[go: up one dir, main page]

File: aaudio.c

package info (click to toggle)
cmus 2.12.0-2
  • links: PTS, VCS
  • area: main
  • in suites: forky, sid
  • size: 2,404 kB
  • sloc: ansic: 40,498; sh: 1,642; makefile: 255; python: 157
file content (899 lines) | stat: -rw-r--r-- 33,040 bytes parent folder | download | duplicates (2)
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
/*
 * Copyright (C) 2024 Patrick Gaskin <patrick@pgaskin.net>
 *
 * This program is free software: you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation, either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program.  If not, see <http://www.gnu.org/licenses/>.
 */

#include <errno.h>
#include <limits.h>
#include <stdbool.h>
#include <stddef.h>
#include <stdint.h>

// for development, can cross-compile with $ANDROID_NDK_HOME/toolchains/llvm/prebuilt/linux-x86_64/bin/clang -target aarch64-linux-android26 -shared -o aaudio.so -fPIC -D__ANDROID_UNAVAILABLE_SYMBOLS_ARE_WEAK__ -Werror=unguarded-availability -Wall -std=gnu11 op/aaudio.c -laaudio
// also see https://github.com/google/oboe/blob/main/docs/AndroidAudioHistory.md
// also see https://android.googlesource.com/platform/frameworks/av/+/master/media/libaaudio/examples/utils/AAudioSimplePlayer.h

#ifndef __ANDROID__
// make ide autocomplete work without using a full ndk toolchain
#define __INTRODUCED_IN(api_level)
#endif

// https://developer.android.com/ndk/guides/using-newer-apis
#define REQUIRES_API(x) __attribute__((__availability__(android,introduced=x)))
#define API_AT_LEAST(x) __builtin_available(android x, *)
#define AAUDIO_MINIMUM_API 26

#include <aaudio/AAudio.h>

#include "../op.h"
#include "../mixer.h"
#include "../sf.h"
#include "../utils.h"
#include "../xmalloc.h"

// mapping from AAUDIO_CHANNEL_* enum values to cmus channel_position_t values
//
// cat "$(find ${ANDROID_NDK_HOME:-$ANDROID_HOME/ndk} -wholename '*/AAudio.h' | sort -n | tail -n1)" |
// grep AAUDIO_CHANNEL | tr -d ' \n' | tr '|,' ' \n' | grep -F '<<' | cut -d '_' -f3- |
// cut -d '=' -f1 | xargs printf '#define A2C__%s\tCHANNEL_POSITION_INVALID\n' |
// sed -E $(printf " -e s/(A2C__%s\\\t)CHANNEL_POSITION_INVALID/\\\1CHANNEL_POSITION_%s/" \
//   FRONT_LEFT             FRONT_LEFT \
//   FRONT_RIGHT            FRONT_RIGHT \
//   FRONT_CENTER           FRONT_CENTER \
//   LOW_FREQUENCY          LFE \
//   BACK_LEFT              REAR_LEFT \
//   BACK_RIGHT             REAR_RIGHT \
//   FRONT_LEFT_OF_CENTER   FRONT_LEFT_OF_CENTER \
//   FRONT_RIGHT_OF_CENTER  FRONT_RIGHT_OF_CENTER \
//   BACK_CENTER            REAR_CENTER \
//   SIDE_LEFT              SIDE_LEFT \
//   SIDE_RIGHT             SIDE_RIGHT \
//   TOP_CENTER             TOP_CENTER \
//   TOP_FRONT_LEFT         TOP_FRONT_LEFT \
//   TOP_FRONT_CENTER       TOP_FRONT_CENTER \
//   TOP_FRONT_RIGHT        TOP_FRONT_RIGHT \
//   TOP_BACK_LEFT          TOP_REAR_LEFT \
//   TOP_BACK_CENTER        TOP_REAR_CENTER \
//   TOP_BACK_RIGHT         TOP_REAR_RIGHT \
// ) |
// column -s $'\t' -t | tee /dev/stderr | cut -d ' ' -f2 | cut -d '_' -f3- |
// xargs printf ' X(%s)' | xargs -0 printf '#define A2C_CHANNELS%s\n'
#define A2C__FRONT_LEFT             CHANNEL_POSITION_FRONT_LEFT
#define A2C__FRONT_RIGHT            CHANNEL_POSITION_FRONT_RIGHT
#define A2C__FRONT_CENTER           CHANNEL_POSITION_FRONT_CENTER
#define A2C__LOW_FREQUENCY          CHANNEL_POSITION_LFE
#define A2C__BACK_LEFT              CHANNEL_POSITION_REAR_LEFT
#define A2C__BACK_RIGHT             CHANNEL_POSITION_REAR_RIGHT
#define A2C__FRONT_LEFT_OF_CENTER   CHANNEL_POSITION_FRONT_LEFT_OF_CENTER
#define A2C__FRONT_RIGHT_OF_CENTER  CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER
#define A2C__BACK_CENTER            CHANNEL_POSITION_REAR_CENTER
#define A2C__SIDE_LEFT              CHANNEL_POSITION_SIDE_LEFT
#define A2C__SIDE_RIGHT             CHANNEL_POSITION_SIDE_RIGHT
#define A2C__TOP_CENTER             CHANNEL_POSITION_TOP_CENTER
#define A2C__TOP_FRONT_LEFT         CHANNEL_POSITION_TOP_FRONT_LEFT
#define A2C__TOP_FRONT_CENTER       CHANNEL_POSITION_TOP_FRONT_CENTER
#define A2C__TOP_FRONT_RIGHT        CHANNEL_POSITION_TOP_FRONT_RIGHT
#define A2C__TOP_BACK_LEFT          CHANNEL_POSITION_TOP_REAR_LEFT
#define A2C__TOP_BACK_CENTER        CHANNEL_POSITION_TOP_REAR_CENTER
#define A2C__TOP_BACK_RIGHT         CHANNEL_POSITION_TOP_REAR_RIGHT
#define A2C__TOP_SIDE_LEFT          CHANNEL_POSITION_INVALID
#define A2C__TOP_SIDE_RIGHT         CHANNEL_POSITION_INVALID
#define A2C__BOTTOM_FRONT_LEFT      CHANNEL_POSITION_INVALID
#define A2C__BOTTOM_FRONT_CENTER    CHANNEL_POSITION_INVALID
#define A2C__BOTTOM_FRONT_RIGHT     CHANNEL_POSITION_INVALID
#define A2C__LOW_FREQUENCY_2        CHANNEL_POSITION_INVALID
#define A2C__FRONT_WIDE_LEFT        CHANNEL_POSITION_INVALID
#define A2C__FRONT_WIDE_RIGHT       CHANNEL_POSITION_INVALID
#define A2C_CHANNELS X(FRONT_LEFT) X(FRONT_RIGHT) X(FRONT_CENTER) X(LOW_FREQUENCY) X(BACK_LEFT) X(BACK_RIGHT) X(FRONT_LEFT_OF_CENTER) X(FRONT_RIGHT_OF_CENTER) X(BACK_CENTER) X(SIDE_LEFT) X(SIDE_RIGHT) X(TOP_CENTER) X(TOP_FRONT_LEFT) X(TOP_FRONT_CENTER) X(TOP_FRONT_RIGHT) X(TOP_BACK_LEFT) X(TOP_BACK_CENTER) X(TOP_BACK_RIGHT) X(TOP_SIDE_LEFT) X(TOP_SIDE_RIGHT) X(BOTTOM_FRONT_LEFT) X(BOTTOM_FRONT_CENTER) X(BOTTOM_FRONT_RIGHT) X(LOW_FREQUENCY_2) X(FRONT_WIDE_LEFT) X(FRONT_WIDE_RIGHT)

// mapping from AAUDIO_CHANNEL_* masks to cmus channel_position_t lists
//
// cat "$(find ${ANDROID_NDK_HOME:-$ANDROID_HOME/ndk} -wholename '*/aaudio/AAudio.h' | sort -n | tail -n1)" |
// grep AAUDIO_CHANNEL | tr -d ' \n' | tr '|,' ',\n' | grep -Fve '<<' -e '-1' | cut -d '_' -f3- |
// xargs printf '#define A2C__%s\n' | tr '=' '\t' | sed -E 's/AAUDIO_CHANNEL_([A-Z0-9_]+)/A2C__\1/g' |
// column -s $'\t' -t | tee /dev/stderr | cut -d ' ' -f2 | cut -d '_' -f3- |
// xargs printf ' X(%s)' | xargs -0 printf '#define A2C_LAYOUTS%s\n'
#define A2C__MONO           A2C__FRONT_LEFT
#define A2C__STEREO         A2C__FRONT_LEFT,A2C__FRONT_RIGHT
#define A2C__2POINT1        A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__LOW_FREQUENCY
#define A2C__TRI            A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__FRONT_CENTER
#define A2C__TRI_BACK       A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__BACK_CENTER
#define A2C__3POINT1        A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__FRONT_CENTER,A2C__LOW_FREQUENCY
#define A2C__2POINT0POINT2  A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__TOP_SIDE_LEFT,A2C__TOP_SIDE_RIGHT
#define A2C__2POINT1POINT2  A2C__2POINT0POINT2,A2C__LOW_FREQUENCY
#define A2C__3POINT0POINT2  A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__FRONT_CENTER,A2C__TOP_SIDE_LEFT,A2C__TOP_SIDE_RIGHT
#define A2C__3POINT1POINT2  A2C__3POINT0POINT2,A2C__LOW_FREQUENCY
#define A2C__QUAD           A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__BACK_LEFT,A2C__BACK_RIGHT
#define A2C__QUAD_SIDE      A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__SIDE_LEFT,A2C__SIDE_RIGHT
#define A2C__SURROUND       A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__FRONT_CENTER,A2C__BACK_CENTER
#define A2C__PENTA          A2C__QUAD,A2C__FRONT_CENTER
#define A2C__5POINT1        A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__FRONT_CENTER,A2C__LOW_FREQUENCY,A2C__BACK_LEFT,A2C__BACK_RIGHT
#define A2C__5POINT1_SIDE   A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__FRONT_CENTER,A2C__LOW_FREQUENCY,A2C__SIDE_LEFT,A2C__SIDE_RIGHT
#define A2C__6POINT1        A2C__FRONT_LEFT,A2C__FRONT_RIGHT,A2C__FRONT_CENTER,A2C__LOW_FREQUENCY,A2C__BACK_LEFT,A2C__BACK_RIGHT,A2C__BACK_CENTER
#define A2C__7POINT1        A2C__5POINT1,A2C__SIDE_LEFT,A2C__SIDE_RIGHT
#define A2C__5POINT1POINT2  A2C__5POINT1,A2C__TOP_SIDE_LEFT,A2C__TOP_SIDE_RIGHT
#define A2C__5POINT1POINT4  A2C__5POINT1,A2C__TOP_FRONT_LEFT,A2C__TOP_FRONT_RIGHT,A2C__TOP_BACK_LEFT,A2C__TOP_BACK_RIGHT
#define A2C__7POINT1POINT2  A2C__7POINT1,A2C__TOP_SIDE_LEFT,A2C__TOP_SIDE_RIGHT
#define A2C__7POINT1POINT4  A2C__7POINT1,A2C__TOP_FRONT_LEFT,A2C__TOP_FRONT_RIGHT,A2C__TOP_BACK_LEFT,A2C__TOP_BACK_RIGHT
#define A2C__9POINT1POINT4  A2C__7POINT1POINT4,A2C__FRONT_WIDE_LEFT,A2C__FRONT_WIDE_RIGHT
#define A2C__9POINT1POINT6  A2C__9POINT1POINT4,A2C__TOP_SIDE_LEFT,A2C__TOP_SIDE_RIGHT
#define A2C__FRONT_BACK     A2C__FRONT_CENTER,A2C__BACK_CENTER
#define A2C_LAYOUTS X(MONO) X(STEREO) X(2POINT1) X(TRI) X(TRI_BACK) X(3POINT1) X(2POINT0POINT2) X(2POINT1POINT2) X(3POINT0POINT2) X(3POINT1POINT2) X(QUAD) X(QUAD_SIDE) X(SURROUND) X(PENTA) X(5POINT1) X(5POINT1_SIDE) X(6POINT1) X(7POINT1) X(5POINT1POINT2) X(5POINT1POINT4) X(7POINT1POINT2) X(7POINT1POINT4) X(9POINT1POINT4) X(9POINT1POINT6) X(FRONT_BACK)

// convert a cmus channel map to an equivalent aaudio channel mask (the returned
// value will either be invalid or have the same number of bits set as the
// number of channels)
static aaudio_channel_mask_t cmus_channel_map_to_aaudio_mask(int channels, const channel_position_t *channel_map)
{
	aaudio_channel_mask_t mask = 0;

	// we can only convert a valid channel map
	if (channels >= CHANNELS_MAX || !channel_map || !channel_map_valid(channel_map)) {
		return AAUDIO_CHANNEL_INVALID;
	}

	// special case for mono since cmus defines a separate channel position
	// for it
	if (channels == 1 && channel_map[0] == CHANNEL_POSITION_MONO) {
		return AAUDIO_CHANNEL_FRONT_LEFT;
	}

	// fill the mask, returning invalid if it has duplicates or no mapping
	for (int i = 0; i < channels; i++) {
		#define X(aaudio) \
		if (A2C__##aaudio != CHANNEL_POSITION_INVALID && channel_map[i] == A2C__##aaudio) { \
			if (mask & AAUDIO_CHANNEL_##aaudio) \
				return AAUDIO_CHANNEL_INVALID; \
			mask |= AAUDIO_CHANNEL_##aaudio; \
		}
		A2C_CHANNELS
		#undef X
	}

	return mask;
}

// get the expected cmus channel order for the specified aaudio channel mask
static bool channel_map_init_aaudio(aaudio_channel_mask_t mask, channel_position_t *map)
{
	switch (mask) {
	#define X(aaudio) \
	case AAUDIO_CHANNEL_##aaudio: channel_map_copy(map, (channel_position_t[CHANNELS_MAX]){ A2C__##aaudio }); return true;
	A2C_LAYOUTS
	#undef X
	}
	return false;
}

// get the name of a known aaudio channel mask
static const char *aaudio_channel_to_string(aaudio_channel_mask_t mask)
{
	switch (mask) {
		#define X(aaudio) \
		case AAUDIO_CHANNEL_##aaudio: return #aaudio;
		A2C_CHANNELS
		#undef X
	}
	switch (mask) {
		#define X(aaudio) \
		case AAUDIO_CHANNEL_##aaudio: return #aaudio;
		A2C_LAYOUTS
		#undef X
	}
	return NULL;
}

// allocate a map of output frame byte indexes to input frame byte indexes (or
// -1 to zero) to remap channels (map must be sf_get_frame_size elements)
static ssize_t *make_channel_remap(const channel_position_t *channel_map_out, const channel_position_t *channel_map_in, sample_format_t sf)
{
	int byte, channel_out, channel_in;
	ssize_t *map;

	map = malloc(sizeof(ssize_t) * (size_t) sf_get_frame_size(sf));
	if (!map) {
		return NULL;
	}

	if (!channel_map_out || !channel_map_valid(channel_map_out) || !channel_map_in || !channel_map_valid(channel_map_in)) {
		for (byte = 0; byte < sf_get_frame_size(sf); byte++) {
			map[byte] = byte;
		}
	} else {
		for (byte = 0; byte < sf_get_frame_size(sf); byte++) {
			map[byte] = -1;
		}
		for (channel_out = 0; channel_out < sf_get_channels(sf); channel_out++) {
			if (channel_map_out[channel_out] != CHANNEL_POSITION_INVALID) {
				for (channel_in = 0; channel_in < sf_get_channels(sf); channel_in++) {
					if (channel_map_in[channel_in] == channel_map_out[channel_out]) {
						for (byte = 0; byte < sf_get_sample_size(sf); byte++) {
							map[sf_get_sample_size(sf) * channel_out + byte] = (ssize_t) sf_get_sample_size(sf) * channel_in + byte;
						}
						break;
					}
				}
			}
		}
	}

	d_print("remap bytes");
	for (byte = 0; byte < sf_get_frame_size(sf); byte++) {
		d_print(" %03zd", map[byte]);
	}
	d_print("\n");

	return map;
}

// if remap is non-null, applies it and returns dst, otherwise returns src
static const uint8_t *apply_channel_remap(uint8_t *dst, const uint8_t *src, size_t n, sample_format_t sf, const ssize_t *remap)
{
	size_t off_frame, off;
	BUG_ON(!src || dst == src);

	if (remap) {
		BUG_ON(!dst);

		for (off_frame = 0; off_frame < n; off_frame += (size_t) sf_get_frame_size(sf)) {
			for (off = 0; off < (size_t) sf_get_frame_size(sf); off++) {
				if (remap[off] != -1) {
					dst[off_frame+off] = src[off_frame+remap[off]];
				} else {
					dst[off_frame+off] = 0;
				}
			}
		}
		return dst;
	}
	return src;
}

// configure builder to use the specified sample format (and on a best-effort
// basis, channel_map, if provided)
//
// successful if the sample format is valid and was configured
//
// on success, out_remap will be set to NULL, or if a channel map was configured
// and requires remapping, an allocated map of target frame byte indexes from
// the source index
REQUIRES_API(AAUDIO_MINIMUM_API)
static aaudio_result_t configure_aaudio_sf(AAudioStreamBuilder *builder, sample_format_t sf, const channel_position_t *channel_map, ssize_t **out_remap)
{
	aaudio_format_t format;
	aaudio_channel_mask_t mask;
	channel_position_t mask_expected_channels[CHANNELS_MAX];

	// apply the sample format
	if (!sf_get_signed(sf)) {
		d_print("aaudio does not support unsigned samples\n");
		return AAUDIO_ERROR_INVALID_FORMAT;
	}
	if (sf_get_bigendian(sf)) {
		d_print("aaudio does not support big-endian samples\n");
		return AAUDIO_ERROR_INVALID_FORMAT;
	}
	switch (sf_get_bits(sf)) {
	case 16: format = AAUDIO_FORMAT_PCM_I16; break;
	case 24: format = AAUDIO_FORMAT_PCM_I24_PACKED; break;
	case 32: format = AAUDIO_FORMAT_PCM_I32; break;
	default:
		d_print("unsupported sample format bits\n");
		return AAUDIO_ERROR_INVALID_FORMAT;
	}
	AAudioStreamBuilder_setFormat(builder, format);

	// apply the sample rate
	AAudioStreamBuilder_setSampleRate(builder, sf_get_rate(sf));

	// set the channel count
	//
	// note: if no channel mask is set, aaudio will treat the first two
	// channels as left/right (duplicating mono to stereo if required), and
	// leave the rest up to the device, dropping them if the device doesn't
	// have that many channels
	AAudioStreamBuilder_setChannelCount(builder, sf_get_channels(sf));

	// if we have a channel map, apply it on a best-effort basis
	*out_remap = NULL;
	if (channel_map && channel_map_valid(channel_map)) {
		if (API_AT_LEAST(32)) {
			mask = cmus_channel_map_to_aaudio_mask(sf_get_channels(sf), channel_map);
			d_print("channel map aaudio mask %d (%s)\n", mask, aaudio_channel_to_string(mask) ? aaudio_channel_to_string(mask) : "(null)");
			if (mask == AAUDIO_CHANNEL_INVALID) {
				d_print("not applying channel map since it contains duplicates or not all channels have an aaudio equivalent\n");
			} else {
				if (!channel_map_init_aaudio(mask, mask_expected_channels)) {
					d_print("not applying channel map since there isn't a valid cmus channel mapping for the aaudio mask\n");
				} else {
					if (!channel_map_equal(channel_map, mask_expected_channels, sf_get_channels(sf))) {
						d_print("will remap channels since the input channel_map order doesn't match the order expected by aaudio\n");
						*out_remap = make_channel_remap(mask_expected_channels, channel_map, sf);
						if (!*out_remap) return AAUDIO_ERROR_NO_MEMORY;
					}
					d_print("applying channel mask\n");
					AAudioStreamBuilder_setChannelMask(builder, mask);
				}
			}
		}
	}

	return AAUDIO_OK;
}

// maps an res to a suitable error code
static int OP_ERROR_AAUDIO(aaudio_result_t res) {
	// see https://android.googlesource.com/platform/bionic/+/refs/heads/main/libc/private/bionic_errdefs.h
	switch (res) {
	case AAUDIO_OK:                                           return 0;
	case AAUDIO_ERROR_INTERNAL:                               return OP_ERROR_INTERNAL;
	case AAUDIO_ERROR_NO_SERVICE:                             return OP_ERROR_NOT_SUPPORTED;
	case AAUDIO_ERROR_INVALID_FORMAT:                         return OP_ERROR_SAMPLE_FORMAT;
	case AAUDIO_ERROR_INVALID_RATE:                           return OP_ERROR_SAMPLE_FORMAT;
	case AAUDIO_ERROR_UNAVAILABLE:      errno = ECONNREFUSED; return OP_ERROR_ERRNO; // Connection refused
	case AAUDIO_ERROR_DISCONNECTED:     errno = ECONNRESET;   return OP_ERROR_ERRNO; // Connection reset by peer
	case AAUDIO_ERROR_TIMEOUT:          errno = ETIMEDOUT;    return OP_ERROR_ERRNO; // Connection timed out
	case AAUDIO_ERROR_WOULD_BLOCK:      errno = ENOBUFS;      return OP_ERROR_ERRNO; // No buffer space available
	case AAUDIO_ERROR_UNIMPLEMENTED:    errno = ENOSYS;       return OP_ERROR_ERRNO; // Function not implemented
	case AAUDIO_ERROR_NO_FREE_HANDLES:  errno = EMFILE;       return OP_ERROR_ERRNO; // Too many open files
	case AAUDIO_ERROR_NO_MEMORY:        errno = ENOMEM;       return OP_ERROR_ERRNO; // Out of memory
	case AAUDIO_ERROR_NULL:             errno = EFAULT;       return OP_ERROR_ERRNO; // Bad address
	case AAUDIO_ERROR_OUT_OF_RANGE:     errno = EINVAL;       return OP_ERROR_ERRNO; // Invalid argument
	case AAUDIO_ERROR_INVALID_HANDLE:   errno = EBADF;        return OP_ERROR_ERRNO; // Bad file descriptor
	case AAUDIO_ERROR_INVALID_STATE:    errno = EBADFD;       return OP_ERROR_ERRNO; // File descriptor in bad state
	case AAUDIO_ERROR_ILLEGAL_ARGUMENT: errno = EINVAL;       return OP_ERROR_ERRNO; // Invalid argument
	default:                                                  return OP_ERROR_INTERNAL;
	}
}
// note: all options require restarting the output stream to apply
static aaudio_performance_mode_t op_aaudio_opt_performance_mode = AAUDIO_PERFORMANCE_MODE_POWER_SAVING;
static aaudio_allowed_capture_policy_t op_aaudio_opt_allowed_capture = AAUDIO_ALLOW_CAPTURE_BY_ALL;
static aaudio_sharing_mode_t op_aaudio_opt_sharing_mode = AAUDIO_SHARING_MODE_SHARED;
static bool op_aaudio_opt_disable_spatialization = false;

// if we ever decide to support AAUDIO_PERFORMANCE_MODE_LOW_LATENCY streams,
// note that disconnection is broken for shared low-latency streams on RQ1A
// (this doesn't affect us right now since we don't use low-latency shared mmap
// streams)
//
// https://issuetracker.google.com/issues/173928197

static int op_aaudio_set_performance_mode(const char *val)
{
	if (!strcmp(val, "none")) {
		op_aaudio_opt_performance_mode = AAUDIO_PERFORMANCE_MODE_NONE;
		return OP_ERROR_SUCCESS;
	}
	if (!strcmp(val, "power_saving")) {
		op_aaudio_opt_performance_mode = AAUDIO_PERFORMANCE_MODE_POWER_SAVING;
		return OP_ERROR_SUCCESS;
	}
	errno = EINVAL;
	return -OP_ERROR_ERRNO;
}

static int op_aaudio_get_performance_mode(char **val)
{
	switch (op_aaudio_opt_performance_mode) {
	default:
		__attribute__((fallthrough));
	case AAUDIO_PERFORMANCE_MODE_NONE:
		*val = xstrdup("none");
		break;
	case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
		*val = xstrdup("power_saving");
		break;
	}
	return OP_ERROR_SUCCESS;
}

static int op_aaudio_set_allowed_capture(const char *val)
{
	if (!strcmp(val, "all")) {
		op_aaudio_opt_allowed_capture = AAUDIO_ALLOW_CAPTURE_BY_ALL;
		return OP_ERROR_SUCCESS;
	}
	if (!strcmp(val, "none")) {
		op_aaudio_opt_allowed_capture = AAUDIO_ALLOW_CAPTURE_BY_NONE;
		return OP_ERROR_SUCCESS;
	}
	if (!strcmp(val, "system")) {
		op_aaudio_opt_allowed_capture = AAUDIO_ALLOW_CAPTURE_BY_SYSTEM;
		return OP_ERROR_SUCCESS;
	}
	errno = EINVAL;
	return -OP_ERROR_ERRNO;
}

static int op_aaudio_get_allowed_capture(char **val)
{
	switch (op_aaudio_opt_allowed_capture) {
	default:
		__attribute__((fallthrough));
	case AAUDIO_ALLOW_CAPTURE_BY_ALL:
		*val = xstrdup("all");
		break;
	case AAUDIO_ALLOW_CAPTURE_BY_NONE:
		*val = xstrdup("none");
		break;
	case AAUDIO_ALLOW_CAPTURE_BY_SYSTEM:
		*val = xstrdup("system");
		break;
	}
	return OP_ERROR_SUCCESS;
}

static int op_aaudio_set_sharing_mode(const char *val)
{
	if (!strcmp(val, "shared")) {
		op_aaudio_opt_sharing_mode = AAUDIO_SHARING_MODE_SHARED;
		return OP_ERROR_SUCCESS;
	}
	if (!strcmp(val, "exclusive")) {
		op_aaudio_opt_sharing_mode = AAUDIO_SHARING_MODE_EXCLUSIVE;
		return OP_ERROR_SUCCESS;
	}
	errno = EINVAL;
	return -OP_ERROR_ERRNO;
}

static int op_aaudio_get_sharing_mode(char **val)
{
	switch (op_aaudio_opt_performance_mode) {
	default:
		__attribute__((fallthrough));
	case AAUDIO_SHARING_MODE_SHARED:
		*val = xstrdup("shared");
		break;
	case AAUDIO_SHARING_MODE_EXCLUSIVE:
		*val = xstrdup("exclusive");
		break;
	}
	return OP_ERROR_SUCCESS;
}

static int op_aaudio_set_disable_spatialization(const char *val)
{
	op_aaudio_opt_disable_spatialization = strcmp(val, "true") ? false : true;
	return OP_ERROR_SUCCESS;
}

static int op_aaudio_get_disable_spatialization(char **val)
{
	*val = xstrdup(op_aaudio_opt_disable_spatialization ? "true" : "false");
	return OP_ERROR_SUCCESS;
}

static bool aaudio_supported() {
	if (API_AT_LEAST(27)) {} else {
		// don't use AAudio on API 26 due to bug causing crash on some
		// devices when closing stream
		//
		// https://github.com/google/oboe/issues/40
		return -OP_ERROR_NOT_SUPPORTED;
	}
	if (API_AT_LEAST(AAUDIO_MINIMUM_API)) {
		return !!&AAudio_createStreamBuilder;
	}
	return false;
}

static struct {
	AAudioStream *stream;
	int32_t device;
	aaudio_result_t error;
	sample_format_t sf;
	ssize_t *remap;
	char *remap_buf;
} op;

int mixer_notify_output_in, mixer_notify_output_out;

static int op_aaudio_init(void)
{
	if (!aaudio_supported()) {
		// skip the output plugin (see op_select_any)
		return -OP_ERROR_NOT_SUPPORTED;
	}

	init_pipes(&mixer_notify_output_out, &mixer_notify_output_in);

	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_exit(void)
{
	close(mixer_notify_output_out);
	close(mixer_notify_output_in);

	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static void handle_error(AAudioStream *stream, void *userData, aaudio_result_t error) {
	if (error == AAUDIO_ERROR_DISCONNECTED) {
		notify_via_pipe(mixer_notify_output_in);
	}
	d_print("stream errored (%d - %s)\n", error, AAudio_convertResultToText(error));
	op.error = error;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_open(sample_format_t sf, const channel_position_t *channel_map)
{
	aaudio_result_t rc;
	AAudioStreamBuilder *bld;

	// create the stream builder
	rc = AAudio_createStreamBuilder(&bld);
	if (rc) {
		d_print("create stream builder failed (%d - %s)\n", rc, AAudio_convertResultToText(rc));
		return -OP_ERROR_AAUDIO(rc);
	}

	// set the error callback
	AAudioStreamBuilder_setErrorCallback(bld, handle_error, NULL);

	// apply the options
	AAudioStreamBuilder_setSharingMode(bld, op_aaudio_opt_sharing_mode);
	AAudioStreamBuilder_setPerformanceMode(bld, op_aaudio_opt_performance_mode);
	if (API_AT_LEAST(28)) AAudioStreamBuilder_setContentType(bld, AAUDIO_CONTENT_TYPE_MUSIC);
	if (API_AT_LEAST(28)) AAudioStreamBuilder_setUsage(bld, AAUDIO_USAGE_MEDIA);
	if (API_AT_LEAST(29)) AAudioStreamBuilder_setAllowedCapturePolicy(bld, op_aaudio_opt_allowed_capture);
	if (API_AT_LEAST(31)) AAudioStreamBuilder_setAttributionTag(bld, "cmus");
	if (API_AT_LEAST(32)) AAudioStreamBuilder_setSpatializationBehavior(bld, op_aaudio_opt_disable_spatialization ? AAUDIO_SPATIALIZATION_BEHAVIOR_NEVER : AAUDIO_SPATIALIZATION_BEHAVIOR_AUTO);

	// configure the sample format and channel map
	rc = configure_aaudio_sf(bld, sf, channel_map, &op.remap);
	if (rc) {
		d_print("configure format failed (%d - %s)\n", rc, AAudio_convertResultToText(rc));
		return -OP_ERROR_AAUDIO(rc);
	}
	if (op.remap) {
		d_print("allocating %zu bytes for remap buffer\n", (size_t) AAudioStream_getBufferCapacityInFrames(op.stream) * (size_t) sf_get_frame_size(sf));
		op.remap_buf = xmalloc((size_t) AAudioStream_getBufferCapacityInFrames(op.stream) * (size_t) sf_get_frame_size(sf));
	}
	op.sf = sf;

	// open the stream
	op.error = 0;
	rc = AAudioStreamBuilder_openStream(bld, &op.stream);
	if (rc) {
		d_print("open stream failed (%d - %s)\n", rc, AAudio_convertResultToText(rc));
		if (op.remap_buf) {
			free(op.remap_buf);
			op.remap_buf = NULL;
		}
		if (op.remap) {
			free(op.remap);
			op.remap = NULL;
		}
		AAudioStreamBuilder_delete(bld);
		return -OP_ERROR_AAUDIO(rc);
	}
	op.device = AAudioStream_getDeviceId(op.stream);

	d_print("optimal buffer frames = %d\n", AAudioStream_getFramesPerBurst(op.stream));
	d_print("buffer capacity frames = %d\n", AAudioStream_getBufferCapacityInFrames(op.stream));


	// cleanup the stream builder
	rc = AAudioStreamBuilder_delete(bld);
	if (rc) {
		d_print("delete stream builder failed (%d - %s)\n", rc, AAudio_convertResultToText(rc));
		if (op.remap_buf) {
			free(op.remap_buf);
			op.remap_buf = NULL;
		}
		if (op.remap) {
			free(op.remap);
			op.remap = NULL;
		}
		AAudioStream_close(op.stream);
		return -OP_ERROR_AAUDIO(rc);
	}

	// done (we don't actually start the stream until the first write)
	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_close(void)
{
	if (op.remap_buf) {
		free(op.remap_buf);
		op.remap_buf = NULL;
	}
	if (op.remap) {
		free(op.remap);
		op.remap = NULL;
	}
	if (op.stream) {
		AAudioStream_close(op.stream);
		op.stream = NULL;
	}
	return OP_ERROR_SUCCESS;
}



REQUIRES_API(AAUDIO_MINIMUM_API)
static aaudio_result_t do_state_change(aaudio_result_t (*request)(AAudioStream *strm), aaudio_stream_state_t state, aaudio_stream_state_t state2)
{
	aaudio_result_t rc;

	if (op.error) {
		rc = op.error;
		return rc;
	}

	if (request) {
		d_print("request state change\n");
		rc = request(op.stream);
		if (rc) {
			return rc;
		}
	}

	d_print("wait state change (%d:%s || %d:%s)\n", state, AAudio_convertStreamStateToText(state), state2, AAudio_convertStreamStateToText(state2));
	aaudio_stream_state_t currentState = AAUDIO_STREAM_STATE_UNKNOWN;
	aaudio_stream_state_t inputState = currentState;
	rc = AAUDIO_OK;
	while (rc == AAUDIO_OK && currentState != state && (state2 == 0 || currentState != state2)) {
		// this is required to prevent hanging during pause_on_output_change
		if (op.error) {
			rc = op.error;
			break;
		}
		if (currentState == AAUDIO_STREAM_STATE_CLOSING || currentState == AAUDIO_STREAM_STATE_CLOSED || currentState == AAUDIO_STREAM_STATE_DISCONNECTED) {
			rc = AAUDIO_ERROR_DISCONNECTED;
			break;
		}
		d_print("current state change %d\r\n", currentState);
		rc = AAudioStream_waitForStateChange(op.stream, inputState, &currentState, INT64_MAX);
		inputState = currentState;
	}
	if (rc) {
		d_print("failed state change (%d - %s) [current=%d:%s]\n", rc, AAudio_convertResultToText(rc), currentState, AAudio_convertStreamStateToText(currentState));
	} else {
		d_print("done state change [current=%d:%s]\n", currentState, AAudio_convertStreamStateToText(currentState));
	}
	return rc;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_drop(void)
{
	aaudio_result_t rc;
	aaudio_stream_state_t orig_state = AAudioStream_getState(op.stream);

	// we can't flush if it's closing
	if (orig_state == AAUDIO_STREAM_STATE_CLOSING || orig_state == AAUDIO_STREAM_STATE_CLOSED) {
		return -OP_ERROR_NOT_OPEN;
	}

	// only flush if it isn't already flushed or closed
	if (orig_state != AAUDIO_STREAM_STATE_FLUSHED) {

		// the stream must be paused to be flushed
		if (orig_state == AAUDIO_STREAM_STATE_STARTED || orig_state == AAUDIO_STREAM_STATE_STARTING) {
			rc = do_state_change(AAudioStream_requestPause, AAUDIO_STREAM_STATE_PAUSED, 0);
			if (rc) {
				return -OP_ERROR_AAUDIO(rc);
			}
			// the stream will be started again on the first write
		}

		// flush the stream
		rc = do_state_change(AAudioStream_requestFlush, AAUDIO_STREAM_STATE_FLUSHED, 0);
		if (rc) {
			return -OP_ERROR_AAUDIO(rc);
		}
	}

	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_write(const char *buf, int count)
{
	int32_t device;
	aaudio_result_t rc;
	aaudio_stream_state_t state;

	// if the stream errored, return an error so cmus restarts the output
	// plugin
	//
	// note that this isn't strictly required since AAudioStream_write will
	// return an error on stream disconnection, which will cause cmus to
	// reopen the output plugin
	//
	// https://github.com/google/oboe/wiki/TechNote_Disconnect
	if (op.error) {
		return -OP_ERROR_AAUDIO(op.error);
	}

	// note: this is cheap; it's just a field getter internally
	device = AAudioStream_getDeviceId(op.stream);
	if (op.device != device) {
		if (op.device != -1) {
			notify_via_pipe(mixer_notify_output_in);
		}
		op.device = device;
	}

	// start the stream on the first write (rather than after opening or
	// flushing since cmus may not always use the stream and starting a
	// stream is somewhat expensive)
	//
	// note: this is cheap; it's just a atomic field getter internally
	state = AAudioStream_getState(op.stream);
	if (state == AAUDIO_STREAM_STATE_CLOSING || state == AAUDIO_STREAM_STATE_CLOSED) {
		return -OP_ERROR_NOT_OPEN;
	}
	if (state != AAUDIO_STREAM_STATE_STARTING && state != AAUDIO_STREAM_STATE_STARTED) {
		rc = do_state_change(AAudioStream_requestStart, AAUDIO_STREAM_STATE_STARTED, AAUDIO_STREAM_STATE_STARTING);
		if (rc) {
			return -OP_ERROR_AAUDIO(rc);
		}
	}

	// this should never happen since op_aaudio_buffer_space should always
	// be less than AAudioStream_getBufferCapacityInFrames, and cmus
	// determines how much to write using it
	BUG_ON(count >= AAudioStream_getBufferCapacityInFrames(op.stream) * sf_get_frame_size(op.sf));

	// remap if necessary
	buf = (char *) apply_channel_remap((uint8_t *) op.remap_buf, (uint8_t *) buf, count, op.sf, op.remap);

	// synchronously write the samples to the buffer
	rc = AAudioStream_write(op.stream, buf, count / (int) sf_get_frame_size(op.sf), INT64_MAX);
	if (rc < 0) {
		d_print("write %d = error %d - %s [device=%d] [state=%d]\n", count / (int) sf_get_frame_size(op.sf), rc, AAudio_convertResultToText(rc), device, state);
		return -OP_ERROR_AAUDIO(rc);
	}
	d_print("write %d = %d (* %d bytes) [device=%d] [state=%d]\n", count / (int) sf_get_frame_size(op.sf), rc, (int) sf_get_frame_size(op.sf), device, state);

	// return the number of bytes we write
	return (int) sf_get_frame_size(op.sf) * rc;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_pause(void)
{
	// request stream pause, wait until it completes
	return -OP_ERROR_AAUDIO(do_state_change(AAudioStream_requestPause, AAUDIO_STREAM_STATE_PAUSED, 0));
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_unpause(void)
{
	// request stream start, wait until it starts to start (i.e., will start
	// consuming frames written to it)
	return -OP_ERROR_AAUDIO(do_state_change(AAudioStream_requestStart, AAUDIO_STREAM_STATE_STARTED, AAUDIO_STREAM_STATE_STARTING));
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_buffer_space(void)
{
	int32_t optimal, nonblock;

	// optimal buffer amount (anecdotally, this generally seems to be less
	// than half the buffer capacity)
	optimal = AAudioStream_getFramesPerBurst(op.stream) * (int32_t) sf_get_frame_size(op.sf);

	// max buffer amount (without blocking)
	nonblock = AAudioStream_getBufferSizeInFrames(op.stream) * (int32_t) sf_get_frame_size(op.sf);

	// want to write the optimal amount (up to the nonblock amount)
	return optimal < nonblock ? optimal : nonblock;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_mixer_init(void)
{
	if (!aaudio_supported()) {
		// skip the output plugin (see op_select_any)
		return -OP_ERROR_NOT_SUPPORTED;
	}
	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_mixer_exit(void)
{
	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_mixer_open(int *volume_max)
{
	*volume_max = UINT16_MAX;

	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_mixer_close(void)
{
	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_mixer_get_fds(int what, int *fds)
{
	switch (what) {
	case MIXER_FDS_OUTPUT:
		fds[0] = mixer_notify_output_out;
		return 1;
	default:
		return 0;
	}
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_mixer_set_volume(int l, int r)
{
	return -OP_ERROR_NOT_SUPPORTED;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
static int op_aaudio_mixer_get_volume(int *l, int *r)
{
	// aaudio doesn't support volume control, so say the volume is 100%
	*l = *r = UINT16_MAX;

	return OP_ERROR_SUCCESS;
}

REQUIRES_API(AAUDIO_MINIMUM_API)
const struct output_plugin_ops op_pcm_ops = {
	.init = op_aaudio_init,
	.exit = op_aaudio_exit,
	.open = op_aaudio_open,
	.close = op_aaudio_close,
	.drop = op_aaudio_drop,
	.write = op_aaudio_write,
	.pause = op_aaudio_pause,
	.unpause = op_aaudio_unpause,
	.buffer_space = op_aaudio_buffer_space,
};

REQUIRES_API(AAUDIO_MINIMUM_API)
const struct mixer_plugin_ops op_mixer_ops = {
	.init = op_aaudio_mixer_init,
	.exit = op_aaudio_mixer_exit,
	.open = op_aaudio_mixer_open,
	.close = op_aaudio_mixer_close,
	.get_fds.abi_2 = op_aaudio_mixer_get_fds,
	.set_volume = op_aaudio_mixer_set_volume,
	.get_volume = op_aaudio_mixer_get_volume,
};

const struct output_plugin_opt op_pcm_options[] = {
	OPT(op_aaudio, performance_mode),
	OPT(op_aaudio, allowed_capture),
	OPT(op_aaudio, sharing_mode),
	OPT(op_aaudio, disable_spatialization),
	{ NULL },
};

const struct mixer_plugin_opt op_mixer_options[] = {
	{ NULL },
};

const int op_priority = -3; // higher priority than pulse (-2)
const unsigned op_abi_version = OP_ABI_VERSION;