summaryrefslogtreecommitdiffstats
path: root/player/audio.c
blob: ea729ce2cc7318e0fdf4861bdece648f73235bef (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
/*
 * This file is part of mpv.
 *
 * mpv 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.
 *
 * mpv 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 mpv.  If not, see <http://www.gnu.org/licenses/>.
 */

#include <stddef.h>
#include <stdbool.h>
#include <inttypes.h>
#include <limits.h>
#include <math.h>
#include <assert.h>

#include "config.h"
#include "talloc.h"

#include "common/msg.h"
#include "common/encode.h"
#include "options/options.h"
#include "common/common.h"

#include "audio/mixer.h"
#include "audio/audio.h"
#include "audio/audio_buffer.h"
#include "audio/decode/dec_audio.h"
#include "audio/filter/af.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "video/decode/dec_video.h"

#include "core.h"
#include "command.h"

static int update_playback_speed_filters(struct MPContext *mpctx)
{
    struct MPOpts *opts = mpctx->opts;
    double speed = opts->playback_speed;
    struct af_stream *afs = mpctx->d_audio->afilter;

    // Make sure only exactly one filter changes speed; resetting them all
    // and setting 1 filter is the easiest way to achieve this.
    af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &(double){1});
    af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE, &(double){1});

    if (speed == 1.0)
        return af_remove_by_label(afs, "playback-speed");

    // Compatibility: if the user uses --af=scaletempo, always use this
    // filter to change speed. Don't insert a second filter (any) either.
    if (!af_find_by_label(afs, "playback-speed") &&
        af_control_any_rev(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &speed))
        return 0;

    int method = AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE;
    if (opts->pitch_correction)
        method = AF_CONTROL_SET_PLAYBACK_SPEED;

    if (!af_control_any_rev(afs, method, &speed)) {
        if (af_remove_by_label(afs, "playback-speed") < 0)
            return -1;

        char *filter = method == AF_CONTROL_SET_PLAYBACK_SPEED
                     ? "scaletempo" : "lavrresample";
        if (af_add(afs, filter, "playback-speed", NULL) < 0)
            return -1;
        // Try again.
        if (!af_control_any_rev(afs, method, &speed))
            return -1;
    }

    return 0;
}

static int recreate_audio_filters(struct MPContext *mpctx)
{
    assert(mpctx->d_audio);

    if (update_playback_speed_filters(mpctx) < 0) {
        mpctx->opts->playback_speed = 1.0;
        mp_notify(mpctx, MP_EVENT_CHANGE_ALL, NULL);
    }

    struct af_stream *afs = mpctx->d_audio->afilter;
    if (afs->initialized < 1 && af_init(afs) < 0) {
        MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
        return -1;
    }

    mixer_reinit_audio(mpctx->mixer, mpctx->ao, afs);

    return 0;
}

int reinit_audio_filters(struct MPContext *mpctx)
{
    struct dec_audio *d_audio = mpctx->d_audio;
    if (!d_audio)
        return 0;

    af_uninit(mpctx->d_audio->afilter);
    if (af_init(mpctx->d_audio->afilter) < 0)
        return -1;
    if (recreate_audio_filters(mpctx) < 0)
        return -1;

    return 1;
}

void set_playback_speed(struct MPContext *mpctx, double new_speed)
{
    struct MPOpts *opts = mpctx->opts;

    // Adjust time until next frame flip for nosound mode
    mpctx->time_frame *= opts->playback_speed / new_speed;

    opts->playback_speed = new_speed;

    if (!mpctx->d_audio || mpctx->d_audio->afilter->initialized < 1)
        return;

    recreate_audio_filters(mpctx);
}

void reset_audio_state(struct MPContext *mpctx)
{
    if (mpctx->d_audio)
        audio_reset_decoding(mpctx->d_audio);
    if (mpctx->ao_buffer)
        mp_audio_buffer_clear(mpctx->ao_buffer);
    mpctx->audio_status = mpctx->d_audio ? STATUS_SYNCING : STATUS_EOF;
    mpctx->delay = 0;
}

void uninit_audio_out(struct MPContext *mpctx)
{
    if (mpctx->ao) {
        // Note: with gapless_audio, stop_play is not correctly set
        if (mpctx->opts->gapless_audio || mpctx->stop_play == AT_END_OF_FILE)
            ao_drain(mpctx->ao);
        mixer_uninit_audio(mpctx->mixer);
        ao_uninit(mpctx->ao);
    }
    mpctx->ao = NULL;
    talloc_free(mpctx->ao_decoder_fmt);
    mpctx->ao_decoder_fmt = NULL;
}

void uninit_audio_chain(struct MPContext *mpctx)
{
    if (mpctx->d_audio) {
        mixer_uninit_audio(mpctx->mixer);
        audio_uninit(mpctx->d_audio);
        mpctx->d_audio = NULL;
        talloc_free(mpctx->ao_buffer);
        mpctx->ao_buffer = NULL;
        mpctx->audio_status = STATUS_EOF;
        reselect_demux_streams(mpctx);
    }
}

void reinit_audio_chain(struct MPContext *mpctx)
{
    struct MPOpts *opts = mpctx->opts;
    struct track *track = mpctx->current_track[0][STREAM_AUDIO];
    struct sh_stream *sh = track ? track->stream : NULL;
    if (!sh) {
        uninit_audio_out(mpctx);
        goto no_audio;
    }

    mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);

    if (!mpctx->d_audio) {
        mpctx->d_audio = talloc_zero(NULL, struct dec_audio);
        mpctx->d_audio->log = mp_log_new(mpctx->d_audio, mpctx->log, "!ad");
        mpctx->d_audio->global = mpctx->global;
        mpctx->d_audio->opts = opts;
        mpctx->d_audio->header = sh;
        mpctx->d_audio->pool = mp_audio_pool_create(mpctx->d_audio);
        mpctx->d_audio->afilter = af_new(mpctx->global);
        mpctx->d_audio->afilter->replaygain_data = sh->audio->replaygain_data;
        mpctx->ao_buffer = mp_audio_buffer_create(NULL);
        if (!audio_init_best_codec(mpctx->d_audio, opts->audio_decoders))
            goto init_error;
        reset_audio_state(mpctx);

        if (mpctx->ao) {
            struct mp_audio fmt;
            ao_get_format(mpctx->ao, &fmt);
            mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt);
        }
    }
    assert(mpctx->d_audio);

    struct mp_audio in_format = mpctx->d_audio->decode_format;

    if (!mp_audio_config_valid(&in_format)) {
        // We don't know the audio format yet - so configure it later as we're
        // resyncing. fill_audio_buffers() will call this function again.
        mpctx->sleeptime = 0;
        return;
    }

    // Weak gapless audio: drain AO on decoder format changes
    if (mpctx->ao_decoder_fmt && mpctx->ao && opts->gapless_audio < 0 &&
        !mp_audio_config_equals(mpctx->ao_decoder_fmt, &in_format))
    {
        uninit_audio_out(mpctx);
    }

    struct af_stream *afs = mpctx->d_audio->afilter;

    afs->output = (struct mp_audio){0};
    if (mpctx->ao) {
        ao_get_format(mpctx->ao, &afs->output);
    } else if (!AF_FORMAT_IS_SPECIAL(in_format.format)) {
        afs->output.rate = opts->force_srate;
        mp_audio_set_format(&afs->output, opts->audio_output_format);
        mp_audio_set_channels(&afs->output, &opts->audio_output_channels);
    }

    // filter input format: same as codec's output format:
    afs->input = in_format;

    // Determine what the filter chain outputs. recreate_audio_filters() also
    // needs this for testing whether playback speed is changed by resampling
    // or using a special filter.
    if (af_init(afs) < 0) {
        MP_ERR(mpctx, "Error at audio filter chain pre-init!\n");
        goto init_error;
    }

    if (!mpctx->ao) {
        mp_chmap_remove_useless_channels(&afs->output.channels,
                                         &opts->audio_output_channels);
        mp_audio_set_channels(&afs->output, &afs->output.channels);

        mpctx->ao = ao_init_best(mpctx->global, mpctx->input,
                                 mpctx->encode_lavc_ctx, afs->output.rate,
                                 afs->output.format, afs->output.channels);
        struct ao *ao = mpctx->ao;
        if (!ao) {
            MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
            mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED;
            goto init_error;
        }

        struct mp_audio fmt;
        ao_get_format(ao, &fmt);

        mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt);
        afs->output = fmt;
        if (!mp_audio_config_equals(&afs->output, &afs->filter_output))
            afs->initialized = 0;

        mpctx->ao_decoder_fmt = talloc(NULL, struct mp_audio);
        *mpctx->ao_decoder_fmt = in_format;

        MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(ao),
                mp_audio_config_to_str(&fmt));
        MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(ao));
        update_window_title(mpctx, true);
    }

    if (recreate_audio_filters(mpctx) < 0)
        goto init_error;

    set_playback_speed(mpctx, opts->playback_speed);

    return;

init_error:
    uninit_audio_chain(mpctx);
    uninit_audio_out(mpctx);
no_audio:
    if (track)
        error_on_track(mpctx, track);
}

// Return pts value corresponding to the end point of audio written to the
// ao so far.
double written_audio_pts(struct MPContext *mpctx)
{
    struct dec_audio *d_audio = mpctx->d_audio;
    if (!d_audio)
        return MP_NOPTS_VALUE;

    struct mp_audio in_format = d_audio->decode_format;

    if (!mp_audio_config_valid(&in_format) || d_audio->afilter->initialized < 1)
        return MP_NOPTS_VALUE;

    // first calculate the end pts of audio that has been output by decoder
    double a_pts = d_audio->pts;
    if (a_pts == MP_NOPTS_VALUE)
        return MP_NOPTS_VALUE;

    // d_audio->pts is the timestamp of the latest input packet with
    // known pts that the decoder has decoded. d_audio->pts_bytes is
    // the amount of bytes the decoder has written after that timestamp.
    a_pts += d_audio->pts_offset / (double)in_format.rate;

    // Now a_pts hopefully holds the pts for end of audio from decoder.
    // Subtract data in buffers between decoder and audio out.

    // Decoded but not filtered
    if (d_audio->waiting)
        a_pts -= d_audio->waiting->samples / (double)in_format.rate;

    // Data buffered in audio filters, measured in seconds of "missing" output
    double buffered_output = af_calc_delay(d_audio->afilter);

    // Data that was ready for ao but was buffered because ao didn't fully
    // accept everything to internal buffers yet
    buffered_output += mp_audio_buffer_seconds(mpctx->ao_buffer);

    // Filters divide audio length by playback_speed, so multiply by it
    // to get the length in original units without speedup or slowdown
    a_pts -= buffered_output * mpctx->opts->playback_speed;

    return a_pts +
        get_track_video_offset(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
}

// Return pts value corresponding to currently playing audio.
double playing_audio_pts(struct MPContext *mpctx)
{
    double pts = written_audio_pts(mpctx);
    if (pts == MP_NOPTS_VALUE || !mpctx->ao)
        return pts;
    return pts - mpctx->opts->playback_speed * ao_get_delay(mpctx->ao);
}

static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags,
                       double pts)
{
    if (mpctx->paused)
        return 0;
    struct ao *ao = mpctx->ao;
    struct mp_audio out_format;
    ao_get_format(ao, &out_format);
#if HAVE_ENCODING
    encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, playing_audio_pts(mpctx));
#endif
    if (data->samples == 0)
        return 0;
    double real_samplerate = out_format.rate / mpctx->opts->playback_speed;
    int played = ao_play(mpctx->ao, data->planes, data->samples, flags);
    assert(played <= data->samples);
    if (played > 0) {
        mpctx->shown_aframes += played;
        mpctx->delay += played / real_samplerate;
        return played;
    }
    return 0;
}

// Return the number of samples that must be skipped or prepended to reach the
// target audio pts after a seek (for A/V sync or hr-seek).
// Return value (*skip):
//   >0: skip this many samples
//   =0: don't do anything
//   <0: prepend this many samples of silence
// Returns false if PTS is not known yet.
static bool get_sync_samples(struct MPContext *mpctx, int *skip)
{
    struct MPOpts *opts = mpctx->opts;
    *skip = 0;

    if (mpctx->audio_status != STATUS_SYNCING)
        return true;

    struct mp_audio out_format = {0};
    ao_get_format(mpctx->ao, &out_format);
    double play_samplerate = out_format.rate / opts->playback_speed;

    if (!opts->initial_audio_sync) {
        mpctx->audio_status = STATUS_FILLING;
        return true;
    }

    double written_pts = written_audio_pts(mpctx);
    if (written_pts == MP_NOPTS_VALUE && !mp_audio_buffer_samples(mpctx->ao_buffer))
        return false; // no audio read yet

    bool sync_to_video = mpctx->d_video && mpctx->sync_audio_to_video &&
                         mpctx->video_status != STATUS_EOF;

    double sync_pts = MP_NOPTS_VALUE;
    if (sync_to_video) {
        if (mpctx->video_status < STATUS_READY)
            return false; // wait until we know a video PTS
        if (mpctx->video_next_pts != MP_NOPTS_VALUE)
            sync_pts = mpctx->video_next_pts - (opts->audio_delay - mpctx->delay);
    } else if (mpctx->hrseek_active) {
        sync_pts = mpctx->hrseek_pts;
    }
    if (sync_pts == MP_NOPTS_VALUE) {
        mpctx->audio_status = STATUS_FILLING;
        return true; // syncing disabled
    }

    double ptsdiff = written_pts - sync_pts;
    // Missing timestamp, or PTS reset, or just broken.
    if (written_pts == MP_NOPTS_VALUE || fabs(ptsdiff) > 3600) {
        MP_WARN(mpctx, "Failed audio resync.\n");
        mpctx->audio_status = STATUS_FILLING;
        return true;
    }

    int align = af_format_sample_alignment(out_format.format);
    *skip = (-ptsdiff * play_samplerate) / align * align;
    return true;
}

static void do_fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
{
    struct MPOpts *opts = mpctx->opts;
    struct dec_audio *d_audio = mpctx->d_audio;

    if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD)) {
        ao_reset(mpctx->ao);
        uninit_audio_out(mpctx);
        if (d_audio)
            mpctx->audio_status = STATUS_SYNCING;
    }

    if (!d_audio)
        return;

    if (d_audio->afilter->initialized < 1 || !mpctx->ao) {
        // Probe the initial audio format. Returns AD_OK (and does nothing) if
        // the format is already known.
        int r = initial_audio_decode(mpctx->d_audio);
        if (r == AD_WAIT)
            return; // continue later when new data is available
        if (r != AD_OK) {
            mpctx->d_audio->init_retries += 1;
            if (mpctx->d_audio->init_retries >= 50) {
                MP_ERR(mpctx, "Error initializing audio.\n");
                error_on_track(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
                return;
            }
        }
        reinit_audio_chain(mpctx);
        mpctx->sleeptime = 0;
        return; // try again next iteration
    }

    struct mp_audio out_format = {0};
    ao_get_format(mpctx->ao, &out_format);
    double play_samplerate = out_format.rate / opts->playback_speed;

    // If audio is infinitely fast, somehow try keeping approximate A/V sync.
    if (mpctx->audio_status == STATUS_PLAYING && ao_untimed(mpctx->ao) &&
        mpctx->video_status != STATUS_EOF && mpctx->delay > 0)
        return;

    int playsize = ao_get_space(mpctx->ao);

    int skip = 0;
    bool sync_known = get_sync_samples(mpctx, &skip);
    if (skip > 0) {
        playsize = MPMIN(skip + 1, MPMAX(playsize, 2500)); // buffer extra data
    } else if (skip < 0) {
        playsize = MPMAX(1, playsize + skip); // silence will be prepended
    }

    int status = AD_OK;
    if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) {
        status = audio_decode(d_audio, mpctx->ao_buffer, playsize);
        if (status == AD_WAIT)
            return;
        if (status == AD_NEW_FMT) {
            /* The format change isn't handled too gracefully. A more precise
             * implementation would require draining buffered old-format audio
             * while displaying video, then doing the output format switch.
             */
            if (mpctx->opts->gapless_audio < 1)
                uninit_audio_out(mpctx);
            reinit_audio_chain(mpctx);
            mpctx->sleeptime = 0;
            return; // retry on next iteration
        }
        if (status == AD_ERR)
            mpctx->sleeptime = 0;
    }

    // If EOF was reached before, but now something can be decoded, try to
    // restart audio properly. This helps with video files where audio starts
    // later. Retrying is needed to get the correct sync PTS.
    if (mpctx->audio_status >= STATUS_DRAINING && status == AD_OK) {
        mpctx->audio_status = STATUS_SYNCING;
        return; // retry on next iteration
    }

    bool end_sync = false;
    if (skip >= 0) {
        int max = mp_audio_buffer_samples(mpctx->ao_buffer);
        mp_audio_buffer_skip(mpctx->ao_buffer, MPMIN(skip, max));
        // If something is left, we definitely reached the target time.
        end_sync |= sync_known && skip < max;
    } else if (skip < 0) {
        if (-skip > playsize) { // heuristic against making the buffer too large
            ao_reset(mpctx->ao); // some AOs repeat data on underflow
            mpctx->audio_status = STATUS_DRAINING;
            mpctx->delay = 0;
            return;
        }
        mp_audio_buffer_prepend_silence(mpctx->ao_buffer, -skip);
        end_sync = true;
    }

    if (mpctx->audio_status == STATUS_SYNCING) {
        if (end_sync)
            mpctx->audio_status = STATUS_FILLING;
        if (status != AD_OK && !mp_audio_buffer_samples(mpctx->ao_buffer))
            mpctx->audio_status = STATUS_EOF;
        mpctx->sleeptime = 0;
        return; // continue on next iteration
    }

    assert(mpctx->audio_status >= STATUS_FILLING);

    // Even if we're done decoding and syncing, let video start first - this is
    // required, because sending audio to the AO already starts playback.
    if (mpctx->audio_status == STATUS_FILLING && mpctx->sync_audio_to_video &&
        mpctx->video_status <= STATUS_READY)
    {
        mpctx->audio_status = STATUS_READY;
        return;
    }

    bool audio_eof = status == AD_EOF;
    bool partial_fill = false;
    int playflags = 0;

    if (endpts != MP_NOPTS_VALUE) {
        double samples = (endpts - written_audio_pts(mpctx) - opts->audio_delay)
                         * play_samplerate;
        if (playsize > samples) {
            playsize = MPMAX(samples, 0);
            audio_eof = true;
            partial_fill = true;
        }
    }

    if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) {
        playsize = mp_audio_buffer_samples(mpctx->ao_buffer);
        partial_fill = true;
    }

    audio_eof &= partial_fill;

    // With gapless audio, delay this to ao_uninit. There must be only
    // 1 final chunk, and that is handled when calling ao_uninit().
    if (audio_eof && !opts->gapless_audio)
        playflags |= AOPLAY_FINAL_CHUNK;

    if (mpctx->paused)
        playsize = 0;

    struct mp_audio data;
    mp_audio_buffer_peek(mpctx->ao_buffer, &data);
    data.samples = MPMIN(data.samples, playsize);
    int played = write_to_ao(mpctx, &data, playflags, written_audio_pts(mpctx));
    assert(played >= 0 && played <= data.samples);
    mp_audio_buffer_skip(mpctx->ao_buffer, played);

    mpctx->audio_status = STATUS_PLAYING;
    if (audio_eof) {
        mpctx->audio_status = STATUS_DRAINING;
        // Wait until the AO has played all queued data. In the gapless case,
        // we trigger EOF immediately, and let it play asynchronously.
        if (ao_eof_reached(mpctx->ao) || opts->gapless_audio)
            mpctx->audio_status = STATUS_EOF;
    }
}

void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
{
    do_fill_audio_out_buffers(mpctx, endpts);
    // Run audio playback state machine again to display the actual audio PTS
    // as current time on OSD in audio-only mode in most situations.
    if (mpctx->audio_status == STATUS_SYNCING)
        do_fill_audio_out_buffers(mpctx, endpts);
}

// Drop data queued for output, or which the AO is currently outputting.
void clear_audio_output_buffers(struct MPContext *mpctx)
{
    if (mpctx->ao)
        ao_reset(mpctx->ao);
    if (mpctx->ao_buffer)
        mp_audio_buffer_clear(mpctx->ao_buffer);
}