summaryrefslogtreecommitdiffstats
path: root/audio/out/ao_alsa.c
blob: cd46e5280638d094e857ca437a2afa067f8837f1 (plain)
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
/*
 * ALSA 0.9.x-1.x audio output driver
 *
 * Copyright (C) 2004 Alex Beregszaszi
 * Zsolt Barat <joy@streamminister.de>
 *
 * modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
 * additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
 * 08/22/2002 iec958-init rewritten and merged with common init, zsolt
 * 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
 * 04/25/2004 printfs converted to mp_msg, Zsolt.
 *
 * This file is part of MPlayer.
 *
 * MPlayer 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.
 *
 * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#include <errno.h>
#include <sys/time.h>
#include <stdlib.h>
#include <stdarg.h>
#include <math.h>
#include <string.h>

#include "config.h"
#include "options/options.h"
#include "options/m_option.h"
#include "common/msg.h"

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API

#include <alsa/asoundlib.h>

#include "ao.h"
#include "internal.h"
#include "audio/format.h"

struct priv {
    snd_pcm_t *alsa;
    snd_pcm_format_t alsa_fmt;
    int can_pause;
    snd_pcm_sframes_t prepause_frames;
    float delay_before_pause;
    int buffersize; // in frames
    int outburst; // in frames

    int cfg_block;
    char *cfg_device;
    char *cfg_mixer_device;
    char *cfg_mixer_name;
    int cfg_mixer_index;
    int cfg_resample;
};

#define BUFFER_TIME 250000  // 250ms
#define FRAGCOUNT 16

#define CHECK_ALSA_ERROR(message) \
    do { \
        if (err < 0) { \
            MP_ERR(ao, "%s: %s\n", (message), snd_strerror(err)); \
            goto alsa_error; \
        } \
    } while (0)

static float get_delay(struct ao *ao);
static void uninit(struct ao *ao);

/* to set/get/query special features/parameters */
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
    struct priv *p = ao->priv;
    snd_mixer_t *handle = NULL;
    switch (cmd) {
    case AOCONTROL_GET_MUTE:
    case AOCONTROL_SET_MUTE:
    case AOCONTROL_GET_VOLUME:
    case AOCONTROL_SET_VOLUME:
    {
        int err;
        snd_mixer_elem_t *elem;
        snd_mixer_selem_id_t *sid;

        long pmin, pmax;
        long get_vol, set_vol;
        float f_multi;

        if (AF_FORMAT_IS_IEC61937(ao->format))
            return CONTROL_FALSE;

        //allocate simple id
        snd_mixer_selem_id_alloca(&sid);

        //sets simple-mixer index and name
        snd_mixer_selem_id_set_index(sid, p->cfg_mixer_index);
        snd_mixer_selem_id_set_name(sid, p->cfg_mixer_name);

        err = snd_mixer_open(&handle, 0);
        CHECK_ALSA_ERROR("Mixer open error");

        err = snd_mixer_attach(handle, p->cfg_mixer_device);
        CHECK_ALSA_ERROR("Mixer attach error");

        err = snd_mixer_selem_register(handle, NULL, NULL);
        CHECK_ALSA_ERROR("Mixer register error");

        err = snd_mixer_load(handle);
        CHECK_ALSA_ERROR("Mixer load error");

        elem = snd_mixer_find_selem(handle, sid);
        if (!elem) {
            MP_VERBOSE(ao, "Unable to find simple control '%s',%i.\n",
                       snd_mixer_selem_id_get_name(sid),
                       snd_mixer_selem_id_get_index(sid));
            goto alsa_error;
        }

        snd_mixer_selem_get_playback_volume_range(elem, &pmin, &pmax);
        f_multi = (100 / (float)(pmax - pmin));

        switch (cmd) {
        case AOCONTROL_SET_VOLUME: {
            ao_control_vol_t *vol = arg;
            set_vol = vol->left / f_multi + pmin + 0.5;

            //setting channels
            err = snd_mixer_selem_set_playback_volume
                    (elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol);
            CHECK_ALSA_ERROR("Error setting left channel");
            MP_DBG(ao, "left=%li, ", set_vol);

            set_vol = vol->right / f_multi + pmin + 0.5;

            err = snd_mixer_selem_set_playback_volume
                    (elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol);
            CHECK_ALSA_ERROR("Error setting right channel");
            MP_DBG(ao, "right=%li, pmin=%li, pmax=%li, mult=%f\n",
                   set_vol, pmin, pmax,
                   f_multi);
            break;
        }
        case AOCONTROL_GET_VOLUME: {
            ao_control_vol_t *vol = arg;
            snd_mixer_selem_get_playback_volume
                (elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
            vol->left = (get_vol - pmin) * f_multi;
            snd_mixer_selem_get_playback_volume
                (elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
            vol->right = (get_vol - pmin) * f_multi;
            MP_DBG(ao, "left=%f, right=%f\n", vol->left, vol->right);
            break;
        }
        case AOCONTROL_SET_MUTE: {
            bool *mute = arg;
            if (!snd_mixer_selem_has_playback_switch(elem))
                goto alsa_error;
            if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
                snd_mixer_selem_set_playback_switch
                    (elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
            }
            snd_mixer_selem_set_playback_switch
                (elem, SND_MIXER_SCHN_FRONT_LEFT, !*mute);
            break;
        }
        case AOCONTROL_GET_MUTE: {
            bool *mute = arg;
            if (!snd_mixer_selem_has_playback_switch(elem))
                goto alsa_error;
            int tmp = 1;
            snd_mixer_selem_get_playback_switch
                (elem, SND_MIXER_SCHN_FRONT_LEFT, &tmp);
            *mute = !tmp;
            if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
                snd_mixer_selem_get_playback_switch
                    (elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
                *mute &= !tmp;
            }
            break;
        }
        }
        snd_mixer_close(handle);
        return CONTROL_OK;
    }

    } //end switch
    return CONTROL_UNKNOWN;

alsa_error:
    if (handle)
        snd_mixer_close(handle);
    return CONTROL_ERROR;
}

static const int mp_to_alsa_format[][2] = {
    {AF_FORMAT_S8,          SND_PCM_FORMAT_S8},
    {AF_FORMAT_U8,          SND_PCM_FORMAT_U8},
    {AF_FORMAT_U16_LE,      SND_PCM_FORMAT_U16_LE},
    {AF_FORMAT_U16_BE,      SND_PCM_FORMAT_U16_BE},
    {AF_FORMAT_S16_LE,      SND_PCM_FORMAT_S16_LE},
    {AF_FORMAT_S16_BE,      SND_PCM_FORMAT_S16_BE},
    {AF_FORMAT_U32_LE,      SND_PCM_FORMAT_U32_LE},
    {AF_FORMAT_U32_BE,      SND_PCM_FORMAT_U32_BE},
    {AF_FORMAT_S32_LE,      SND_PCM_FORMAT_S32_LE},
    {AF_FORMAT_S32_BE,      SND_PCM_FORMAT_S32_BE},
    {AF_FORMAT_U24_LE,      SND_PCM_FORMAT_U24_3LE},
    {AF_FORMAT_U24_BE,      SND_PCM_FORMAT_U24_3BE},
    {AF_FORMAT_S24_LE,      SND_PCM_FORMAT_S24_3LE},
    {AF_FORMAT_S24_BE,      SND_PCM_FORMAT_S24_3BE},
    {AF_FORMAT_FLOAT_LE,    SND_PCM_FORMAT_FLOAT_LE},
    {AF_FORMAT_FLOAT_BE,    SND_PCM_FORMAT_FLOAT_BE},
    {AF_FORMAT_AC3_LE,      SND_PCM_FORMAT_S16_LE},
    {AF_FORMAT_AC3_BE,      SND_PCM_FORMAT_S16_BE},
    {AF_FORMAT_IEC61937_LE, SND_PCM_FORMAT_S16_LE},
    {AF_FORMAT_IEC61937_BE, SND_PCM_FORMAT_S16_BE},
    {AF_FORMAT_MPEG2,       SND_PCM_FORMAT_MPEG},
    {AF_FORMAT_UNKNOWN,     SND_PCM_FORMAT_UNKNOWN},
};

static int find_alsa_format(int af_format)
{
    af_format = af_fmt_from_planar(af_format);
    for (int n = 0; mp_to_alsa_format[n][0] != AF_FORMAT_UNKNOWN; n++) {
        if (mp_to_alsa_format[n][0] == af_format)
            return mp_to_alsa_format[n][1];
    }
    return SND_PCM_FORMAT_UNKNOWN;
}

// Lists device names and their implied channel map.
// The second item must be resolvable with mp_chmap_from_str().
// Source: http://www.alsa-project.org/main/index.php/DeviceNames
// (Speaker names are slightly different from mpv's.)
static const char *const device_channel_layouts[][2] = {
    {"default",         "fc"},
    {"default",         "fl-fr"},
    {"rear",            "bl-br"},
    {"center_lfe",      "fc-lfe"},
    {"side",            "sl-sr"},
    {"surround40",      "fl-fr-bl-br"},
    {"surround50",      "fl-fr-bl-br-fc"},
    {"surround41",      "fl-fr-bl-br-lfe"},
    {"surround51",      "fl-fr-bl-br-fc-lfe"},
    {"surround71",      "fl-fr-bl-br-fc-lfe-sl-sr"},
};

#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))

#define NUM_ALSA_CHMAPS ARRAY_LEN(device_channel_layouts)

static const char *select_chmap(struct ao *ao)
{
    struct mp_chmap_sel sel = {0};
    struct mp_chmap maps[NUM_ALSA_CHMAPS];
    for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
        mp_chmap_from_str(&maps[n], bstr0(device_channel_layouts[n][1]));
        mp_chmap_sel_add_map(&sel, &maps[n]);
    };

    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
        return NULL;

    for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
        if (mp_chmap_equals(&ao->channels, &maps[n]))
            return device_channel_layouts[n][0];
    }

    char *name = mp_chmap_to_str(&ao->channels);
    MP_ERR(ao, "channel layout %s (%d ch) not supported.\n",
           name, ao->channels.num);
    talloc_free(name);
    return "default";
}

static int map_iec958_srate(int srate)
{
    switch (srate) {
    case 44100:     return IEC958_AES3_CON_FS_44100;
    case 48000:     return IEC958_AES3_CON_FS_48000;
    case 32000:     return IEC958_AES3_CON_FS_32000;
    case 22050:     return IEC958_AES3_CON_FS_22050;
    case 24000:     return IEC958_AES3_CON_FS_24000;
    case 88200:     return IEC958_AES3_CON_FS_88200;
    case 768000:    return IEC958_AES3_CON_FS_768000;
    case 96000:     return IEC958_AES3_CON_FS_96000;
    case 176400:    return IEC958_AES3_CON_FS_176400;
    case 192000:    return IEC958_AES3_CON_FS_192000;
    default:        return IEC958_AES3_CON_FS_NOTID;
    }
}

static int try_open_device(struct ao *ao, const char *device, int open_mode)
{
    struct priv *p = ao->priv;

    if (AF_FORMAT_IS_IEC61937(ao->format)) {
        void *tmp = talloc_new(NULL);
        /* to set the non-audio bit, use AES0=6 */
        char *params = talloc_asprintf(tmp,
                        "AES0=%d,AES1=%d,AES2=0,AES3=%d",
                        IEC958_AES0_NONAUDIO | IEC958_AES0_PRO_EMPHASIS_NONE,
                        IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
                        map_iec958_srate(ao->samplerate));
        const char *ac3_device = device;
        int len = strlen(device);
        char *end = strchr(device, ':');
        if (!end) {
            /* no existing parameters: add it behind device name */
            ac3_device = talloc_asprintf(tmp, "%s:%s", device, params);
        } else if (end[1] == '\0') {
            /* ":" but no parameters */
            ac3_device = talloc_asprintf(tmp, "%s%s", device, params);
        } else if (end[1] == '{' && device[len - 1] == '}') {
            /* parameters in config syntax: add it inside the { } block */
            ac3_device = talloc_asprintf(tmp, "%.*s %s}", len - 1, device, params);
        } else {
            /* a simple list of parameters: add it at the end of the list */
            ac3_device = talloc_asprintf(tmp, "%s,%s", device, params);
        }
        int err = snd_pcm_open
                    (&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, open_mode);
        talloc_free(tmp);
        if (!err)
            return 0;
    }

    return snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, open_mode);
}

/*
    open & setup audio device
    return: 0=success -1=fail
 */
static int init(struct ao *ao)
{
    int err;
    snd_pcm_uframes_t chunk_size;
    snd_pcm_uframes_t bufsize;
    snd_pcm_uframes_t boundary;

    struct priv *p = ao->priv;

    /* switch for spdif
     * sets opening sequence for SPDIF
     * sets also the playback and other switches 'on the fly'
     * while opening the abstract alias for the spdif subdevice
     * 'iec958'
     */
    const char *device;
    if (AF_FORMAT_IS_IEC61937(ao->format)) {
        device = "iec958";
        MP_VERBOSE(ao, "playing AC3/iec61937/iec958, %i channels\n",
                   ao->channels.num);
    } else {
        device = select_chmap(ao);
        if (strcmp(device, "default") != 0 && (ao->format & AF_FORMAT_F)) {
            // hack - use the converter plugin (why the heck?)
            device = talloc_asprintf(ao, "plug:%s", device);
        }
    }
    if (p->cfg_device && p->cfg_device[0])
        device = p->cfg_device;

    MP_VERBOSE(ao, "using device: %s\n", device);
    MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());

    int open_mode = p->cfg_block ? 0 : SND_PCM_NONBLOCK;
    //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
    err = try_open_device(ao, device, open_mode);
    if (err < 0) {
        if (err != -EBUSY && !p->cfg_block) {
            MP_WARN(ao, "Open in nonblock-mode "
                    "failed, trying to open in block-mode.\n");
            err = try_open_device(ao, device, 0);
        }
        CHECK_ALSA_ERROR("Playback open error");
    }

    err = snd_pcm_nonblock(p->alsa, 0);
    if (err < 0) {
        MP_ERR(ao, "Error setting block-mode: %s.\n", snd_strerror(err));
    } else {
        MP_VERBOSE(ao, "pcm opened in blocking mode\n");
    }

    snd_pcm_hw_params_t *alsa_hwparams;
    snd_pcm_sw_params_t *alsa_swparams;

    snd_pcm_hw_params_alloca(&alsa_hwparams);
    snd_pcm_sw_params_alloca(&alsa_swparams);

    // setting hw-parameters
    err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
    CHECK_ALSA_ERROR("Unable to get initial parameters");

    p->alsa_fmt = find_alsa_format(ao->format);
    if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) {
        p->alsa_fmt = SND_PCM_FORMAT_S16;
        ao->format = AF_FORMAT_S16;
    }

    err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
    if (err < 0) {
        MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n",
                af_fmt_to_str(ao->format));
        p->alsa_fmt = SND_PCM_FORMAT_S16_LE;
        if (AF_FORMAT_IS_AC3(ao->format))
            ao->format = AF_FORMAT_AC3_LE;
        else if (AF_FORMAT_IS_IEC61937(ao->format))
            ao->format = AF_FORMAT_IEC61937_LE;
        else
            ao->format = AF_FORMAT_S16_LE;
    }

    err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
    CHECK_ALSA_ERROR("Unable to set format");

    snd_pcm_access_t access = af_fmt_is_planar(ao->format)
                                    ? SND_PCM_ACCESS_RW_NONINTERLEAVED
                                    : SND_PCM_ACCESS_RW_INTERLEAVED;
    err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
    if (err < 0 && af_fmt_is_planar(ao->format)) {
        ao->format = af_fmt_from_planar(ao->format);
        access = SND_PCM_ACCESS_RW_INTERLEAVED;
        err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
    }
    CHECK_ALSA_ERROR("Unable to set access type");

    int num_channels = ao->channels.num;
    err = snd_pcm_hw_params_set_channels_near
            (p->alsa, alsa_hwparams, &num_channels);
    CHECK_ALSA_ERROR("Unable to set channels");

    if (num_channels != ao->channels.num) {
        MP_ERR(ao, "Couldn't get requested number of channels.\n");
        mp_chmap_from_channels_alsa(&ao->channels, num_channels);
    }

    // Some ALSA drivers have broken delay reporting, so disable the ALSA
    // resampling plugin by default.
    if (!p->cfg_resample) {
        err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
        CHECK_ALSA_ERROR("Unable to disable resampling");
    }

    err = snd_pcm_hw_params_set_rate_near
            (p->alsa, alsa_hwparams, &ao->samplerate, NULL);
    CHECK_ALSA_ERROR("Unable to set samplerate-2");

    err = snd_pcm_hw_params_set_buffer_time_near
            (p->alsa, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL);
    CHECK_ALSA_ERROR("Unable to set buffer time near");

    err = snd_pcm_hw_params_set_periods_near
            (p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
    CHECK_ALSA_ERROR("Unable to set periods");

    /* finally install hardware parameters */
    err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
    CHECK_ALSA_ERROR("Unable to set hw-parameters");

    // end setting hw-params

    // gets buffersize for control
    err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
    CHECK_ALSA_ERROR("Unable to get buffersize");

    p->buffersize = bufsize;
    MP_VERBOSE(ao, "got buffersize=%i samples\n", p->buffersize);

    err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL);
    CHECK_ALSA_ERROR("Unable to get period size");

    MP_VERBOSE(ao, "got period size %li\n", chunk_size);
    p->outburst = chunk_size;

    /* setting software parameters */
    err = snd_pcm_sw_params_current(p->alsa, alsa_swparams);
    CHECK_ALSA_ERROR("Unable to get sw-parameters");

    err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
    CHECK_ALSA_ERROR("Unable to get boundary");

    /* start playing when one period has been written */
    err = snd_pcm_sw_params_set_start_threshold
            (p->alsa, alsa_swparams, chunk_size);
    CHECK_ALSA_ERROR("Unable to set start threshold");

    /* disable underrun reporting */
    err = snd_pcm_sw_params_set_stop_threshold
            (p->alsa, alsa_swparams, boundary);
    CHECK_ALSA_ERROR("Unable to set stop threshold");

    /* play silence when there is an underrun */
    err = snd_pcm_sw_params_set_silence_size
            (p->alsa, alsa_swparams, boundary);
    CHECK_ALSA_ERROR("Unable to set silence size");

    err = snd_pcm_sw_params(p->alsa, alsa_swparams);
    CHECK_ALSA_ERROR("Unable to get sw-parameters");

    /* end setting sw-params */

    p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);

    MP_VERBOSE(ao, "opened: %d Hz/%d channels/%d bps/%d samples buffer/%s\n",
               ao->samplerate, ao->channels.num, af_fmt2bits(ao->format),
               p->buffersize, snd_pcm_format_description(p->alsa_fmt));

    return 0;

alsa_error:
    uninit(ao);
    return -1;
} // end init


/* close audio device */
static void uninit(struct ao *ao)
{
    struct priv *p = ao->priv;

    if (p->alsa) {
        int err;

        err = snd_pcm_close(p->alsa);
        CHECK_ALSA_ERROR("pcm close error");

        MP_VERBOSE(ao, "uninit: pcm closed\n");
    }

alsa_error:
    p->alsa = NULL;
}

static void drain(struct ao *ao)
{
    struct priv *p = ao->priv;
    snd_pcm_drain(p->alsa);
}

static void audio_pause(struct ao *ao)
{
    struct priv *p = ao->priv;
    int err;

    if (p->can_pause) {
        if (snd_pcm_state(p->alsa) == SND_PCM_STATE_RUNNING) {
            p->delay_before_pause = get_delay(ao);
            err = snd_pcm_pause(p->alsa, 1);
            CHECK_ALSA_ERROR("pcm pause error");
        }
    } else {
        MP_VERBOSE(ao, "pause not supported by hardware\n");
        if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0
            || p->prepause_frames < 0)
            p->prepause_frames = 0;
        p->delay_before_pause = p->prepause_frames / (float)ao->samplerate;

        err = snd_pcm_drop(p->alsa);
        CHECK_ALSA_ERROR("pcm drop error");
    }

alsa_error: ;
}

static void audio_resume(struct ao *ao)
{
    struct priv *p = ao->priv;
    int err;

    if (snd_pcm_state(p->alsa) == SND_PCM_STATE_SUSPENDED) {
        MP_INFO(ao, "PCM in suspend mode, trying to resume.\n");

        while ((err = snd_pcm_resume(p->alsa)) == -EAGAIN)
            sleep(1);
    }

    if (p->can_pause) {
        if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED) {
            err = snd_pcm_pause(p->alsa, 0);
            CHECK_ALSA_ERROR("pcm resume error");
        }
    } else {
        MP_VERBOSE(ao, "resume not supported by hardware\n");
        err = snd_pcm_prepare(p->alsa);
        CHECK_ALSA_ERROR("pcm prepare error");
        if (p->prepause_frames)
            ao_play_silence(ao, p->prepause_frames);
    }

alsa_error: ;
}

/* stop playing and empty buffers (for seeking/pause) */
static void reset(struct ao *ao)
{
    struct priv *p = ao->priv;
    int err;

    p->prepause_frames = 0;
    p->delay_before_pause = 0;
    err = snd_pcm_drop(p->alsa);
    CHECK_ALSA_ERROR("pcm prepare error");
    err = snd_pcm_prepare(p->alsa);
    CHECK_ALSA_ERROR("pcm prepare error");

alsa_error: ;
}

static int play(struct ao *ao, void **data, int samples, int flags)
{
    struct priv *p = ao->priv;
    snd_pcm_sframes_t res = 0;
    if (!(flags & AOPLAY_FINAL_CHUNK))
        samples = samples / p->outburst * p->outburst;

    if (samples == 0)
        return 0;

    do {
        if (af_fmt_is_planar(ao->format)) {
            res = snd_pcm_writen(p->alsa, data, samples);
        } else {
            res = snd_pcm_writei(p->alsa, data[0], samples);
        }

        if (res == -EINTR) {
            /* nothing to do */
            res = 0;
        } else if (res == -ESTRPIPE) {  /* suspend */
            audio_resume(ao);
        } else if (res < 0) {
            MP_ERR(ao, "Write error: %s\n", snd_strerror(res));
            res = snd_pcm_prepare(p->alsa);
            int err = res;
            CHECK_ALSA_ERROR("pcm prepare error");
            res = 0;
        }
    } while (res == 0);

    return res < 0 ? -1 : res;

alsa_error:
    return -1;
}

static int get_space(struct ao *ao)
{
    struct priv *p = ao->priv;
    snd_pcm_status_t *status;
    int err;

    snd_pcm_status_alloca(&status);

    err = snd_pcm_status(p->alsa, status);
    CHECK_ALSA_ERROR("cannot get pcm status");

    unsigned space = snd_pcm_status_get_avail(status);
    if (space > p->buffersize) // Buffer underrun?
        space = p->buffersize;
    return space;

alsa_error:
    return 0;
}

/* delay in seconds between first and last sample in buffer */
static float get_delay(struct ao *ao)
{
    struct priv *p = ao->priv;
    snd_pcm_sframes_t delay;

    if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED)
        return p->delay_before_pause;

    if (snd_pcm_delay(p->alsa, &delay) < 0)
        return 0;

    if (delay < 0) {
        /* underrun - move the application pointer forward to catch up */
        snd_pcm_forward(p->alsa, -delay);
        delay = 0;
    }
    return (float)delay / (float)ao->samplerate;
}

#define MAX_POLL_FDS 20
static int audio_wait(struct ao *ao, pthread_mutex_t *lock)
{
    struct priv *p = ao->priv;
    int err;

    int num_fds = snd_pcm_poll_descriptors_count(p->alsa);
    if (num_fds <= 0 || num_fds >= MAX_POLL_FDS)
        goto alsa_error;

    struct pollfd fds[MAX_POLL_FDS];
    err = snd_pcm_poll_descriptors(p->alsa, fds, num_fds);
    CHECK_ALSA_ERROR("cannot get pollfds");

    while (1) {
        int r = ao_wait_poll(ao, fds, num_fds, lock);
        if (r)
            return r;

        unsigned short revents;
        snd_pcm_poll_descriptors_revents(p->alsa, fds, num_fds, &revents);
        CHECK_ALSA_ERROR("cannot read poll events");

        if (revents & POLLERR)
            return -1;
        if (revents & POLLOUT)
            return 0;
    }
    return 0;

alsa_error:
    return -1;
}

#define OPT_BASE_STRUCT struct priv

const struct ao_driver audio_out_alsa = {
    .description = "ALSA-0.9.x-1.x audio output",
    .name      = "alsa",
    .init      = init,
    .uninit    = uninit,
    .control   = control,
    .get_space = get_space,
    .play      = play,
    .get_delay = get_delay,
    .pause     = audio_pause,
    .resume    = audio_resume,
    .reset     = reset,
    .drain     = drain,
    .wait      = audio_wait,
    .wakeup    = ao_wakeup_poll,
    .priv_size = sizeof(struct priv),
    .priv_defaults = &(const struct priv) {
        .cfg_block = 1,
        .cfg_mixer_device = "default",
        .cfg_mixer_name = "Master",
        .cfg_mixer_index = 0,
    },
    .options = (const struct m_option[]) {
        OPT_STRING("device", cfg_device, 0),
        OPT_FLAG("resample", cfg_resample, 0),
        OPT_FLAG("block", cfg_block, 0),
        OPT_STRING("mixer-device", cfg_mixer_device, 0),
        OPT_STRING("mixer-name", cfg_mixer_name, 0),
        OPT_INTRANGE("mixer-index", cfg_mixer_index, 0, 0, 99),
        {0}
    },
};