aboutsummaryrefslogtreecommitdiffstats
path: root/src/output/alsa_output_plugin.c
blob: 7714e8a666fc4b5160078b5ab0e316cbc52fb92b (plain) (blame)
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
/*
 * Copyright (C) 2003-2011 The Music Player Daemon Project
 * http://www.musicpd.org
 *
 * 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, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#include "config.h"
#include "alsa_output_plugin.h"
#include "output_api.h"
#include "mixer_list.h"
#include "pcm_export.h"

#include <glib.h>
#include <alsa/asoundlib.h>

#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "alsa"

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API

static const char default_device[] = "default";

enum {
	MPD_ALSA_BUFFER_TIME_US = 500000,
};

#define MPD_ALSA_RETRY_NR 5

typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
					snd_pcm_uframes_t size);

struct alsa_data {
	struct audio_output base;

	struct pcm_export_state export;

	/** the configured name of the ALSA device; NULL for the
	    default device */
	char *device;

	/** use memory mapped I/O? */
	bool use_mmap;

	/** libasound's buffer_time setting (in microseconds) */
	unsigned int buffer_time;

	/** libasound's period_time setting (in microseconds) */
	unsigned int period_time;

	/** the mode flags passed to snd_pcm_open */
	int mode;

	/** the libasound PCM device handle */
	snd_pcm_t *pcm;

	/**
	 * a pointer to the libasound writei() function, which is
	 * snd_pcm_writei() or snd_pcm_mmap_writei(), depending on the
	 * use_mmap configuration
	 */
	alsa_writei_t *writei;

	/** the size of one audio frame */
	size_t frame_size;

	/**
	 * The size of one period, in number of frames.
	 */
	snd_pcm_uframes_t period_frames;

	/**
	 * The number of frames written in the current period.
	 */
	snd_pcm_uframes_t period_position;
};

/**
 * The quark used for GError.domain.
 */
static inline GQuark
alsa_output_quark(void)
{
	return g_quark_from_static_string("alsa_output");
}

static const char *
alsa_device(const struct alsa_data *ad)
{
	return ad->device != NULL ? ad->device : default_device;
}

static struct alsa_data *
alsa_data_new(void)
{
	struct alsa_data *ret = g_new(struct alsa_data, 1);

	ret->mode = 0;
	ret->writei = snd_pcm_writei;

	return ret;
}

static void
alsa_configure(struct alsa_data *ad, const struct config_param *param)
{
	ad->device = config_dup_block_string(param, "device", NULL);

	ad->use_mmap = config_get_block_bool(param, "use_mmap", false);

	ad->buffer_time = config_get_block_unsigned(param, "buffer_time",
			MPD_ALSA_BUFFER_TIME_US);
	ad->period_time = config_get_block_unsigned(param, "period_time", 0);

#ifdef SND_PCM_NO_AUTO_RESAMPLE
	if (!config_get_block_bool(param, "auto_resample", true))
		ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
#endif

#ifdef SND_PCM_NO_AUTO_CHANNELS
	if (!config_get_block_bool(param, "auto_channels", true))
		ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
#endif

#ifdef SND_PCM_NO_AUTO_FORMAT
	if (!config_get_block_bool(param, "auto_format", true))
		ad->mode |= SND_PCM_NO_AUTO_FORMAT;
#endif
}

static struct audio_output *
alsa_init(const struct config_param *param, GError **error_r)
{
	struct alsa_data *ad = alsa_data_new();

	if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
		g_free(ad);
		return NULL;
	}

	alsa_configure(ad, param);

	return &ad->base;
}

static void
alsa_finish(struct audio_output *ao)
{
	struct alsa_data *ad = (struct alsa_data *)ao;

	ao_base_finish(&ad->base);

	g_free(ad->device);
	g_free(ad);

	/* free libasound's config cache */
	snd_config_update_free_global();
}

static bool
alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
{
	struct alsa_data *ad = (struct alsa_data *)ao;

	pcm_export_init(&ad->export);
	return true;
}

static void
alsa_output_disable(struct audio_output *ao)
{
	struct alsa_data *ad = (struct alsa_data *)ao;

	pcm_export_deinit(&ad->export);
}

static bool
alsa_test_default_device(void)
{
	snd_pcm_t *handle;

	int ret = snd_pcm_open(&handle, default_device,
	                       SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
	if (ret) {
		g_message("Error opening default ALSA device: %s\n",
			  snd_strerror(-ret));
		return false;
	} else
		snd_pcm_close(handle);

	return true;
}

static snd_pcm_format_t
get_bitformat(enum sample_format sample_format)
{
	switch (sample_format) {
	case SAMPLE_FORMAT_UNDEFINED:
	case SAMPLE_FORMAT_DSD:
	case SAMPLE_FORMAT_DSD_OVER_USB:
		return SND_PCM_FORMAT_UNKNOWN;

	case SAMPLE_FORMAT_S8:
		return SND_PCM_FORMAT_S8;

	case SAMPLE_FORMAT_S16:
		return SND_PCM_FORMAT_S16;

	case SAMPLE_FORMAT_S24_P32:
		return SND_PCM_FORMAT_S24;

	case SAMPLE_FORMAT_S24:
		return G_BYTE_ORDER == G_BIG_ENDIAN
			? SND_PCM_FORMAT_S24_3BE
			: SND_PCM_FORMAT_S24_3LE;

	case SAMPLE_FORMAT_S32:
		return SND_PCM_FORMAT_S32;

	case SAMPLE_FORMAT_FLOAT:
		return SND_PCM_FORMAT_FLOAT;
	}

	assert(false);
	return SND_PCM_FORMAT_UNKNOWN;
}

static snd_pcm_format_t
byteswap_bitformat(snd_pcm_format_t fmt)
{
	switch(fmt) {
	case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
	case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
	case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
	case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
	case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;

	case SND_PCM_FORMAT_S24_3BE:
		return SND_PCM_FORMAT_S24_3LE;

	case SND_PCM_FORMAT_S24_3LE:
		return SND_PCM_FORMAT_S24_3BE;

	case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
	default: return SND_PCM_FORMAT_UNKNOWN;
	}
}

/**
 * Attempts to configure the specified sample format.
 */
static int
alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
		       struct audio_format *audio_format,
		       enum sample_format sample_format)
{
	snd_pcm_format_t alsa_format = get_bitformat(sample_format);
	if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
		return -EINVAL;

	int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format);
	if (err == 0)
		audio_format->format = sample_format;

	return err;
}

/**
 * Attempts to configure the specified sample format with reversed
 * host byte order.
 */
static int
alsa_output_try_reverse(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
		       struct audio_format *audio_format,
		       enum sample_format sample_format)
{
	snd_pcm_format_t alsa_format =
		byteswap_bitformat(get_bitformat(sample_format));
	if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
		return -EINVAL;

	int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format);
	if (err == 0)
		audio_format->format = sample_format;

	return err;
}

/**
 * Attempts to configure the specified sample format, and tries the
 * reversed host byte order if was not supported.
 */
static int
alsa_output_try_format_both(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
			    struct audio_format *audio_format,
			    bool *reverse_endian_r,
			    enum sample_format sample_format)
{
	*reverse_endian_r = false;

	int err = alsa_output_try_format(pcm, hwparams, audio_format,
					 sample_format);
	if (err == -EINVAL) {
		*reverse_endian_r = true;
		err = alsa_output_try_reverse(pcm, hwparams, audio_format,
					      sample_format);
	}

	return err;
}

/**
 * Configure a sample format, and probe other formats if that fails.
 */
static int
alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
			 struct audio_format *audio_format,
			 bool *reverse_endian_r)
{
	/* try the input format first */

	int err = alsa_output_try_format_both(pcm, hwparams, audio_format,
					      reverse_endian_r,
					      audio_format->format);
	if (err != -EINVAL)
		return err;

	/* if unsupported by the hardware, try other formats */

	static const enum sample_format probe_formats[] = {
		SAMPLE_FORMAT_S24_P32,
		SAMPLE_FORMAT_S32,
		SAMPLE_FORMAT_S24,
		SAMPLE_FORMAT_S16,
		SAMPLE_FORMAT_S8,
		SAMPLE_FORMAT_UNDEFINED,
	};

	for (unsigned i = 0; probe_formats[i] != SAMPLE_FORMAT_UNDEFINED; ++i) {
		if (probe_formats[i] == audio_format->format)
			continue;

		err = alsa_output_try_format_both(pcm, hwparams, audio_format,
						  reverse_endian_r,
						  probe_formats[i]);
		if (err != -EINVAL)
			return err;
	}

	return -EINVAL;
}

/**
 * Set up the snd_pcm_t object which was opened by the caller.  Set up
 * the configured settings and the audio format.
 */
static bool
alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
	   GError **error)
{
	snd_pcm_hw_params_t *hwparams;
	snd_pcm_sw_params_t *swparams;
	unsigned int sample_rate = audio_format->sample_rate;
	unsigned int channels = audio_format->channels;
	snd_pcm_uframes_t alsa_buffer_size;
	snd_pcm_uframes_t alsa_period_size;
	int err;
	const char *cmd = NULL;
	int retry = MPD_ALSA_RETRY_NR;
	unsigned int period_time, period_time_ro;
	unsigned int buffer_time;

	period_time_ro = period_time = ad->period_time;
configure_hw:
	/* configure HW params */
	snd_pcm_hw_params_alloca(&hwparams);
	cmd = "snd_pcm_hw_params_any";
	err = snd_pcm_hw_params_any(ad->pcm, hwparams);
	if (err < 0)
		goto error;

	if (ad->use_mmap) {
		err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
						   SND_PCM_ACCESS_MMAP_INTERLEAVED);
		if (err < 0) {
			g_warning("Cannot set mmap'ed mode on ALSA device \"%s\":  %s\n",
				  alsa_device(ad), snd_strerror(-err));
			g_warning("Falling back to direct write mode\n");
			ad->use_mmap = false;
		} else
			ad->writei = snd_pcm_mmap_writei;
	}

	if (!ad->use_mmap) {
		cmd = "snd_pcm_hw_params_set_access";
		err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
						   SND_PCM_ACCESS_RW_INTERLEAVED);
		if (err < 0)
			goto error;
		ad->writei = snd_pcm_writei;
	}

	bool reverse_endian;
	err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
				       &reverse_endian);
	if (err < 0) {
		g_set_error(error, alsa_output_quark(), err,
			    "ALSA device \"%s\" does not support format %s: %s",
			    alsa_device(ad),
			    sample_format_to_string(audio_format->format),
			    snd_strerror(-err));
		return false;
	}

	err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
						  &channels);
	if (err < 0) {
		g_set_error(error, alsa_output_quark(), err,
			    "ALSA device \"%s\" does not support %i channels: %s",
			    alsa_device(ad), (int)audio_format->channels,
			    snd_strerror(-err));
		return false;
	}
	audio_format->channels = (int8_t)channels;

	err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
					      &sample_rate, NULL);
	if (err < 0 || sample_rate == 0) {
		g_set_error(error, alsa_output_quark(), err,
			    "ALSA device \"%s\" does not support %u Hz audio",
			    alsa_device(ad), audio_format->sample_rate);
		return false;
	}
	audio_format->sample_rate = sample_rate;

	snd_pcm_uframes_t buffer_size_min, buffer_size_max;
	snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min);
	snd_pcm_hw_params_get_buffer_size_max(hwparams, &buffer_size_max);
	unsigned buffer_time_min, buffer_time_max;
	snd_pcm_hw_params_get_buffer_time_min(hwparams, &buffer_time_min, 0);
	snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time_max, 0);
	g_debug("buffer: size=%u..%u time=%u..%u",
		(unsigned)buffer_size_min, (unsigned)buffer_size_max,
		buffer_time_min, buffer_time_max);

	snd_pcm_uframes_t period_size_min, period_size_max;
	snd_pcm_hw_params_get_period_size_min(hwparams, &period_size_min, 0);
	snd_pcm_hw_params_get_period_size_max(hwparams, &period_size_max, 0);
	unsigned period_time_min, period_time_max;
	snd_pcm_hw_params_get_period_time_min(hwparams, &period_time_min, 0);
	snd_pcm_hw_params_get_period_time_max(hwparams, &period_time_max, 0);
	g_debug("period: size=%u..%u time=%u..%u",
		(unsigned)period_size_min, (unsigned)period_size_max,
		period_time_min, period_time_max);

	if (ad->buffer_time > 0) {
		buffer_time = ad->buffer_time;
		cmd = "snd_pcm_hw_params_set_buffer_time_near";
		err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm, hwparams,
							     &buffer_time, NULL);
		if (err < 0)
			goto error;
	} else {
		err = snd_pcm_hw_params_get_buffer_time(hwparams, &buffer_time,
							NULL);
		if (err < 0)
			buffer_time = 0;
	}

	if (period_time_ro == 0 && buffer_time >= 10000) {
		period_time_ro = period_time = buffer_time / 4;

		g_debug("default period_time = buffer_time/4 = %u/4 = %u",
			buffer_time, period_time);
	}

	if (period_time_ro > 0) {
		period_time = period_time_ro;
		cmd = "snd_pcm_hw_params_set_period_time_near";
		err = snd_pcm_hw_params_set_period_time_near(ad->pcm, hwparams,
							     &period_time, NULL);
		if (err < 0)
			goto error;
	}

	cmd = "snd_pcm_hw_params";
	err = snd_pcm_hw_params(ad->pcm, hwparams);
	if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
		period_time_ro = period_time_ro >> 1;
		goto configure_hw;
	} else if (err < 0)
		goto error;
	if (retry != MPD_ALSA_RETRY_NR)
		g_debug("ALSA period_time set to %d\n", period_time);

	cmd = "snd_pcm_hw_params_get_buffer_size";
	err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
	if (err < 0)
		goto error;

	cmd = "snd_pcm_hw_params_get_period_size";
	err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
						NULL);
	if (err < 0)
		goto error;

	/* configure SW params */
	snd_pcm_sw_params_alloca(&swparams);

	cmd = "snd_pcm_sw_params_current";
	err = snd_pcm_sw_params_current(ad->pcm, swparams);
	if (err < 0)
		goto error;

	cmd = "snd_pcm_sw_params_set_start_threshold";
	err = snd_pcm_sw_params_set_start_threshold(ad->pcm, swparams,
						    alsa_buffer_size -
						    alsa_period_size);
	if (err < 0)
		goto error;

	cmd = "snd_pcm_sw_params_set_avail_min";
	err = snd_pcm_sw_params_set_avail_min(ad->pcm, swparams,
					      alsa_period_size);
	if (err < 0)
		goto error;

	cmd = "snd_pcm_sw_params";
	err = snd_pcm_sw_params(ad->pcm, swparams);
	if (err < 0)
		goto error;

	g_debug("buffer_size=%u period_size=%u",
		(unsigned)alsa_buffer_size, (unsigned)alsa_period_size);

	if (alsa_period_size == 0)
		/* this works around a SIGFPE bug that occurred when
		   an ALSA driver indicated period_size==0; this
		   caused a division by zero in alsa_play().  By using
		   the fallback "1", we make sure that this won't
		   happen again. */
		alsa_period_size = 1;

	ad->period_frames = alsa_period_size;
	ad->period_position = 0;

	pcm_export_open(&ad->export, audio_format->format,
			reverse_endian);

	return true;

error:
	g_set_error(error, alsa_output_quark(), err,
		    "Error opening ALSA device \"%s\" (%s): %s",
		    alsa_device(ad), cmd, snd_strerror(-err));
	return false;
}

static bool
alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
{
	struct alsa_data *ad = (struct alsa_data *)ao;
	int err;
	bool success;

	err = snd_pcm_open(&ad->pcm, alsa_device(ad),
			   SND_PCM_STREAM_PLAYBACK, ad->mode);
	if (err < 0) {
		g_set_error(error, alsa_output_quark(), err,
			    "Failed to open ALSA device \"%s\": %s",
			    alsa_device(ad), snd_strerror(err));
		return false;
	}

	success = alsa_setup(ad, audio_format, error);
	if (!success) {
		snd_pcm_close(ad->pcm);
		return false;
	}

	ad->frame_size = audio_format_frame_size(audio_format);

	return true;
}

static int
alsa_recover(struct alsa_data *ad, int err)
{
	if (err == -EPIPE) {
		g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
	} else if (err == -ESTRPIPE) {
		g_debug("ALSA device \"%s\" was suspended\n", alsa_device(ad));
	}

	switch (snd_pcm_state(ad->pcm)) {
	case SND_PCM_STATE_PAUSED:
		err = snd_pcm_pause(ad->pcm, /* disable */ 0);
		break;
	case SND_PCM_STATE_SUSPENDED:
		err = snd_pcm_resume(ad->pcm);
		if (err == -EAGAIN)
			return 0;
		/* fall-through to snd_pcm_prepare: */
	case SND_PCM_STATE_SETUP:
	case SND_PCM_STATE_XRUN:
		ad->period_position = 0;
		err = snd_pcm_prepare(ad->pcm);
		break;
	case SND_PCM_STATE_DISCONNECTED:
		break;
	/* this is no error, so just keep running */
	case SND_PCM_STATE_RUNNING:
		err = 0;
		break;
	default:
		/* unknown state, do nothing */
		break;
	}

	return err;
}

static void
alsa_drain(struct audio_output *ao)
{
	struct alsa_data *ad = (struct alsa_data *)ao;

	if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
		return;

	if (ad->period_position > 0) {
		/* generate some silence to finish the partial
		   period */
		snd_pcm_uframes_t nframes =
			ad->period_frames - ad->period_position;
		size_t nbytes = nframes * ad->frame_size;
		void *buffer = g_malloc(nbytes);
		snd_pcm_hw_params_t *params;
		snd_pcm_format_t format;
		unsigned channels;

		snd_pcm_hw_params_alloca(&params);
		snd_pcm_hw_params_current(ad->pcm, params);
		snd_pcm_hw_params_get_format(params, &format);
		snd_pcm_hw_params_get_channels(params, &channels);

		snd_pcm_format_set_silence(format, buffer, nframes * channels);
		ad->writei(ad->pcm, buffer, nframes);
		g_free(buffer);
	}

	snd_pcm_drain(ad->pcm);

	ad->period_position = 0;
}

static void
alsa_cancel(struct audio_output *ao)
{
	struct alsa_data *ad = (struct alsa_data *)ao;

	ad->period_position = 0;

	snd_pcm_drop(ad->pcm);
}

static void
alsa_close(struct audio_output *ao)
{
	struct alsa_data *ad = (struct alsa_data *)ao;

	snd_pcm_close(ad->pcm);
}

static size_t
alsa_play(struct audio_output *ao, const void *chunk, size_t size,
	  GError **error)
{
	struct alsa_data *ad = (struct alsa_data *)ao;

	chunk = pcm_export(&ad->export, chunk, size, &size);

	size /= ad->frame_size;

	while (true) {
		snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
		if (ret > 0) {
			ad->period_position = (ad->period_position + ret)
				% ad->period_frames;
			return ret * ad->frame_size;
		}

		if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
		    alsa_recover(ad, ret) < 0) {
			g_set_error(error, alsa_output_quark(), errno,
				    "%s", snd_strerror(-errno));
			return 0;
		}
	}
}

const struct audio_output_plugin alsa_output_plugin = {
	.name = "alsa",
	.test_default_device = alsa_test_default_device,
	.init = alsa_init,
	.finish = alsa_finish,
	.enable = alsa_output_enable,
	.disable = alsa_output_disable,
	.open = alsa_open,
	.play = alsa_play,
	.drain = alsa_drain,
	.cancel = alsa_cancel,
	.close = alsa_close,

	.mixer_plugin = &alsa_mixer_plugin,
};