forked from mikebrady/shairport-sync
-
Notifications
You must be signed in to change notification settings - Fork 0
/
Copy pathaudio_alsa.c
1897 lines (1675 loc) · 70.6 KB
/
audio_alsa.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
/*
* libalsa output driver. This file is part of Shairport.
* Copyright (c) Muffinman, Skaman 2013
* Copyright (c) Mike Brady 2014 -- 2019
* All rights reserved.
*
* 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.
*/
#define ALSA_PCM_NEW_HW_PARAMS_API
#include <alsa/asoundlib.h>
#include <inttypes.h>
#include <math.h>
#include <memory.h>
#include <pthread.h>
#include <stdio.h>
#include <unistd.h>
#include "config.h"
#include "common.h"
#include "activity_monitor.h"
#include "audio.h"
enum alsa_backend_mode {
abm_disconnected,
abm_connected,
abm_playing
} alsa_backend_state; // under the control of alsa_mutex
typedef struct {
snd_pcm_format_t alsa_code;
int frame_size;
} format_record;
static void help(void);
static int init(int argc, char **argv);
static void deinit(void);
static void start(int i_sample_rate, int i_sample_format);
static int play(void *buf, int samples);
static void stop(void);
static void flush(void);
int delay(long *the_delay);
int get_rate_information(uint64_t *elapsed_time, uint64_t *frames_played);
void *alsa_buffer_monitor_thread_code(void *arg);
static void volume(double vol);
void do_volume(double vol);
int prepare(void);
static void parameters(audio_parameters *info);
int mute(int do_mute); // returns true if it actually is allowed to use the mute
static double set_volume;
static int output_method_signalled = 0; // for reporting whether it's using mmap or not
int delay_type_notified = -1; // for controlling the reporting of whether the output device can do precison delays (e.g. alsa->pulsaudio virtual devices can't)
int use_monotonic_clock = 0; // this value will be set when the hardware is initialised
audio_output audio_alsa = {
.name = "alsa",
.help = &help,
.init = &init,
.deinit = &deinit,
.prepare = &prepare,
.start = &start,
.stop = &stop,
.is_running = NULL,
.flush = &flush,
.delay = &delay,
.play = &play,
.rate_info = &get_rate_information,
.mute = NULL, // a function will be provided if it can, and is allowed to,
// do hardware mute
.volume = NULL, // a function will be provided if it can do hardware volume
.parameters = NULL}; // a function will be provided if it can do hardware volume
static pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t alsa_mixer_mutex = PTHREAD_MUTEX_INITIALIZER;
pthread_t alsa_buffer_monitor_thread;
// for deciding when to activate mute
// there are two sources of requests to mute -- the backend itself, e.g. when it
// is flushing
// and the player, e.g. when volume goes down to -144, i.e. mute.
// we may not be allowed to use hardware mute, so we must reflect that too.
int mute_requested_externally = 0;
int mute_requested_internally = 0;
// for tracking how long the output device has stalled
uint64_t stall_monitor_start_time; // zero if not initialised / not started /
// zeroed by flush
long stall_monitor_frame_count; // set to delay at start of time, incremented by
// any writes
uint64_t stall_monitor_error_threshold; // if the time is longer than this, it's
// an error
static snd_output_t *output = NULL;
int frame_size; // in bytes for interleaved stereo
int alsa_device_initialised; // boolean to ensure the initialisation is only
// done once
snd_pcm_t *alsa_handle = NULL;
static snd_pcm_hw_params_t *alsa_params = NULL;
static snd_pcm_sw_params_t *alsa_swparams = NULL;
static snd_ctl_t *ctl = NULL;
static snd_ctl_elem_id_t *elem_id = NULL;
static snd_mixer_t *alsa_mix_handle = NULL;
static snd_mixer_elem_t *alsa_mix_elem = NULL;
static snd_mixer_selem_id_t *alsa_mix_sid = NULL;
static long alsa_mix_minv, alsa_mix_maxv;
static long alsa_mix_mindb, alsa_mix_maxdb;
static char *alsa_out_dev = "default";
static char *alsa_mix_dev = NULL;
static char *alsa_mix_ctrl = "Master";
static int alsa_mix_index = 0;
static int hardware_mixer = 0;
static int has_softvol = 0;
int64_t dither_random_number_store = 0;
static int volume_set_request = 0; // set when an external request is made to set the volume.
int mixer_volume_setting_gives_mute = 0; // set when it is discovered that
// particular mixer volume setting
// causes a mute.
long alsa_mix_mute; // setting the volume to this value mutes output, if
// mixer_volume_setting_gives_mute is true
int volume_based_mute_is_active =
0; // set when muting is being done by a setting the volume to a magic value
// use this to allow the use of snd_pcm_writei or snd_pcm_mmap_writei
snd_pcm_sframes_t (*alsa_pcm_write)(snd_pcm_t *, const void *,
snd_pcm_uframes_t) = snd_pcm_writei;
int precision_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, enum yndk_type *using_update_timestamps);
int standard_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, enum yndk_type *using_update_timestamps);
// use this to allow the use of standard or precision delay calculations, with standard the, uh, standard.
int (*delay_and_status)(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, enum yndk_type *using_update_timestamps) = standard_delay_and_status;
int precision_delay_available() {
// this is very crude -- if the device is a hardware device, then it's assumed the delay is precise
const char *output_device_name = snd_pcm_name(alsa_handle);
return (strstr(output_device_name,"hw:") == output_device_name);
}
// static int play_number;
// static int64_t accumulated_delay, accumulated_da_delay;
int alsa_characteristics_already_listed = 0;
static snd_pcm_uframes_t period_size_requested, buffer_size_requested;
static int set_period_size_request, set_buffer_size_request;
static uint64_t measurement_start_time;
static uint64_t frames_played_at_measurement_start_time;
static uint64_t measurement_time;
static uint64_t frames_played_at_measurement_time;
volatile uint64_t most_recent_write_time;
static uint64_t frames_sent_for_playing;
static uint64_t frame_index;
static int measurement_data_is_valid;
static void help(void) {
printf(" -d output-device set the output device, default is \"default\".\n"
" -c mixer-control set the mixer control name, default is to use no mixer.\n"
" -m mixer-device set the mixer device, default is the output device.\n"
" -i mixer-index set the mixer index, default is 0.\n");
system("if [ -d /proc/asound ] ; then echo \" hardware output devices:\" ; ls -al /proc/asound/ 2>/dev/null | grep '\\->' | tr -s ' ' | cut -d ' ' -f 9 | while read line; do echo \" \\\"hw:$line\\\"\" ; done ; fi");
}
void set_alsa_out_dev(char *dev) { alsa_out_dev = dev; }
// assuming pthread cancellation is disabled
int open_mixer() {
int response = 0;
if (hardware_mixer) {
debug(3, "Open Mixer");
int ret = 0;
snd_mixer_selem_id_alloca(&alsa_mix_sid);
snd_mixer_selem_id_set_index(alsa_mix_sid, alsa_mix_index);
snd_mixer_selem_id_set_name(alsa_mix_sid, alsa_mix_ctrl);
if ((snd_mixer_open(&alsa_mix_handle, 0)) < 0) {
debug(1, "Failed to open mixer");
response = -1;
} else {
debug(3, "Mixer device name is \"%s\".", alsa_mix_dev);
if ((snd_mixer_attach(alsa_mix_handle, alsa_mix_dev)) < 0) {
debug(1, "Failed to attach mixer");
response = -2;
} else {
if ((snd_mixer_selem_register(alsa_mix_handle, NULL, NULL)) < 0) {
debug(1, "Failed to register mixer element");
response = -3;
} else {
ret = snd_mixer_load(alsa_mix_handle);
if (ret < 0) {
debug(1, "Failed to load mixer element");
response = -4;
} else {
debug(3, "Mixer Control name is \"%s\".", alsa_mix_ctrl);
alsa_mix_elem = snd_mixer_find_selem(alsa_mix_handle, alsa_mix_sid);
if (!alsa_mix_elem) {
warn("failed to find mixer control \"%s\".", alsa_mix_ctrl);
response = -5;
} else {
response = 1; // we found a hardware mixer and successfully opened it
}
}
}
}
}
}
return response;
}
// assuming pthread cancellation is disabled
void close_mixer() {
if (alsa_mix_handle) {
snd_mixer_close(alsa_mix_handle);
alsa_mix_handle = NULL;
}
}
// assuming pthread cancellation is disabled
void do_snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t *mix_elem, double vol) {
if (snd_mixer_selem_set_playback_dB_all(mix_elem, vol, 0) != 0) {
debug(1, "Can't set playback volume accurately to %f dB.", vol);
if (snd_mixer_selem_set_playback_dB_all(mix_elem, vol, -1) != 0)
if (snd_mixer_selem_set_playback_dB_all(mix_elem, vol, 1) != 0)
debug(1, "Could not set playback dB volume on the mixer.");
}
}
void actual_close_alsa_device() {
debug(1, "actual close");
if (alsa_handle) {
int derr;
if ((derr = snd_pcm_hw_free(alsa_handle)))
debug(1, "Error %d (\"%s\") freeing the output device hardware while "
"closing it.",
derr, snd_strerror(derr));
if ((derr = snd_pcm_close(alsa_handle)))
debug(1, "Error %d (\"%s\") closing the output device.", derr, snd_strerror(derr));
alsa_handle = NULL;
}
}
// This array is a sequence of the output rates to be tried if automatic speed selection is requested.
// There is no benefit to upconverting the frame rate, other than for compatibility.
// The lowest rate that the DAC is capable of is chosen.
unsigned int auto_speed_output_rates[] = {
44100,
88200,
176400,
352800,
};
// This array is of all the formats known to Shairport Sync, in order of the SPS_FORMAT definitions, with their equivalent alsa codes and their frame sizes.
// If just one format is requested, then its entry is searched for in the array and checked on the device
// If auto format is requested, then each entry in turn is tried until a working format is found.
// So, it should be in the search order.
format_record fr[] = {
{SND_PCM_FORMAT_UNKNOWN,0}, // unknown
{SND_PCM_FORMAT_S8,2},
{SND_PCM_FORMAT_U8,2},
{SND_PCM_FORMAT_S16,4},
{SND_PCM_FORMAT_S16_LE,4},
{SND_PCM_FORMAT_S16_BE,4},
{SND_PCM_FORMAT_S24,4},
{SND_PCM_FORMAT_S24_LE,8},
{SND_PCM_FORMAT_S24_BE,8},
{SND_PCM_FORMAT_S24_3LE,6},
{SND_PCM_FORMAT_S24_3BE,6},
{SND_PCM_FORMAT_S32,8},
{SND_PCM_FORMAT_S32_LE,8},
{SND_PCM_FORMAT_S32_BE,8},
{SND_PCM_FORMAT_UNKNOWN,0}, // auto
{SND_PCM_FORMAT_UNKNOWN,0}, // illegal
};
// This array is the sequence of formats to be tried if automatic selection of the format is requested.
// Ideally, audio should pass through Shairport Sync unaltered, apart from occasional interpolation.
// If the user chooses a hardware mixer, then audio could go straight through, unaltered, as signed 16 bit stereo.
// However, the user might, at any point, select an option that requires modification, such as stereo to mono mixing,
// additional volume attenuation, convolution, and so on. For this reason,
// we look for the greatest depth the DAC is capable of, since upconverting it is completely lossless.
// If audio processing is required, then the dither that must be added will
// be added at the lowest possible level.
// Hence, selecting the greatest bit depth is always either beneficial or neutral.
enum sps_format_t auto_format_check_sequence[] = {
SPS_FORMAT_S32,
SPS_FORMAT_S32_LE,
SPS_FORMAT_S32_BE,
SPS_FORMAT_S24,
SPS_FORMAT_S24_LE,
SPS_FORMAT_S24_BE,
SPS_FORMAT_S24_3LE,
SPS_FORMAT_S24_3BE,
SPS_FORMAT_S16,
SPS_FORMAT_S16_LE,
SPS_FORMAT_S16_BE,
SPS_FORMAT_S8,
SPS_FORMAT_U8,
};
// assuming pthread cancellation is disabled
// if do_auto_setting is true and auto format or auto speed has been requested,
// select the settings as appropriate and store them
int actual_open_alsa_device(int do_auto_setup) {
// the alsa mutex is already acquired when this is called
const snd_pcm_uframes_t minimal_buffer_headroom =
352 * 2; // we accept this much headroom in the hardware buffer, but we'll
// accept less
/*
const snd_pcm_uframes_t requested_buffer_headroom =
minimal_buffer_headroom + 2048; // we ask for this much headroom in the
// hardware buffer, but we'll accept
less
*/
int ret, dir = 0;
unsigned int actual_sample_rate; // this will be given the rate requested and will be given the actual rate
// snd_pcm_uframes_t frames = 441 * 10;
snd_pcm_uframes_t actual_buffer_length;
snd_pcm_access_t access;
// ensure no calls are made to the alsa device enquiring about the buffer
// length if
// synchronisation is disabled.
if (config.no_sync != 0)
audio_alsa.delay = NULL;
// ensure no calls are made to the alsa device enquiring about the buffer
// length if
// synchronisation is disabled.
if (config.no_sync != 0)
audio_alsa.delay = NULL;
ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0);
if (ret < 0) {
if (ret == -ENOENT) {
warn("the alsa output_device \"%s\" can not be found.", alsa_out_dev);
} else {
char errorstring[1024];
strerror_r(-ret, (char *)errorstring, sizeof(errorstring));
warn("alsa: error %d (\"%s\") opening alsa device \"%s\".", ret, (char *)errorstring, alsa_out_dev);
}
return ret;
}
snd_pcm_hw_params_alloca(&alsa_params);
snd_pcm_sw_params_alloca(&alsa_swparams);
ret = snd_pcm_hw_params_any(alsa_handle, alsa_params);
if (ret < 0) {
die("audio_alsa: Broken configuration for device \"%s\": no configurations "
"available",
alsa_out_dev);
return ret;
}
if ((config.no_mmap == 0) &&
(snd_pcm_hw_params_set_access(alsa_handle, alsa_params, SND_PCM_ACCESS_MMAP_INTERLEAVED) >=
0)) {
if (output_method_signalled == 0) {
debug(3, "Output written using MMAP");
output_method_signalled = 1;
}
access = SND_PCM_ACCESS_MMAP_INTERLEAVED;
alsa_pcm_write = snd_pcm_mmap_writei;
} else {
if (output_method_signalled == 0) {
debug(3, "Output written with RW");
output_method_signalled = 1;
}
access = SND_PCM_ACCESS_RW_INTERLEAVED;
alsa_pcm_write = snd_pcm_writei;
}
ret = snd_pcm_hw_params_set_access(alsa_handle, alsa_params, access);
if (ret < 0) {
warn("audio_alsa: Access type not available for device \"%s\": %s", alsa_out_dev,
snd_strerror(ret));
return ret;
}
ret = snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2);
if (ret < 0) {
warn("audio_alsa: Channels count (2) not available for device \"%s\": %s", alsa_out_dev,
snd_strerror(ret));
return ret;
}
snd_pcm_format_t sf;
if ((do_auto_setup == 0) || (config.output_format_auto_requested == 0)) { // no auto format
if ((config.output_format > SPS_FORMAT_UNKNOWN) && (config.output_format < SPS_FORMAT_AUTO)) {
sf = fr[config.output_format].alsa_code;
frame_size = fr[config.output_format].frame_size;
} else {
warn("alsa: unexpected output format %d. Set to S16_LE.",config.output_format);
config.output_format = SPS_FORMAT_S16_LE;
sf = fr[config.output_format].alsa_code;
frame_size = fr[config.output_format].frame_size;
}
ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params, sf);
if (ret < 0) {
warn("audio_alsa: Alsa sample format %d not available for device \"%s\": %s", sf,
alsa_out_dev, snd_strerror(ret));
return ret;
}
} else { // auto format
int number_of_formats_to_try;
enum sps_format_t *formats;
formats = auto_format_check_sequence;
number_of_formats_to_try = sizeof(auto_format_check_sequence)/sizeof(sps_format_t);
int i = 0;
int format_found = 0;
enum sps_format_t trial_format = SPS_FORMAT_UNKNOWN;
while ((i < number_of_formats_to_try) && (format_found == 0)) {
trial_format = formats[i];
sf = fr[trial_format].alsa_code;
frame_size = fr[trial_format].frame_size;
ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params, sf);
if (ret == 0)
format_found = 1;
else
i++;
}
if (ret == 0) {
config.output_format = trial_format;
debug(1,"alsa: output format chosen is \"%s\".",sps_format_description_string(config.output_format));
} else {
warn("audio_alsa: Could not automatically set the output format for device \"%s\": %s",
alsa_out_dev, snd_strerror(ret));
return ret;
}
}
if ((do_auto_setup == 0) || (config.output_rate_auto_requested == 0)) { // no auto format
actual_sample_rate = config.output_rate; // this is the requested rate -- it'll be changed to the actual rate
ret = snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, &actual_sample_rate, &dir);
if (ret < 0) {
warn("audio_alsa: Rate %iHz not available for playback: %s", config.output_rate,
snd_strerror(ret));
return ret;
}
} else {
int number_of_speeds_to_try;
unsigned int *speeds;
speeds = auto_speed_output_rates;
number_of_speeds_to_try = sizeof(auto_speed_output_rates)/sizeof(int);
int i = 0;
int speed_found = 0;
while ((i < number_of_speeds_to_try) && (speed_found == 0)) {
actual_sample_rate = speeds[i];
ret = snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, &actual_sample_rate, &dir);
if (ret == 0) {
speed_found = 1;
if (actual_sample_rate != speeds[i])
warn("Speed requested: %d. Speed available: %d.",speeds[i],actual_sample_rate);
} else {
i++;
}
}
if (ret == 0) {
config.output_rate = actual_sample_rate;
debug(1,"alsa: output speed chosen is %d.",config.output_rate);
} else {
warn("audio_alsa: Could not automatically set the output rate for device \"%s\": %s",
alsa_out_dev, snd_strerror(ret));
return ret;
}
}
if (set_period_size_request != 0) {
debug(1, "Attempting to set the period size");
ret = snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params, &period_size_requested,
&dir);
if (ret < 0) {
warn("audio_alsa: cannot set period size of %lu: %s", period_size_requested,
snd_strerror(ret));
return ret;
} else {
snd_pcm_uframes_t actual_period_size;
snd_pcm_hw_params_get_period_size(alsa_params, &actual_period_size, &dir);
if (actual_period_size != period_size_requested)
inform("Actual period size set to a different value than requested. "
"Requested: %lu, actual "
"setting: %lu",
period_size_requested, actual_period_size);
}
}
if (set_buffer_size_request != 0) {
debug(1, "Attempting to set the buffer size to %lu", buffer_size_requested);
ret = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params, &buffer_size_requested);
if (ret < 0) {
warn("audio_alsa: cannot set buffer size of %lu: %s", buffer_size_requested,
snd_strerror(ret));
return ret;
} else {
snd_pcm_uframes_t actual_buffer_size;
snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_size);
if (actual_buffer_size != buffer_size_requested)
inform("Actual period size set to a different value than requested. "
"Requested: %lu, actual "
"setting: %lu",
buffer_size_requested, actual_buffer_size);
}
}
ret = snd_pcm_hw_params(alsa_handle, alsa_params);
if (ret < 0) {
warn("audio_alsa: Unable to set hw parameters for device \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
// check parameters after attempting to set them
if (set_period_size_request != 0) {
snd_pcm_uframes_t actual_period_size;
snd_pcm_hw_params_get_period_size(alsa_params, &actual_period_size, &dir);
if (actual_period_size != period_size_requested)
inform("Actual period size set to a different value than requested. "
"Requested: %lu, actual "
"setting: %lu",
period_size_requested, actual_period_size);
}
if (set_buffer_size_request != 0) {
snd_pcm_uframes_t actual_buffer_size;
snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_size);
if (actual_buffer_size != buffer_size_requested)
inform("Actual period size set to a different value than requested. "
"Requested: %lu, actual "
"setting: %lu",
buffer_size_requested, actual_buffer_size);
}
if (actual_sample_rate != config.output_rate) {
warn("Can't set the D/A converter to sample rate %d.", config.output_rate);
return -EINVAL;
}
use_monotonic_clock = snd_pcm_hw_params_is_monotonic(alsa_params);
ret = snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_length);
if (ret < 0) {
warn("audio_alsa: Unable to get hw buffer length for device \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
ret = snd_pcm_sw_params_current(alsa_handle, alsa_swparams);
if (ret < 0) {
warn("audio_alsa: Unable to get current sw parameters for device \"%s\": "
"%s.",
alsa_out_dev, snd_strerror(ret));
return ret;
}
ret = snd_pcm_sw_params_set_tstamp_mode(alsa_handle, alsa_swparams, SND_PCM_TSTAMP_ENABLE);
if (ret < 0) {
warn("audio_alsa: Can't enable timestamp mode of device: \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
/* write the sw parameters */
ret = snd_pcm_sw_params(alsa_handle, alsa_swparams);
if (ret < 0) {
warn("audio_alsa: Unable to set software parameters of device: \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
ret = snd_pcm_prepare(alsa_handle);
if (ret < 0) {
warn("audio_alsa: Unable to prepare the device: \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
if (actual_buffer_length < config.audio_backend_buffer_desired_length + minimal_buffer_headroom) {
/*
// the dac buffer is too small, so let's try to set it
buffer_size =
config.audio_backend_buffer_desired_length + requested_buffer_headroom;
ret = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params,
&buffer_size);
if (ret < 0)
die("audio_alsa: Unable to set hw buffer size to %lu for device \"%s\": "
"%s.",
config.audio_backend_buffer_desired_length +
requested_buffer_headroom,
alsa_out_dev, snd_strerror(ret));
if (config.audio_backend_buffer_desired_length + minimal_buffer_headroom >
buffer_size) {
die("audio_alsa: Can't set hw buffer size to %lu or more for device "
"\"%s\". Requested size: %lu, granted size: %lu.",
config.audio_backend_buffer_desired_length + minimal_buffer_headroom,
alsa_out_dev, config.audio_backend_buffer_desired_length +
requested_buffer_headroom,
buffer_size);
}
*/
debug(1, "The alsa buffer is smaller (%lu bytes) than the desired backend "
"buffer "
"length (%ld) you have chosen.",
actual_buffer_length, config.audio_backend_buffer_desired_length);
}
if (config.use_precision_timing == YNA_YES)
delay_and_status = precision_delay_and_status;
else if (config.use_precision_timing == YNA_AUTO) {
if (precision_delay_available()) {
delay_and_status = precision_delay_and_status;
debug(2,"alsa: precision timing selected for \"auto\" mode");
}
}
if (alsa_characteristics_already_listed == 0) {
alsa_characteristics_already_listed = 1;
int log_level = 2; // the level at which debug information should be output
// int rc;
snd_pcm_access_t access_type;
snd_pcm_format_t format_type;
snd_pcm_subformat_t subformat_type;
// unsigned int val, val2;
unsigned int uval, uval2;
int sval;
int dir;
snd_pcm_uframes_t frames;
debug(log_level, "PCM handle name = '%s'", snd_pcm_name(alsa_handle));
// ret = snd_pcm_hw_params_any(alsa_handle, alsa_params);
// if (ret < 0) {
// die("audio_alsa: Cannpot get configuration for
// device
//\"%s\":
// no
// configurations
//"
// "available",
// alsa_out_dev);
// }
debug(log_level, "alsa device parameters:");
snd_pcm_hw_params_get_access(alsa_params, &access_type);
debug(log_level, " access type = %s", snd_pcm_access_name(access_type));
snd_pcm_hw_params_get_format(alsa_params, &format_type);
debug(log_level, " format = '%s' (%s)", snd_pcm_format_name(format_type),
snd_pcm_format_description(format_type));
snd_pcm_hw_params_get_subformat(alsa_params, &subformat_type);
debug(log_level, " subformat = '%s' (%s)", snd_pcm_subformat_name(subformat_type),
snd_pcm_subformat_description(subformat_type));
snd_pcm_hw_params_get_channels(alsa_params, &uval);
debug(log_level, " number of channels = %u", uval);
sval = snd_pcm_hw_params_get_sbits(alsa_params);
debug(log_level, " number of significant bits = %d", sval);
snd_pcm_hw_params_get_rate(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " rate = %u frames per second (<).", uval);
break;
case 0:
debug(log_level, " rate = %u frames per second (precisely).", uval);
break;
case 1:
debug(log_level, " rate = %u frames per second (>).", uval);
break;
}
if (snd_pcm_hw_params_get_rate_numden(alsa_params, &uval, &uval2) == 0)
debug(log_level, " precise (rational) rate = %.3f frames per second (i.e. %u/%u).", uval,
uval2, ((double)uval) / uval2);
else
debug(log_level, " precise (rational) rate information unavailable.");
snd_pcm_hw_params_get_period_time(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " period_time = %u us (<).", uval);
break;
case 0:
debug(log_level, " period_time = %u us (precisely).", uval);
break;
case 1:
debug(log_level, " period_time = %u us (>).", uval);
break;
}
snd_pcm_hw_params_get_period_size(alsa_params, &frames, &dir);
switch (dir) {
case -1:
debug(log_level, " period_size = %lu frames (<).", frames);
break;
case 0:
debug(log_level, " period_size = %lu frames (precisely).", frames);
break;
case 1:
debug(log_level, " period_size = %lu frames (>).", frames);
break;
}
snd_pcm_hw_params_get_buffer_time(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " buffer_time = %u us (<).", uval);
break;
case 0:
debug(log_level, " buffer_time = %u us (precisely).", uval);
break;
case 1:
debug(log_level, " buffer_time = %u us (>).", uval);
break;
}
snd_pcm_hw_params_get_buffer_size(alsa_params, &frames);
switch (dir) {
case -1:
debug(log_level, " buffer_size = %lu frames (<).", frames);
break;
case 0:
debug(log_level, " buffer_size = %lu frames (precisely).", frames);
break;
case 1:
debug(log_level, " buffer_size = %lu frames (>).", frames);
break;
}
snd_pcm_hw_params_get_periods(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " periods_per_buffer = %u (<).", uval);
break;
case 0:
debug(log_level, " periods_per_buffer = %u (precisely).", uval);
break;
case 1:
debug(log_level, " periods_per_buffer = %u (>).", uval);
break;
}
}
return 0;
}
int open_alsa_device(int do_auto_setup) {
int result;
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
result = actual_open_alsa_device(do_auto_setup);
pthread_setcancelstate(oldState, NULL);
return result;
}
int do_alsa_device_init_if_needed() {
int response = 0;
// do any alsa device initialisation (general case) if needed
// at present, this is only needed if a hardware mixer is being used
// if there's a hardware mixer, it needs to be initialised before first use
if (alsa_device_initialised == 0) {
alsa_device_initialised = 1;
if (hardware_mixer) {
debug(2, "alsa: hardware mixer init");
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
if (alsa_mix_dev == NULL)
alsa_mix_dev = alsa_out_dev;
// Now, start trying to initialise the alsa device with the settings
// obtained
pthread_cleanup_debug_mutex_lock(&alsa_mixer_mutex, 1000, 1);
if (open_mixer() == 1) {
if (snd_mixer_selem_get_playback_volume_range(alsa_mix_elem, &alsa_mix_minv,
&alsa_mix_maxv) < 0)
debug(1, "Can't read mixer's [linear] min and max volumes.");
else {
if (snd_mixer_selem_get_playback_dB_range(alsa_mix_elem, &alsa_mix_mindb,
&alsa_mix_maxdb) == 0) {
audio_alsa.volume = &volume; // insert the volume function now we
// know it can do dB stuff
audio_alsa.parameters = ¶meters; // likewise the parameters stuff
if (alsa_mix_mindb == SND_CTL_TLV_DB_GAIN_MUTE) {
// For instance, the Raspberry Pi does this
debug(1, "Lowest dB value is a mute");
mixer_volume_setting_gives_mute = 1;
alsa_mix_mute = SND_CTL_TLV_DB_GAIN_MUTE; // this may not be
// necessary -- it's
// always
// going to be SND_CTL_TLV_DB_GAIN_MUTE, right?
// debug(1, "Try minimum volume + 1 as lowest true attenuation
// value");
if (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv + 1,
&alsa_mix_mindb) != 0)
debug(1, "Can't get dB value corresponding to a minimum volume "
"+ 1.");
}
debug(3, "Hardware mixer has dB volume from %f to %f.", (1.0 * alsa_mix_mindb) / 100.0,
(1.0 * alsa_mix_maxdb) / 100.0);
} else {
// use the linear scale and do the db conversion ourselves
warn("The hardware mixer specified -- \"%s\" -- does not have "
"a dB volume scale.",
alsa_mix_ctrl);
if (snd_ctl_open(&ctl, alsa_mix_dev, 0) < 0) {
warn("Cannot open control \"%s\"", alsa_mix_dev);
response = -1;
}
if (snd_ctl_elem_id_malloc(&elem_id) < 0) {
debug(1, "Cannot allocate memory for control \"%s\"", alsa_mix_dev);
elem_id = NULL;
response = -2;
} else {
snd_ctl_elem_id_set_interface(elem_id, SND_CTL_ELEM_IFACE_MIXER);
snd_ctl_elem_id_set_name(elem_id, alsa_mix_ctrl);
if (snd_ctl_get_dB_range(ctl, elem_id, &alsa_mix_mindb, &alsa_mix_maxdb) == 0) {
debug(1, "alsa: hardware mixer \"%s\" selected, with dB volume "
"from %f to %f.",
alsa_mix_ctrl, (1.0 * alsa_mix_mindb) / 100.0,
(1.0 * alsa_mix_maxdb) / 100.0);
has_softvol = 1;
audio_alsa.volume = &volume; // insert the volume function now
// we know it can do dB stuff
audio_alsa.parameters = ¶meters; // likewise the parameters stuff
} else {
debug(1, "Cannot get the dB range from the volume control \"%s\"", alsa_mix_ctrl);
}
}
/*
debug(1, "Min and max volumes are %d and
%d.",alsa_mix_minv,alsa_mix_maxv);
alsa_mix_maxdb = 0;
if ((alsa_mix_maxv!=0) && (alsa_mix_minv!=0))
alsa_mix_mindb =
-20*100*(log10(alsa_mix_maxv*1.0)-log10(alsa_mix_minv*1.0));
else if (alsa_mix_maxv!=0)
alsa_mix_mindb = -20*100*log10(alsa_mix_maxv*1.0);
audio_alsa.volume = &linear_volume; // insert the linear volume
function
audio_alsa.parameters = ¶meters; // likewise the parameters
stuff
debug(1,"Max and min dB calculated are %d and
%d.",alsa_mix_maxdb,alsa_mix_mindb);
*/
}
}
if (((config.alsa_use_hardware_mute == 1) &&
(snd_mixer_selem_has_playback_switch(alsa_mix_elem))) ||
mixer_volume_setting_gives_mute) {
audio_alsa.mute = &mute; // insert the mute function now we know it
// can do muting stuff
// debug(1, "Has mixer and mute ability we will use.");
} else {
// debug(1, "Has mixer but not using hardware mute.");
}
close_mixer();
}
debug_mutex_unlock(&alsa_mixer_mutex, 3); // release the mutex
pthread_cleanup_pop(0);
pthread_setcancelstate(oldState, NULL);
}
}
return response;
}
static int init(int argc, char **argv) {
// for debugging
snd_output_stdio_attach(&output, stdout, 0);
// debug(2,"audio_alsa init called.");
int response = 0; // this will be what we return to the caller.
alsa_device_initialised = 0;
const char *str;
int value;
// double dvalue;
// set up default values first
alsa_backend_state = abm_disconnected; // startup state
debug(2, "alsa: init() -- alsa_backend_state => abm_disconnected.");
set_period_size_request = 0;
set_buffer_size_request = 0;
config.alsa_use_hardware_mute = 0; // don't use it by default
config.audio_backend_latency_offset = 0;
config.audio_backend_buffer_desired_length = 0.200;
config.audio_backend_buffer_interpolation_threshold_in_seconds =
0.120; // below this, basic interpolation will be used to save time.
config.alsa_maximum_stall_time = 0.200; // 200 milliseconds -- if it takes longer, it's a problem
config.audio_backend_silence_threshold =
0.040; // start sending silent frames if the delay goes below this time
config.audio_backend_silence_scan_interval = 0.004; // check silence threshold this often
stall_monitor_error_threshold =
(uint64_t)1000000 * config.alsa_maximum_stall_time; // stall time max to microseconds;
stall_monitor_error_threshold = (stall_monitor_error_threshold << 32) / 1000000; // now in fp form
debug(1,
"stall_monitor_error_threshold is 0x%" PRIx64 ", with alsa_maximum_stall_time of %f sec.",
stall_monitor_error_threshold, config.alsa_maximum_stall_time);
stall_monitor_start_time = 0;
stall_monitor_frame_count = 0;
config.disable_standby_mode = disable_standby_off;
config.keep_dac_busy = 0;
config.use_precision_timing = YNA_AUTO;
// get settings from settings file first, allow them to be overridden by
// command line options
// do the "general" audio options. Note, these options are in the "general"
// stanza!
parse_general_audio_options();
if (config.cfg != NULL) {
double dvalue;
/* Get the Output Device Name. */
if (config_lookup_string(config.cfg, "alsa.output_device", &str)) {
alsa_out_dev = (char *)str;
}
/* Get the Mixer Type setting. */
if (config_lookup_string(config.cfg, "alsa.mixer_type", &str)) {
inform("The alsa mixer_type setting is deprecated and has been ignored. "
"FYI, using the \"mixer_control_name\" setting automatically "
"chooses a hardware mixer.");
}
/* Get the Mixer Device Name. */
if (config_lookup_string(config.cfg, "alsa.mixer_device", &str)) {
alsa_mix_dev = (char *)str;
}
/* Get the Mixer Control Name. */
if (config_lookup_string(config.cfg, "alsa.mixer_control_name", &str)) {
alsa_mix_ctrl = (char *)str;
hardware_mixer = 1;
}
/* Get the disable_synchronization setting. */
if (config_lookup_string(config.cfg, "alsa.disable_synchronization", &str)) {
if (strcasecmp(str, "no") == 0)
config.no_sync = 0;
else if (strcasecmp(str, "yes") == 0)
config.no_sync = 1;
else {
warn("Invalid disable_synchronization option choice \"%s\". It should "