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
|
/*
Copyright (C) 2005-2009 Michel de Boer <michel@twinklephone.com>
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 <https://www.gnu.org/licenses/>.
*/
#include "audio_device.h"
#include <iostream>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#include "sys_settings.h"
#include "translator.h"
#include "log.h"
#include "userintf.h"
#include "util.h"
#include "audits/memman.h"
#ifdef HAVE_LIBASOUND
#include <alsa/asoundlib.h>
#endif
t_audio_io* t_audio_io::open(const t_audio_device& dev, bool playback, bool capture, bool blocking, int channels, t_audio_sampleformat format, int sample_rate, bool short_latency)
{
t_audio_io* aio;
if(dev.type == t_audio_device::OSS) {
aio = new t_oss_io();
MEMMAN_NEW(aio);
#ifdef HAVE_LIBASOUND
} else if (dev.type == t_audio_device::ALSA) {
aio = new t_alsa_io();
MEMMAN_NEW(aio);
#endif
} else {
string msg("Audio device not implemented");
log_file->write_report(msg, "t_audio_io::open",
LOG_NORMAL, LOG_CRITICAL);
return 0;
}
if (aio->open(dev.device, playback, capture, blocking, channels, format,
sample_rate, short_latency)) {
return aio;
} else {
string msg("Open audio device failed");
log_file->write_report(msg, "t_audio_io::open",
LOG_NORMAL, LOG_CRITICAL);
MEMMAN_DELETE(aio);
delete aio;
return 0L;
}
}
bool t_audio_io::validate(const t_audio_device& dev, bool playback, bool capture) {
t_audio_io *aio = open(dev, playback, capture, false, 1, SAMPLEFORMAT_S16, 8000, true);
if (aio) {
MEMMAN_DELETE(aio);
delete aio;
return true;
}
return false;
}
t_audio_io::~t_audio_io() {}
int t_audio_io::get_sample_rate(void) const {
return _sample_rate;
}
bool t_audio_io::open(const string& device, bool playback, bool capture, bool blocking, int channels, t_audio_sampleformat format, int sample_rate, bool short_latency)
{
_sample_rate = sample_rate;
return true;
}
t_oss_io::t_oss_io() : fd(-1), play_buffersize(0), rec_buffersize(0) {
}
t_oss_io::~t_oss_io()
{
if (fd > 0) {
int arg = 0;
ioctl(fd, SNDCTL_DSP_RESET, &arg);
close(fd);
}
fd = -1;
}
bool t_oss_io::open(const string& device, bool playback, bool capture, bool blocking, int channels, t_audio_sampleformat format, int sample_rate, bool short_latency)
{
t_audio_io::open(device, playback, capture, blocking, channels, format, sample_rate,
short_latency);
int mode = 0;
int status;
log_file->write_header("t_oss_io::open", LOG_NORMAL);
log_file->write_raw("Opening OSS device: ");
log_file->write_raw(device);
log_file->write_endl();
if (playback) log_file->write_raw("play\n");
if (capture) log_file->write_raw("capture\n");
log_file->write_footer();
assert (playback || capture);
if (playback && capture) mode |= O_RDWR;
else if (playback) mode |= O_WRONLY;
else if (capture) mode |= O_RDONLY;
// On some systems opening the audio devices blocks if another
// process or thread has opened it already. To prevent a deadlock
// first try to open the device in non-blocking mode.
// If the device is still open by another twinkle thread then that
// is a bug, but this way at least non deadlock is caused.
if(blocking) {
fd = ::open(device.c_str(), mode | O_NONBLOCK);
if (fd == -1) {
string msg("OSS audio device open failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
return false;
} else {
close(fd);
fd = -1;
}
} else mode |= O_NONBLOCK;
fd = ::open(device.c_str(), mode);
if (fd < 0) {
string msg("OSS audio device open failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
return false;
}
// Full duplex
if (playback && capture) {
status = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
if (status == -1) {
string msg("SNDCTL_DSP_SETDUPLEX ioctl failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
ui->cb_display_msg(TRANSLATE("Sound card cannot be set to full duplex."),
MSG_CRITICAL);
close(fd);
return false;
}
}
// Set fragment size
int arg;
if (short_latency) {
switch (sys_config->get_oss_fragment_size()) {
case 16:
arg = 0x00ff0004; // 255 buffers of 2^4 bytes each
break;
case 32:
arg = 0x00ff0005; // 255 buffers of 2^5 bytes each
break;
case 64:
arg = 0x00ff0006; // 255 buffers of 2^5 bytes each
break;
case 128:
arg = 0x00ff0007; // 255 buffers of 2^7 bytes each
break;
case 256:
arg = 0x00ff0008; // 255 buffers of 2^8 bytes each
break;
default:
arg = 0x00ff0007; // 255 buffers of 2^7 bytes each
}
} else {
arg = 0x00ff000a; // 255 buffers of 2^10 bytes each
}
status = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &arg);
if (status == -1) {
string msg("SNDCTL_DSP_FRAGMENT ioctl failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
ui->cb_display_msg(TRANSLATE("Cannot set buffer size on sound card."),
MSG_CRITICAL);
close(fd);
return false;
}
// Channels
arg = channels;
status = ioctl(fd, SNDCTL_DSP_CHANNELS, &arg);
if (status == -1) {
string msg("SNDCTL_DSP_CHANNELS ioctl failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
msg = TRANSLATE("Sound card cannot be set to %1 channels.");
msg = replace_first(msg, "%1", int2str(channels));
ui->cb_display_msg(msg, MSG_CRITICAL);
return false;
}
if (arg != channels) {
log_file->write_report("Unable to set channels",
"t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
string msg = "Sound card cannot be set to ";
msg = TRANSLATE("Sound card cannot be set to %1 channels.");
msg = replace_first(msg, "%1", int2str(channels));
ui->cb_display_msg(msg, MSG_CRITICAL);
return false;
}
// Sample format
int fmt;
switch (format) {
case SAMPLEFORMAT_S16:
#ifdef WORDS_BIGENDIAN
fmt = AFMT_S16_BE;
#else
fmt = AFMT_S16_LE;
#endif
break;
case SAMPLEFORMAT_U16:
#ifdef WORDS_BIGENDIAN
fmt = AFMT_U16_BE;
#else
fmt = AFMT_U16_LE;
#endif
break;
case SAMPLEFORMAT_S8:
fmt = AFMT_S8;
break;
case SAMPLEFORMAT_U8:
fmt = AFMT_U8;
break;
default:
fmt = 0; // fail
}
arg = fmt;
status = ioctl(fd, SNDCTL_DSP_SETFMT, &arg);
if (status == -1) {
string msg("SNDCTL_DSP_SETFMT ioctl failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
ui->cb_display_msg(TRANSLATE("Cannot set sound card to 16 bits recording."),
MSG_CRITICAL);
return false;
}
arg = fmt;
status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
if (status == -1) {
string msg("SOUND_PCM_WRITE_BITS ioctl failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
ui->cb_display_msg(TRANSLATE("Cannot set sound card to 16 bits playing."),
MSG_CRITICAL);
return false;
}
// Sample rate
arg = sample_rate;
status = ioctl(fd, SNDCTL_DSP_SPEED, &arg);
if (status == -1) {
string msg("SNDCTL_DSP_SPEED ioctl failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::open",
LOG_NORMAL, LOG_CRITICAL);
msg = TRANSLATE("Cannot set sound card sample rate to %1");
msg = replace_first(msg, "%1", int2str(sample_rate));
ui->cb_display_msg(msg, MSG_CRITICAL);
return false;
}
play_buffersize = rec_buffersize = 0;
if (playback) play_buffersize = get_buffer_space(false);
if (capture) rec_buffersize = get_buffer_space(true);
return true;
}
void t_oss_io::enable(bool enable_playback, bool enable_recording) {
int arg, status;
arg = enable_recording ? PCM_ENABLE_INPUT : 0;
arg |= enable_playback ? PCM_ENABLE_OUTPUT : 0;
status = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &arg);
if (status == -1) {
string msg("SNDCTL_DSP_SETTRIGGER ioctl failed: ");
msg += get_error_str(errno);
log_file->write_report(msg, "t_oss_io::enable",
LOG_NORMAL, LOG_CRITICAL);
}
}
void t_oss_io::flush(bool playback_buffer, bool recording_buffer) {
for (int i = 0; i < 2; i++) {
// i == 0: flush playback buffer, 1: flush recording buffer
if (i == 0 && playback_buffer || i == 1 && recording_buffer) {
int skip_bytes = ( (i==0) ? play_buffersize :
rec_buffersize) - get_buffer_space(i == 1);
if(skip_bytes <= 0) continue;
unsigned char *trash = new unsigned char[skip_bytes];
MEMMAN_NEW_ARRAY(trash);
read(trash, skip_bytes);
MEMMAN_DELETE_ARRAY(trash);
delete [] trash;
}
}
}
int t_oss_io::get_buffer_space(bool is_recording_buffer)
{
audio_buf_info dsp_info;
int status = ioctl(fd, is_recording_buffer ? SNDCTL_DSP_GETISPACE :
SNDCTL_DSP_GETOSPACE, &dsp_info);
if (status == -1) return 0;
return dsp_info.bytes;
}
int t_oss_io::get_buffer_size(bool is_recording_buffer)
{
if (is_recording_buffer) return rec_buffersize;
else return play_buffersize;
}
bool t_oss_io::play_buffer_underrun(void) {
return get_buffer_space(false) >= get_buffer_size(false);
}
int t_oss_io::read(unsigned char* buf, int len) {
return ::read(fd, buf, len);
}
int t_oss_io::write(const unsigned char* buf, int len) {
return ::write(fd, buf, len);
}
#ifdef HAVE_LIBASOUND
t_alsa_io::t_alsa_io() : pcm_play_ptr(0), pcm_rec_ptr(0), play_framesize(1), rec_framesize(1),
play_buffersize(0), rec_buffersize(0) {
}
t_alsa_io::~t_alsa_io() {
if (pcm_play_ptr) {
log_file->write_header("t_alsa_io::~t_alsa_io", LOG_NORMAL, LOG_DEBUG);
log_file->write_raw("snd_pcm_close, handle = ");
log_file->write_raw(ptr2str(pcm_play_ptr));
log_file->write_endl();
log_file->write_footer();
// Without the snd_pcm_hw_free, snd_pcm_close sometimes fails.
snd_pcm_hw_free(pcm_play_ptr);
snd_pcm_close(pcm_play_ptr);
pcm_play_ptr = 0;
}
if (pcm_rec_ptr) {
log_file->write_header("t_alsa_io::~t_alsa_io", LOG_NORMAL, LOG_DEBUG);
log_file->write_raw("snd_pcm_close, handle = ");
log_file->write_raw(ptr2str(pcm_rec_ptr));
log_file->write_endl();
log_file->write_footer();
snd_pcm_hw_free(pcm_rec_ptr);
snd_pcm_close(pcm_rec_ptr);
pcm_rec_ptr = 0;
}
}
bool t_alsa_io::open(const string& device, bool playback, bool capture, bool blocking, int channels, t_audio_sampleformat format, int sample_rate, bool short_latency)
{
t_audio_io::open(device, playback, capture, blocking, channels, format, sample_rate,
short_latency);
int mode = 0;
string msg;
this->short_latency = short_latency;
const char* dev = device.c_str();
if(dev[0] == 0) dev = "default";
log_file->write_header("t_alsa_io::open", LOG_NORMAL);
log_file->write_raw("Opening ALSA device: ");
log_file->write_raw(dev);
log_file->write_endl();
if (playback) log_file->write_raw("play\n");
if (capture) log_file->write_raw("capture\n");
log_file->write_footer();
if (playback && capture) {
// Full duplex: Perform the operation in two steps
if (!open(device, true, false, blocking, channels, format,
sample_rate, short_latency))
return false;
if (!open(device, false, true, blocking, channels, format,
sample_rate, short_latency))
return false;
return true;
}
snd_pcm_t* pcm_ptr;
#define HANDLE_ALSA_ERROR(func) string msg(func); msg += " failed: "; msg += snd_strerror(err); \
log_file->write_report(msg, "t_alsa_io::open", LOG_NORMAL, LOG_CRITICAL); msg = TRANSLATE("Opening ALSA driver failed") + ": " + msg; \
ui->cb_display_msg(msg, MSG_CRITICAL); if(pcm_ptr) snd_pcm_close(pcm_ptr); return false;
if (!blocking) mode = SND_PCM_NONBLOCK;
int err = snd_pcm_open(&pcm_ptr, dev, playback ? SND_PCM_STREAM_PLAYBACK :
SND_PCM_STREAM_CAPTURE, mode);
if (err < 0) {
string msg("snd_pcm_open failed: ");
msg += snd_strerror(err);
log_file->write_report(msg, "t_alsa_io::open",
LOG_NORMAL, LOG_CRITICAL);
msg = "Cannot open ALSA driver for PCM ";
if (playback) {
msg = TRANSLATE("Cannot open ALSA driver for PCM playback");
} else {
msg = TRANSLATE("Cannot open ALSA driver for PCM capture");
}
msg += ": ";
msg += snd_strerror(err);
ui->cb_display_msg(msg, MSG_CRITICAL);
return false;
}
log_file->write_header("t_alsa_io::open", LOG_NORMAL, LOG_DEBUG);
log_file->write_raw("snd_pcm_open succeeded, handle = ");
log_file->write_raw(ptr2str(pcm_ptr));
log_file->write_endl();
log_file->write_footer();
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
// Set HW params
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
HANDLE_ALSA_ERROR("snd_pcm_hw_params_malloc");
}
MEMMAN_NEW(hw_params);
if ((err = snd_pcm_hw_params_any (pcm_ptr, hw_params)) < 0) {
HANDLE_ALSA_ERROR("snd_pcm_hw_params_any");
}
if ((err = snd_pcm_hw_params_set_access (pcm_ptr, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
HANDLE_ALSA_ERROR("snd_pcm_hw_params_set_access");
}
snd_pcm_format_t fmt;
int sample_bits;
switch (format) {
case SAMPLEFORMAT_S16:
#ifdef WORDS_BIGENDIAN
fmt = SND_PCM_FORMAT_S16_BE;
#else
fmt = SND_PCM_FORMAT_S16_LE;
#endif
sample_bits = 16;
break;
case SAMPLEFORMAT_U16:
#ifdef WORDS_BIGENDIAN
fmt = SND_PCM_FORMAT_U16_BE;
#else
fmt = SND_PCM_FORMAT_U16_LE;
#endif
sample_bits = 16;
break;
case SAMPLEFORMAT_S8:
fmt = SND_PCM_FORMAT_S8;
sample_bits = 8;
break;
case SAMPLEFORMAT_U8:
fmt = SND_PCM_FORMAT_U8;
sample_bits = 8;
break;
default:
{HANDLE_ALSA_ERROR("invalid sample format");}
}
if ((err = snd_pcm_hw_params_set_format (pcm_ptr, hw_params, fmt)) < 0) {
string s("snd_pcm_hw_params_set_format(");
s += int2str(fmt);
s += ")";
HANDLE_ALSA_ERROR(s);
}
// Some sound cards do not support the exact sample rate specified in the
// wav files. Still we first try to set the exact sample rate instead of
// specifying the 3rd parameter as -1 to choose an approximate.
// For some reason on my first sound card that supports the exact rate,
// I get mono sound when the -1 parameter is specified.
if ((err = snd_pcm_hw_params_set_rate (pcm_ptr, hw_params, sample_rate, 0)) < 0) {
msg = "Cannot set soundcard to exact sample rate of ";
msg += int2str(sample_rate);
msg += ".\nThe card will choose a lower approximate rate.";
log_file->write_report(msg, "t_alsa_io::open", LOG_NORMAL, LOG_WARNING);
if ((err = snd_pcm_hw_params_set_rate (pcm_ptr, hw_params, sample_rate, -1)) < 0) {
string s("snd_pcm_hw_params_set_rate(");
s += int2str(sample_rate);
s += ")";
HANDLE_ALSA_ERROR(s);
}
}
// Read back card rate for reporting in the log file.
unsigned int card_rate;
int card_dir;
snd_pcm_hw_params_get_rate(hw_params, &card_rate, &card_dir);
if ((err = snd_pcm_hw_params_set_channels (pcm_ptr, hw_params, channels)) < 0) {
string s("snd_pcm_hw_params_set_channels(");
s += int2str(channels);
s += ")";
HANDLE_ALSA_ERROR(s);
}
// Note: The buffersize is in FRAMES, not BYTES (one frame = sizeof(sample) * channels)
snd_pcm_uframes_t buffersize;
unsigned int periods = 8; // Double buffering
int dir = 1;
// Set the size of one period in samples
if (short_latency) {
if (playback) {
buffersize = sys_config->get_alsa_play_period_size();
} else {
buffersize = sys_config->get_alsa_capture_period_size();
}
} else {
buffersize = 1024;
}
if ((err = snd_pcm_hw_params_set_period_size_near (pcm_ptr, hw_params,
&buffersize, &dir)) < 0)
{
HANDLE_ALSA_ERROR("snd_pcm_hw_params_set_period_size_near");
}
// The number of periods determines the ALSA application buffer size.
// This size must be larger than the jitter buffer.
// TODO: use some more sophisticated algorithm here: read back the period
// size and calculate the number of periods needed (only in the
// short latency case)?
if (buffersize <= 64) {
periods *= 8;
} else if (buffersize <= 256) {
periods *= 4;
}
dir = 1;
if ((err = snd_pcm_hw_params_set_periods(pcm_ptr, hw_params, periods, dir)) < 0) {
if ((err = snd_pcm_hw_params_set_periods_near(pcm_ptr, hw_params,
&periods, &dir)) < 0)
{
HANDLE_ALSA_ERROR("snd_pcm_hw_params_set_periods");
}
}
if ((err = snd_pcm_hw_params (pcm_ptr, hw_params)) < 0) {
HANDLE_ALSA_ERROR("snd_pcm_hw_params");
}
// Find out if the sound card supports pause functionality
can_pause = (snd_pcm_hw_params_can_pause(hw_params) == 1);
MEMMAN_DELETE(hw_params);
snd_pcm_hw_params_free(hw_params);
// Set SW params
if ((err = snd_pcm_sw_params_malloc(&sw_params)) < 0) {
HANDLE_ALSA_ERROR("snd_pcm_sw_params_malloc");
}
MEMMAN_NEW(sw_params);
if ((err = snd_pcm_sw_params_current (pcm_ptr, sw_params)) < 0) {
HANDLE_ALSA_ERROR("snd_pcm_sw_params_current");
}
if ((err = snd_pcm_sw_params (pcm_ptr, sw_params)) < 0) {
HANDLE_ALSA_ERROR("snd_pcm_sw_params");
}
MEMMAN_DELETE(sw_params);
snd_pcm_sw_params_free(sw_params);
if ((err = snd_pcm_prepare (pcm_ptr)) < 0) {
HANDLE_ALSA_ERROR("snd_pcm_prepare");
}
if (playback) {
pcm_play_ptr = pcm_ptr;
play_framesize = (sample_bits / 8) * channels;
play_buffersize = (int)buffersize * periods * play_framesize;
play_periods = periods;
log_file->write_header("t_alsa_io::open", LOG_NORMAL, LOG_DEBUG);
log_file->write_raw("ALSA playback buffer settings.\n");
log_file->write_raw("Rate = ");
log_file->write_raw(card_rate);
log_file->write_raw(" frames/sec\n");
log_file->write_raw("Frame size = ");
log_file->write_raw(play_framesize);
log_file->write_raw(" bytes\n");
log_file->write_raw("Periods = ");
log_file->write_raw(play_periods);
log_file->write_endl();
log_file->write_raw("Period size = ");
log_file->write_raw(buffersize * play_framesize);
log_file->write_raw(" bytes\n");
log_file->write_raw("Buffer size = ");
log_file->write_raw(play_buffersize);
log_file->write_raw(" bytes\n");
log_file->write_raw("Can pause: ");
log_file->write_bool(can_pause);
log_file->write_endl();
log_file->write_footer();
} else {
// Since audio_rx checks buffer before reading, start manually
if ((err = snd_pcm_start(pcm_ptr)) < 0) {
HANDLE_ALSA_ERROR("snd_pcm_start");
}
pcm_rec_ptr = pcm_ptr;
rec_framesize = (sample_bits / 8) * channels;
rec_buffersize = (int)buffersize * periods * rec_framesize;
rec_periods = periods;
// HACK: snd_pcm_delay seems not to work for the dsnoop device.
// This code determines if snd_pcm is working. As the capture
// device just started, it should return zero or a small number.
// So if it returns that more than half of the buffer is filled
// already, it seems broken.
rec_delay_broken = false;
if (get_buffer_space(true) > rec_buffersize / 2) {
rec_delay_broken = true;
log_file->write_report(
"snd_pcm_delay does not work for capture buffer.",
"t_alsa_io::open", LOG_NORMAL, LOG_DEBUG);
}
log_file->write_header("t_alsa_io::open", LOG_NORMAL, LOG_DEBUG);
log_file->write_raw("ALSA capture buffer settings.\n");
log_file->write_raw("Rate = ");
log_file->write_raw(card_rate);
log_file->write_raw(" frames/sec\n");
log_file->write_raw("Frame size = ");
log_file->write_raw(rec_framesize);
log_file->write_raw(" bytes\n");
log_file->write_raw("Periods = ");
log_file->write_raw(rec_periods);
log_file->write_endl();
log_file->write_raw("Period size = ");
log_file->write_raw(buffersize * rec_framesize);
log_file->write_raw(" bytes\n");
log_file->write_raw("Buffer size = ");
log_file->write_raw(rec_buffersize);
log_file->write_raw(" bytes\n");
log_file->write_raw("Can pause: ");
log_file->write_bool(can_pause);
log_file->write_endl();
log_file->write_footer();
}
return true;
#undef HANDLE_ALSA_ERROR
}
void t_alsa_io::enable(bool enable_playback, bool enable_recording) {
if (!can_pause) return;
if (pcm_play_ptr) {
snd_pcm_pause(pcm_play_ptr, (int)enable_playback);
}
if (pcm_rec_ptr) {
snd_pcm_pause(pcm_rec_ptr, (int)enable_recording);
}
}
void t_alsa_io::flush(bool playback_buffer, bool recording_buffer) {
if (playback_buffer && pcm_play_ptr) {
// snd_pcm_reset(pcm_play_ptr);
snd_pcm_drop(pcm_play_ptr);
snd_pcm_prepare(pcm_play_ptr);
snd_pcm_start(pcm_play_ptr);
}
if (recording_buffer && pcm_rec_ptr) {
// For some obscure reason snd_pcm_reset causes the CPU
// load to rise to 99.9% when playing and capturing is
// done on the same sound card.
// Therefor snd_pcm_reset is replaced by functions to
// stop the card, drop samples and start again.
// snd_pcm_reset(pcm_rec_ptr);
snd_pcm_drop(pcm_rec_ptr);
snd_pcm_prepare(pcm_rec_ptr);
snd_pcm_start(pcm_rec_ptr);
}
}
int t_alsa_io::get_buffer_space(bool is_recording_buffer) {
int rv;
int err;
snd_pcm_sframes_t delay;
snd_pcm_status_t* status;
snd_pcm_status_alloca(&status);
if(is_recording_buffer) {
if(!pcm_rec_ptr) return 0;
// This does not work as snd_pcm_status_get_avail_max does not return
// accurate results.
// snd_pcm_status(pcm_rec_ptr, status);
// rv = rec_framesize * snd_pcm_status_get_avail_max(status);
snd_pcm_hwsync(pcm_rec_ptr);
if ((err = snd_pcm_delay(pcm_rec_ptr, &delay)) < 0) {
string msg = "snd_pcm_delay for capture buffer failed: ";
msg += snd_strerror(err);
log_file->write_report(msg, "t_alsa_io::get_buffer_space",
LOG_NORMAL, LOG_DEBUG);
delay = 0;
snd_pcm_prepare(pcm_rec_ptr);
}
if (rec_delay_broken) {
rv = 0; // there is no way to get an accurate number
} else {
rv = int(delay * rec_framesize);
}
if (rv > rec_buffersize) {
rv = rec_buffersize; // capture buffer overrun
snd_pcm_prepare(pcm_rec_ptr);
}
} else {
if(!pcm_play_ptr) return 0;
snd_pcm_status(pcm_play_ptr, status);
rv = play_framesize * snd_pcm_status_get_avail_max(status);
if (rv > play_buffersize) {
rv = play_buffersize; // playback buffer underrun
snd_pcm_prepare(pcm_play_ptr);
}
}
return rv;
}
int t_alsa_io::get_buffer_size(bool is_recording_buffer)
{
if (is_recording_buffer) return rec_buffersize;
else return play_buffersize;
}
bool t_alsa_io::play_buffer_underrun(void) {
if (!pcm_play_ptr) return false;
return snd_pcm_state(pcm_play_ptr) == SND_PCM_STATE_XRUN;
}
int t_alsa_io::read(unsigned char* buf, int len) {
string msg;
if (!pcm_rec_ptr) {
log_file->write_report("Illegal pcm_rec_prt.",
"t_alsa_io::read", LOG_NORMAL, LOG_CRITICAL);
return -1;
}
int len_frames = len / rec_framesize;
int read_frames = 0;
for(;;) {
int read = snd_pcm_readi(pcm_rec_ptr, buf, len_frames);
if (read == -EPIPE) {
msg = "Capture buffer overrun.";
log_file->write_report(msg, "t_alsa_io::read", LOG_NORMAL, LOG_DEBUG);
snd_pcm_prepare(pcm_rec_ptr);
snd_pcm_start(pcm_rec_ptr);
continue;
} else if (read <= 0) {
msg = "PCM read error: ";
msg += snd_strerror(read);
log_file->write_report(msg, "t_alsa_io::read", LOG_NORMAL, LOG_DEBUG);
return -1;
} else if (read < len_frames) {
buf += rec_framesize * read;
len_frames -= read;
read_frames += read;
continue;
}
return (read_frames + read) * rec_framesize;
}
}
int t_alsa_io::write(const unsigned char* buf, int len) {
int len_frames = len / play_framesize;
int frames_written = 0;
string msg;
for (;;) {
if(!pcm_play_ptr) return -1;
int written = snd_pcm_writei(pcm_play_ptr, buf, len_frames);
if (written == -EPIPE) {
msg = "Playback buffer underrun.";
log_file->write_report(msg, "t_alsa_io::write", LOG_NORMAL, LOG_DEBUG);
snd_pcm_prepare(pcm_play_ptr);
continue;
} else if (written == -EINVAL) {
msg = "Invalid argument passed to snd_pcm_writei: ";
msg += snd_strerror(written);
log_file->write_report(msg, "t_alsa_io::write", LOG_NORMAL, LOG_DEBUG);
} else if (written < 0) {
msg = "PCM write error: ";
msg += snd_strerror(written);
log_file->write_report(msg, "t_alsa_io::write", LOG_NORMAL, LOG_DEBUG);
return -1;
} else if (written < len_frames) {
buf += written * play_framesize;
len_frames -= written;
frames_written += written;
continue;
}
return (frames_written + written) * play_framesize;
}
}
// This function fills the specified list with ALSA hardware soundcards found on the system.
// It uses plughw:xx instead of hw:xx for specifiers, because hw:xx are not practical to
// use (e.g. they require a resampler/channel mixer in the application).
// playback indicates if a list with playback or capture devices should be created.
void alsa_fill_soundcards(list<t_audio_device>& l, bool playback)
{
int err = 0;
int card = -1, device = -1;
snd_ctl_t *handle;
snd_ctl_card_info_t *info;
snd_pcm_info_t *pcminfo;
snd_ctl_card_info_alloca(&info);
snd_pcm_info_alloca(&pcminfo);
for (;;) {
err = snd_card_next(&card);
if (err < 0 || card < 0) break;
if (card >= 0) {
string name = "hw:";
name += int2str(card);
if ((err = snd_ctl_open(&handle, name.c_str(), 0)) < 0) continue;
if ((err = snd_ctl_card_info(handle, info)) < 0) {
snd_ctl_close(handle);
continue;
}
const char *card_name = snd_ctl_card_info_get_name(info);
for (;;) {
err = snd_ctl_pcm_next_device(handle, &device);
if (err < 0 || device < 0) break;
snd_pcm_info_set_device(pcminfo, device);
snd_pcm_info_set_subdevice(pcminfo, 0);
if (playback) {
snd_pcm_info_set_stream(pcminfo, SND_PCM_STREAM_PLAYBACK);
} else {
snd_pcm_info_set_stream(pcminfo, SND_PCM_STREAM_CAPTURE);
}
if ((err = snd_ctl_pcm_info(handle, pcminfo)) < 0) continue;
t_audio_device dev;
dev.device = string("plughw:") + int2str(card) +
string(",") + int2str(device);
dev.name = string(card_name) + " (";
dev.name += snd_pcm_info_get_name(pcminfo);
dev.name += ")";
dev.type = t_audio_device::ALSA;
l.push_back(dev);
}
snd_ctl_close(handle);
}
}
}
// endif ifdef HAVE_LIBASOUND
#endif
|