-
Notifications
You must be signed in to change notification settings - Fork 0
/
Copy pathaudio.c
2061 lines (1756 loc) · 50.7 KB
/
audio.c
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
/*
* QEMU Audio subsystem
*
* Copyright (c) 2003-2005 Vassili Karpov (malc)
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "qemu/osdep.h"
#include "hw/hw.h"
#include "audio.h"
#include "monitor/monitor.h"
#include "qemu/timer.h"
#include "sysemu/sysemu.h"
#include "qemu/cutils.h"
#define AUDIO_CAP "audio"
#include "audio_int.h"
/* #define DEBUG_LIVE */
/* #define DEBUG_OUT */
/* #define DEBUG_CAPTURE */
/* #define DEBUG_POLL */
#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
/* Order of CONFIG_AUDIO_DRIVERS is import.
The 1st one is the one used by default, that is the reason
that we generate the list.
*/
static struct audio_driver *drvtab[] = {
#ifdef CONFIG_SPICE
&spice_audio_driver,
#endif
CONFIG_AUDIO_DRIVERS
&no_audio_driver,
&wav_audio_driver
};
struct fixed_settings {
int enabled;
int nb_voices;
int greedy;
struct audsettings settings;
};
static struct {
struct fixed_settings fixed_out;
struct fixed_settings fixed_in;
union {
int hertz;
int64_t ticks;
} period;
int try_poll_in;
int try_poll_out;
} conf = {
.fixed_out = { /* DAC fixed settings */
.enabled = 1,
.nb_voices = 1,
.greedy = 1,
.settings = {
.freq = 44100,
.nchannels = 2,
.fmt = AUD_FMT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
.fixed_in = { /* ADC fixed settings */
.enabled = 1,
.nb_voices = 1,
.greedy = 1,
.settings = {
.freq = 44100,
.nchannels = 2,
.fmt = AUD_FMT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
.period = { .hertz = 100 },
.try_poll_in = 1,
.try_poll_out = 1,
};
static AudioState glob_audio_state;
const struct mixeng_volume nominal_volume = {
.mute = 0,
#ifdef FLOAT_MIXENG
.r = 1.0,
.l = 1.0,
#else
.r = 1ULL << 32,
.l = 1ULL << 32,
#endif
};
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
static void audio_print_options (const char *prefix,
struct audio_option *opt);
int audio_bug (const char *funcname, int cond)
{
if (cond) {
static int shown;
AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
if (!shown) {
struct audio_driver *d;
shown = 1;
AUD_log (NULL, "Save all your work and restart without audio\n");
AUD_log (NULL, "Please send bug report to [email protected]\n");
AUD_log (NULL, "I am sorry\n");
d = glob_audio_state.drv;
if (d) {
audio_print_options (d->name, d->options);
}
}
AUD_log (NULL, "Context:\n");
#if defined AUDIO_BREAKPOINT_ON_BUG
# if defined HOST_I386
# if defined __GNUC__
__asm__ ("int3");
# elif defined _MSC_VER
_asm _emit 0xcc;
# else
abort ();
# endif
# else
abort ();
# endif
#endif
}
return cond;
}
#endif
static inline int audio_bits_to_index (int bits)
{
switch (bits) {
case 8:
return 0;
case 16:
return 1;
case 32:
return 2;
default:
audio_bug ("bits_to_index", 1);
AUD_log (NULL, "invalid bits %d\n", bits);
return 0;
}
}
void *audio_calloc (const char *funcname, int nmemb, size_t size)
{
int cond;
size_t len;
len = nmemb * size;
cond = !nmemb || !size;
cond |= nmemb < 0;
cond |= len < size;
if (audio_bug ("audio_calloc", cond)) {
AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
funcname);
AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
return NULL;
}
return g_malloc0 (len);
}
static char *audio_alloc_prefix (const char *s)
{
const char qemu_prefix[] = "QEMU_";
size_t len, i;
char *r, *u;
if (!s) {
return NULL;
}
len = strlen (s);
r = g_malloc (len + sizeof (qemu_prefix));
u = r + sizeof (qemu_prefix) - 1;
pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
pstrcat (r, len + sizeof (qemu_prefix), s);
for (i = 0; i < len; ++i) {
u[i] = qemu_toupper(u[i]);
}
return r;
}
static const char *audio_audfmt_to_string (audfmt_e fmt)
{
switch (fmt) {
case AUD_FMT_U8:
return "U8";
case AUD_FMT_U16:
return "U16";
case AUD_FMT_S8:
return "S8";
case AUD_FMT_S16:
return "S16";
case AUD_FMT_U32:
return "U32";
case AUD_FMT_S32:
return "S32";
}
dolog ("Bogus audfmt %d returning S16\n", fmt);
return "S16";
}
static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
int *defaultp)
{
if (!strcasecmp (s, "u8")) {
*defaultp = 0;
return AUD_FMT_U8;
}
else if (!strcasecmp (s, "u16")) {
*defaultp = 0;
return AUD_FMT_U16;
}
else if (!strcasecmp (s, "u32")) {
*defaultp = 0;
return AUD_FMT_U32;
}
else if (!strcasecmp (s, "s8")) {
*defaultp = 0;
return AUD_FMT_S8;
}
else if (!strcasecmp (s, "s16")) {
*defaultp = 0;
return AUD_FMT_S16;
}
else if (!strcasecmp (s, "s32")) {
*defaultp = 0;
return AUD_FMT_S32;
}
else {
dolog ("Bogus audio format `%s' using %s\n",
s, audio_audfmt_to_string (defval));
*defaultp = 1;
return defval;
}
}
static audfmt_e audio_get_conf_fmt (const char *envname,
audfmt_e defval,
int *defaultp)
{
const char *var = getenv (envname);
if (!var) {
*defaultp = 1;
return defval;
}
return audio_string_to_audfmt (var, defval, defaultp);
}
static int audio_get_conf_int (const char *key, int defval, int *defaultp)
{
int val;
char *strval;
strval = getenv (key);
if (strval) {
*defaultp = 0;
val = atoi (strval);
return val;
}
else {
*defaultp = 1;
return defval;
}
}
static const char *audio_get_conf_str (const char *key,
const char *defval,
int *defaultp)
{
const char *val = getenv (key);
if (!val) {
*defaultp = 1;
return defval;
}
else {
*defaultp = 0;
return val;
}
}
void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
if (cap) {
fprintf(stderr, "%s: ", cap);
}
vfprintf(stderr, fmt, ap);
}
void AUD_log (const char *cap, const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (cap, fmt, ap);
va_end (ap);
}
static void audio_print_options (const char *prefix,
struct audio_option *opt)
{
char *uprefix;
if (!prefix) {
dolog ("No prefix specified\n");
return;
}
if (!opt) {
dolog ("No options\n");
return;
}
uprefix = audio_alloc_prefix (prefix);
for (; opt->name; opt++) {
const char *state = "default";
printf (" %s_%s: ", uprefix, opt->name);
if (opt->overriddenp && *opt->overriddenp) {
state = "current";
}
switch (opt->tag) {
case AUD_OPT_BOOL:
{
int *intp = opt->valp;
printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
}
break;
case AUD_OPT_INT:
{
int *intp = opt->valp;
printf ("integer, %s = %d\n", state, *intp);
}
break;
case AUD_OPT_FMT:
{
audfmt_e *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
audio_audfmt_to_string (*fmtp)
);
}
break;
case AUD_OPT_STR:
{
const char **strp = opt->valp;
printf ("string, %s = %s\n",
state,
*strp ? *strp : "(not set)");
}
break;
default:
printf ("???\n");
dolog ("Bad value tag for option %s_%s %d\n",
uprefix, opt->name, opt->tag);
break;
}
printf (" %s\n", opt->descr);
}
g_free (uprefix);
}
static void audio_process_options (const char *prefix,
struct audio_option *opt)
{
char *optname;
const char qemu_prefix[] = "QEMU_";
size_t preflen, optlen;
if (audio_bug (AUDIO_FUNC, !prefix)) {
dolog ("prefix = NULL\n");
return;
}
if (audio_bug (AUDIO_FUNC, !opt)) {
dolog ("opt = NULL\n");
return;
}
preflen = strlen (prefix);
for (; opt->name; opt++) {
size_t len, i;
int def;
if (!opt->valp) {
dolog ("Option value pointer for `%s' is not set\n",
opt->name);
continue;
}
len = strlen (opt->name);
/* len of opt->name + len of prefix + size of qemu_prefix
* (includes trailing zero) + zero + underscore (on behalf of
* sizeof) */
optlen = len + preflen + sizeof (qemu_prefix) + 1;
optname = g_malloc (optlen);
pstrcpy (optname, optlen, qemu_prefix);
/* copy while upper-casing, including trailing zero */
for (i = 0; i <= preflen; ++i) {
optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
}
pstrcat (optname, optlen, "_");
pstrcat (optname, optlen, opt->name);
def = 1;
switch (opt->tag) {
case AUD_OPT_BOOL:
case AUD_OPT_INT:
{
int *intp = opt->valp;
*intp = audio_get_conf_int (optname, *intp, &def);
}
break;
case AUD_OPT_FMT:
{
audfmt_e *fmtp = opt->valp;
*fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
}
break;
case AUD_OPT_STR:
{
const char **strp = opt->valp;
*strp = audio_get_conf_str (optname, *strp, &def);
}
break;
default:
dolog ("Bad value tag for option `%s' - %d\n",
optname, opt->tag);
break;
}
if (!opt->overriddenp) {
opt->overriddenp = &opt->overridden;
}
*opt->overriddenp = !def;
g_free (optname);
}
}
static void audio_print_settings (struct audsettings *as)
{
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
switch (as->fmt) {
case AUD_FMT_S8:
AUD_log (NULL, "S8");
break;
case AUD_FMT_U8:
AUD_log (NULL, "U8");
break;
case AUD_FMT_S16:
AUD_log (NULL, "S16");
break;
case AUD_FMT_U16:
AUD_log (NULL, "U16");
break;
case AUD_FMT_S32:
AUD_log (NULL, "S32");
break;
case AUD_FMT_U32:
AUD_log (NULL, "U32");
break;
default:
AUD_log (NULL, "invalid(%d)", as->fmt);
break;
}
AUD_log (NULL, " endianness=");
switch (as->endianness) {
case 0:
AUD_log (NULL, "little");
break;
case 1:
AUD_log (NULL, "big");
break;
default:
AUD_log (NULL, "invalid");
break;
}
AUD_log (NULL, "\n");
}
static int audio_validate_settings (struct audsettings *as)
{
int invalid;
invalid = as->nchannels != 1 && as->nchannels != 2;
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
case AUD_FMT_S16:
case AUD_FMT_U16:
case AUD_FMT_S32:
case AUD_FMT_U32:
break;
default:
invalid = 1;
break;
}
invalid |= as->freq <= 0;
return invalid ? -1 : 0;
}
static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8, sign = 0;
switch (as->fmt) {
case AUD_FMT_S8:
sign = 1;
/* fall through */
case AUD_FMT_U8:
break;
case AUD_FMT_S16:
sign = 1;
/* fall through */
case AUD_FMT_U16:
bits = 16;
break;
case AUD_FMT_S32:
sign = 1;
/* fall through */
case AUD_FMT_U32:
bits = 32;
break;
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
&& info->sign == sign
&& info->bits == bits
&& info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
case AUD_FMT_S8:
sign = 1;
case AUD_FMT_U8:
break;
case AUD_FMT_S16:
sign = 1;
case AUD_FMT_U16:
bits = 16;
shift = 1;
break;
case AUD_FMT_S32:
sign = 1;
case AUD_FMT_U32:
bits = 32;
shift = 2;
break;
}
info->freq = as->freq;
info->bits = bits;
info->sign = sign;
info->nchannels = as->nchannels;
info->shift = (as->nchannels == 2) + shift;
info->align = (1 << info->shift) - 1;
info->bytes_per_second = info->freq << info->shift;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
if (!len) {
return;
}
if (info->sign) {
memset (buf, 0x00, len << info->shift);
}
else {
switch (info->bits) {
case 8:
memset (buf, 0x80, len << info->shift);
break;
case 16:
{
int i;
uint16_t *p = buf;
int shift = info->nchannels - 1;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
for (i = 0; i < len << shift; i++) {
p[i] = s;
}
}
break;
case 32:
{
int i;
uint32_t *p = buf;
int shift = info->nchannels - 1;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
for (i = 0; i < len << shift; i++) {
p[i] = s;
}
}
break;
default:
AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
info->bits);
break;
}
}
}
/*
* Capture
*/
static void noop_conv (struct st_sample *dst, const void *src, int samples)
{
(void) src;
(void) dst;
(void) samples;
}
static CaptureVoiceOut *audio_pcm_capture_find_specific (
struct audsettings *as
)
{
CaptureVoiceOut *cap;
AudioState *s = &glob_audio_state;
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
if (audio_pcm_info_eq (&cap->hw.info, as)) {
return cap;
}
}
return NULL;
}
static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
{
struct capture_callback *cb;
#ifdef DEBUG_CAPTURE
dolog ("notification %d sent\n", cmd);
#endif
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.notify (cb->opaque, cmd);
}
}
static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
{
if (cap->hw.enabled != enabled) {
audcnotification_e cmd;
cap->hw.enabled = enabled;
cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
audio_notify_capture (cap, cmd);
}
}
static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
{
HWVoiceOut *hw = &cap->hw;
SWVoiceOut *sw;
int enabled = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
enabled = 1;
break;
}
}
audio_capture_maybe_changed (cap, enabled);
}
static void audio_detach_capture (HWVoiceOut *hw)
{
SWVoiceCap *sc = hw->cap_head.lh_first;
while (sc) {
SWVoiceCap *sc1 = sc->entries.le_next;
SWVoiceOut *sw = &sc->sw;
CaptureVoiceOut *cap = sc->cap;
int was_active = sw->active;
if (sw->rate) {
st_rate_stop (sw->rate);
sw->rate = NULL;
}
QLIST_REMOVE (sw, entries);
QLIST_REMOVE (sc, entries);
g_free (sc);
if (was_active) {
/* We have removed soft voice from the capture:
this might have changed the overall status of the capture
since this might have been the only active voice */
audio_recalc_and_notify_capture (cap);
}
sc = sc1;
}
}
static int audio_attach_capture (HWVoiceOut *hw)
{
AudioState *s = &glob_audio_state;
CaptureVoiceOut *cap;
audio_detach_capture (hw);
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
SWVoiceCap *sc;
SWVoiceOut *sw;
HWVoiceOut *hw_cap = &cap->hw;
sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
if (!sc) {
dolog ("Could not allocate soft capture voice (%zu bytes)\n",
sizeof (*sc));
return -1;
}
sc->cap = cap;
sw = &sc->sw;
sw->hw = hw_cap;
sw->info = hw->info;
sw->empty = 1;
sw->active = hw->enabled;
sw->conv = noop_conv;
sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
sw->vol = nominal_volume;
sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
if (!sw->rate) {
dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
g_free (sw);
return -1;
}
QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
#ifdef DEBUG_CAPTURE
sw->name = g_strdup_printf ("for %p %d,%d,%d",
hw, sw->info.freq, sw->info.bits,
sw->info.nchannels);
dolog ("Added %s active = %d\n", sw->name, sw->active);
#endif
if (sw->active) {
audio_capture_maybe_changed (cap, 1);
}
}
return 0;
}
/*
* Hard voice (capture)
*/
static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
SWVoiceIn *sw;
int m = hw->total_samples_captured;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
m = audio_MIN (m, sw->total_hw_samples_acquired);
}
}
return m;
}
int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
{
int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
return live;
}
int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
int live, int pending)
{
int left = hw->samples - pending;
int len = audio_MIN (left, live);
int clipped = 0;
while (len) {
struct st_sample *src = hw->mix_buf + hw->rpos;
uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
int samples_till_end_of_buf = hw->samples - hw->rpos;
int samples_to_clip = audio_MIN (len, samples_till_end_of_buf);
hw->clip (dst, src, samples_to_clip);
hw->rpos = (hw->rpos + samples_to_clip) % hw->samples;
len -= samples_to_clip;
clipped += samples_to_clip;
}
return clipped;
}
/*
* Soft voice (capture)
*/
static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
{
HWVoiceIn *hw = sw->hw;
int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
int rpos;
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
rpos = hw->wpos - live;
if (rpos >= 0) {
return rpos;
}
else {
return hw->samples + rpos;
}
}
int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
{
HWVoiceIn *hw = sw->hw;
int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
samples = size >> sw->info.shift;
if (!live) {
return 0;
}
swlim = (live * sw->ratio) >> 32;
swlim = audio_MIN (swlim, samples);
while (swlim) {
src = hw->conv_buf + rpos;
isamp = hw->wpos - rpos;
/* XXX: <= ? */
if (isamp <= 0) {
isamp = hw->samples - rpos;
}
if (!isamp) {
break;
}
osamp = swlim;
if (audio_bug (AUDIO_FUNC, osamp < 0)) {
dolog ("osamp=%d\n", osamp);
return 0;
}
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
rpos = (rpos + isamp) % hw->samples;
dst += osamp;
ret += osamp;
total += isamp;
}
if (!(hw->ctl_caps & VOICE_VOLUME_CAP)) {
mixeng_volume (sw->buf, ret, &sw->vol);
}
sw->clip (buf, sw->buf, ret);
sw->total_hw_samples_acquired += total;
return ret << sw->info.shift;
}
/*
* Hard voice (playback)
*/
static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
{
SWVoiceOut *sw;
int m = INT_MAX;
int nb_live = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active || !sw->empty) {
m = audio_MIN (m, sw->total_hw_samples_mixed);
nb_live += 1;
}
}
*nb_livep = nb_live;
return m;
}
static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
{
int smin;
int nb_live1;
smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
if (nb_live) {
*nb_live = nb_live1;
}
if (nb_live1) {
int live = smin;
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
return live;
}
return 0;
}