv = NULL;
1889 const char* drvtype = out ? "output" : "input";
1890
1891 envname = out ? "QEMU_AUDIO_OUT_DRV" : "QEMU_AUDIO_IN_DRV";
1892 drvname = audio_get_conf_str(envname, NULL, &def);
1893 if (drvname == NULL) {
1894 drvname = audio_get_conf_str("QEMU_AUDIO_DRV", NULL, &def);
1895 }
1896
1897 if (drvname != NULL) { /* look for a specific driver */
1898 for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
1899 if (!strcmp (drvname, drvtab[i]->name)) {
1900 drv = drvtab[i];
1901 break;
1902 }
1903 }
1904 }
1905
1906 if (drv != NULL) {
1907 done = !audio_driver_init (s, drv, out);
1908 if (!done) {
1909 dolog ("Could not initialize '%s' %s audio backend, trying default one.\n",
1910 drvname, drvtype);
1911 dolog ("Run with -qemu -audio-help to list available backends\n");
1912 drv = NULL;
1913 }
1914 }
1915
1916 if (!drv) {
1917 for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
1918 if (drvtab[i]->can_be_default) {
1919 drv = drvtab[i];
1920 done = !audio_driver_init (s, drv, out);
1921 if (done)
1922 break;
1923 }
1924 }
1925 }
1926
1927 if (!done) {
1928 drv = &no_audio_driver;
1929 done = !audio_driver_init (s, drv, out);
1930 if (!done) {
1931 /* this should never happen */
1932 dolog ("Could not initialize audio subsystem\n");
1933 return -1;
1934 }
1935 dolog ("warning: Could not find suitable audio %s backend\n", drvtype);
1936 }
1937
1938 if (VERBOSE_CHECK(init))
1939 dprint("using '%s' audio %s backend", drv->name, drvtype );
1940 return 0;
1941 }
1942
1943
audio_init(void)1944 static void audio_init (void)
1945 {
1946 AudioState *s = &glob_audio_state;
1947
1948 if (s->drv_out && s->drv_in) {
1949 return;
1950 }
1951
1952 LIST_INIT (&s->hw_head_out);
1953 LIST_INIT (&s->hw_head_in);
1954 LIST_INIT (&s->cap_head);
1955 atexit (audio_atexit);
1956
1957 s->ts = qemu_new_timer (vm_clock, audio_timer, s);
1958 if (!s->ts) {
1959 dolog ("Could not create audio timer\n");
1960 return;
1961 }
1962
1963 audio_process_options ("AUDIO", audio_options);
1964
1965 s->nb_hw_voices_out = conf.fixed_out.nb_voices;
1966 s->nb_hw_voices_in = conf.fixed_in.nb_voices;
1967
1968 if (s->nb_hw_voices_out <= 0) {
1969 dolog ("Bogus number of playback voices %d, setting to 1\n",
1970 s->nb_hw_voices_out);
1971 s->nb_hw_voices_out = 1;
1972 }
1973
1974 if (s->nb_hw_voices_in <= 0) {
1975 dolog ("Bogus number of capture voices %d, setting to 0\n",
1976 s->nb_hw_voices_in);
1977 s->nb_hw_voices_in = 0;
1978 }
1979
1980 if ( find_audio_driver (s, 0) != 0 ||
1981 find_audio_driver (s, 1) != 0 ) {
1982 qemu_del_timer (s->ts);
1983 return;
1984 }
1985
1986 VMChangeStateEntry *e;
1987
1988 if (conf.period.hertz <= 0) {
1989 if (conf.period.hertz < 0) {
1990 dolog ("warning: Timer period is negative - %d "
1991 "treating as zero\n",
1992 conf.period.hertz);
1993 }
1994 conf.period.ticks = 1;
1995 } else {
1996 conf.period.ticks = ticks_per_sec / conf.period.hertz;
1997 }
1998
1999 e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
2000 if (!e) {
2001 dolog ("warning: Could not register change state handler\n"
2002 "(Audio can continue looping even after stopping the VM)\n");
2003 }
2004
2005 initialized = 1;
2006
2007 LIST_INIT (&s->card_head);
2008 register_savevm ("audio", 0, 1, audio_save, audio_load, s);
2009 qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
2010 }
2011
AUD_register_card(const char * name,QEMUSoundCard * card)2012 void AUD_register_card (const char *name, QEMUSoundCard *card)
2013 {
2014 audio_init ();
2015 card->name = qemu_strdup (name);
2016 memset (&card->entries, 0, sizeof (card->entries));
2017 LIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
2018 }
2019
AUD_remove_card(QEMUSoundCard * card)2020 void AUD_remove_card (QEMUSoundCard *card)
2021 {
2022 LIST_REMOVE (card, entries);
2023 qemu_free (card->name);
2024 }
2025
2026 // this was added to work around a deadlock in SDL when quitting
AUD_cleanup()2027 void AUD_cleanup()
2028 {
2029 audio_atexit();
2030 }
2031
AUD_add_capture(struct audsettings * as,struct audio_capture_ops * ops,void * cb_opaque)2032 CaptureVoiceOut *AUD_add_capture (
2033 struct audsettings *as,
2034 struct audio_capture_ops *ops,
2035 void *cb_opaque
2036 )
2037 {
2038 AudioState *s = &glob_audio_state;
2039 CaptureVoiceOut *cap;
2040 struct capture_callback *cb;
2041
2042 if (audio_validate_settings (as)) {
2043 dolog ("Invalid settings were passed when trying to add capture\n");
2044 audio_print_settings (as);
2045 goto err0;
2046 }
2047
2048 cb = audio_calloc (AUDIO_FUNC, 1, sizeof (*cb));
2049 if (!cb) {
2050 dolog ("Could not allocate capture callback information, size %zu\n",
2051 sizeof (*cb));
2052 goto err0;
2053 }
2054 cb->ops = *ops;
2055 cb->opaque = cb_opaque;
2056
2057 cap = audio_pcm_capture_find_specific (as);
2058 if (cap) {
2059 LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
2060 return cap;
2061 }
2062 else {
2063 HWVoiceOut *hw;
2064 CaptureVoiceOut *cap;
2065
2066 cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap));
2067 if (!cap) {
2068 dolog ("Could not allocate capture voice, size %zu\n",
2069 sizeof (*cap));
2070 goto err1;
2071 }
2072
2073 hw = &cap->hw;
2074 LIST_INIT (&hw->sw_head);
2075 LIST_INIT (&cap->cb_head);
2076
2077 /* XXX find a more elegant way */
2078 hw->samples = 4096 * 4;
2079 hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
2080 sizeof (struct st_sample));
2081 if (!hw->mix_buf) {
2082 dolog ("Could not allocate capture mix buffer (%d samples)\n",
2083 hw->samples);
2084 goto err2;
2085 }
2086
2087 audio_pcm_init_info (&hw->info, as);
2088
2089 cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
2090 if (!cap->buf) {
2091 dolog ("Could not allocate capture buffer "
2092 "(%d samples, each %d bytes)\n",
2093 hw->samples, 1 << hw->info.shift);
2094 goto err3;
2095 }
2096
2097 hw->clip = mixeng_clip
2098 [hw->info.nchannels == 2]
2099 [hw->info.sign]
2100 [hw->info.swap_endianness]
2101 [audio_bits_to_index (hw->info.bits)];
2102
2103 LIST_INSERT_HEAD (&s->cap_head, cap, entries);
2104 LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
2105
2106 hw = NULL;
2107 while ((hw = audio_pcm_hw_find_any_out (hw))) {
2108 audio_attach_capture (hw);
2109 }
2110 return cap;
2111
2112 err3:
2113 qemu_free (cap->hw.mix_buf);
2114 err2:
2115 qemu_free (cap);
2116 err1:
2117 qemu_free (cb);
2118 err0:
2119 return NULL;
2120 }
2121 }
2122
AUD_del_capture(CaptureVoiceOut * cap,void * cb_opaque)2123 void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
2124 {
2125 struct capture_callback *cb;
2126
2127 for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
2128 if (cb->opaque == cb_opaque) {
2129 cb->ops.destroy (cb_opaque);
2130 LIST_REMOVE (cb, entries);
2131 qemu_free (cb);
2132
2133 if (!cap->cb_head.lh_first) {
2134 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
2135
2136 while (sw) {
2137 SWVoiceCap *sc = (SWVoiceCap *) sw;
2138 #ifdef DEBUG_CAPTURE
2139 dolog ("freeing %s\n", sw->name);
2140 #endif
2141
2142 sw1 = sw->entries.le_next;
2143 if (sw->rate) {
2144 st_rate_stop (sw->rate);
2145 sw->rate = NULL;
2146 }
2147 LIST_REMOVE (sw, entries);
2148 LIST_REMOVE (sc, entries);
2149 qemu_free (sc);
2150 sw = sw1;
2151 }
2152 LIST_REMOVE (cap, entries);
2153 qemu_free (cap);
2154 }
2155 return;
2156 }
2157 }
2158 }
2159
AUD_set_volume_out(SWVoiceOut * sw,int mute,uint8_t lvol,uint8_t rvol)2160 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
2161 {
2162 if (sw) {
2163 sw->vol.mute = mute;
2164 sw->vol.l = nominal_volume.l * lvol / 255;
2165 sw->vol.r = nominal_volume.r * rvol / 255;
2166 }
2167 }
2168
AUD_set_volume_in(SWVoiceIn * sw,int mute,uint8_t lvol,uint8_t rvol)2169 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
2170 {
2171 if (sw) {
2172 sw->vol.mute = mute;
2173 sw->vol.l = nominal_volume.l * lvol / 255;
2174 sw->vol.r = nominal_volume.r * rvol / 255;
2175 }
2176 }
2177