Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / opusdec.c
CommitLineData
2ba45a60
DM
1/*
2 * Opus decoder
3 * Copyright (c) 2012 Andrew D'Addesio
4 * Copyright (c) 2013-2014 Mozilla Corporation
5 *
6 * This file is part of FFmpeg.
7 *
8 * FFmpeg is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * FFmpeg is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23/**
24 * @file
25 * Opus decoder
26 * @author Andrew D'Addesio, Anton Khirnov
27 *
28 * Codec homepage: http://opus-codec.org/
29 * Specification: http://tools.ietf.org/html/rfc6716
30 * Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03
31 *
32 * Ogg-contained .opus files can be produced with opus-tools:
33 * http://git.xiph.org/?p=opus-tools.git
34 */
35
36#include <stdint.h>
37
38#include "libavutil/attributes.h"
39#include "libavutil/audio_fifo.h"
40#include "libavutil/channel_layout.h"
41#include "libavutil/opt.h"
42
43#include "libswresample/swresample.h"
44
45#include "avcodec.h"
46#include "celp_filters.h"
47#include "fft.h"
48#include "get_bits.h"
49#include "internal.h"
50#include "mathops.h"
51#include "opus.h"
52
53static const uint16_t silk_frame_duration_ms[16] = {
54 10, 20, 40, 60,
55 10, 20, 40, 60,
56 10, 20, 40, 60,
57 10, 20,
58 10, 20,
59};
60
61/* number of samples of silence to feed to the resampler
62 * at the beginning */
63static const int silk_resample_delay[] = {
64 4, 8, 11, 11, 11
65};
66
67static const uint8_t celt_band_end[] = { 13, 17, 17, 19, 21 };
68
69static int get_silk_samplerate(int config)
70{
71 if (config < 4)
72 return 8000;
73 else if (config < 8)
74 return 12000;
75 return 16000;
76}
77
78/**
79 * Range decoder
80 */
81static int opus_rc_init(OpusRangeCoder *rc, const uint8_t *data, int size)
82{
83 int ret = init_get_bits8(&rc->gb, data, size);
84 if (ret < 0)
85 return ret;
86
87 rc->range = 128;
88 rc->value = 127 - get_bits(&rc->gb, 7);
89 rc->total_read_bits = 9;
90 opus_rc_normalize(rc);
91
92 return 0;
93}
94
95static void opus_raw_init(OpusRangeCoder *rc, const uint8_t *rightend,
96 unsigned int bytes)
97{
98 rc->rb.position = rightend;
99 rc->rb.bytes = bytes;
100 rc->rb.cachelen = 0;
101 rc->rb.cacheval = 0;
102}
103
104static void opus_fade(float *out,
105 const float *in1, const float *in2,
106 const float *window, int len)
107{
108 int i;
109 for (i = 0; i < len; i++)
110 out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
111}
112
113static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
114{
115 int celt_size = av_audio_fifo_size(s->celt_delay);
116 int ret, i;
117 ret = swr_convert(s->swr,
118 (uint8_t**)s->out, nb_samples,
119 NULL, 0);
120 if (ret < 0)
121 return ret;
122 else if (ret != nb_samples) {
123 av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
124 ret);
125 return AVERROR_BUG;
126 }
127
128 if (celt_size) {
129 if (celt_size != nb_samples) {
130 av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
131 return AVERROR_BUG;
132 }
133 av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
134 for (i = 0; i < s->output_channels; i++) {
135 s->fdsp->vector_fmac_scalar(s->out[i],
136 s->celt_output[i], 1.0,
137 nb_samples);
138 }
139 }
140
141 if (s->redundancy_idx) {
142 for (i = 0; i < s->output_channels; i++)
143 opus_fade(s->out[i], s->out[i],
144 s->redundancy_output[i] + 120 + s->redundancy_idx,
145 ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
146 s->redundancy_idx = 0;
147 }
148
149 s->out[0] += nb_samples;
150 s->out[1] += nb_samples;
151 s->out_size -= nb_samples * sizeof(float);
152
153 return 0;
154}
155
156static int opus_init_resample(OpusStreamContext *s)
157{
158 static const float delay[16] = { 0.0 };
159 const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
160 int ret;
161
162 av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
163 ret = swr_init(s->swr);
164 if (ret < 0) {
165 av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
166 return ret;
167 }
168
169 ret = swr_convert(s->swr,
170 NULL, 0,
171 delayptr, silk_resample_delay[s->packet.bandwidth]);
172 if (ret < 0) {
173 av_log(s->avctx, AV_LOG_ERROR,
174 "Error feeding initial silence to the resampler.\n");
175 return ret;
176 }
177
178 return 0;
179}
180
181static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
182{
183 int ret;
184 enum OpusBandwidth bw = s->packet.bandwidth;
185
186 if (s->packet.mode == OPUS_MODE_SILK &&
187 bw == OPUS_BANDWIDTH_MEDIUMBAND)
188 bw = OPUS_BANDWIDTH_WIDEBAND;
189
190 ret = opus_rc_init(&s->redundancy_rc, data, size);
191 if (ret < 0)
192 goto fail;
193 opus_raw_init(&s->redundancy_rc, data + size, size);
194
195 ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
196 s->redundancy_output,
197 s->packet.stereo + 1, 240,
198 0, celt_band_end[s->packet.bandwidth]);
199 if (ret < 0)
200 goto fail;
201
202 return 0;
203fail:
204 av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
205 return ret;
206}
207
208static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
209{
210 int samples = s->packet.frame_duration;
211 int redundancy = 0;
212 int redundancy_size, redundancy_pos;
213 int ret, i, consumed;
214 int delayed_samples = s->delayed_samples;
215
216 ret = opus_rc_init(&s->rc, data, size);
217 if (ret < 0)
218 return ret;
219
220 /* decode the silk frame */
221 if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
222 if (!swr_is_initialized(s->swr)) {
223 ret = opus_init_resample(s);
224 if (ret < 0)
225 return ret;
226 }
227
228 samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
229 FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
230 s->packet.stereo + 1,
231 silk_frame_duration_ms[s->packet.config]);
232 if (samples < 0) {
233 av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
234 return samples;
235 }
236 samples = swr_convert(s->swr,
237 (uint8_t**)s->out, s->packet.frame_duration,
238 (const uint8_t**)s->silk_output, samples);
239 if (samples < 0) {
240 av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
241 return samples;
242 }
243 av_assert2((samples & 7) == 0);
244 s->delayed_samples += s->packet.frame_duration - samples;
245 } else
246 ff_silk_flush(s->silk);
247
248 // decode redundancy information
249 consumed = opus_rc_tell(&s->rc);
250 if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
251 redundancy = opus_rc_p2model(&s->rc, 12);
252 else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
253 redundancy = 1;
254
255 if (redundancy) {
256 redundancy_pos = opus_rc_p2model(&s->rc, 1);
257
258 if (s->packet.mode == OPUS_MODE_HYBRID)
259 redundancy_size = opus_rc_unimodel(&s->rc, 256) + 2;
260 else
261 redundancy_size = size - (consumed + 7) / 8;
262 size -= redundancy_size;
263 if (size < 0) {
264 av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
265 return AVERROR_INVALIDDATA;
266 }
267
268 if (redundancy_pos) {
269 ret = opus_decode_redundancy(s, data + size, redundancy_size);
270 if (ret < 0)
271 return ret;
272 ff_celt_flush(s->celt);
273 }
274 }
275
276 /* decode the CELT frame */
277 if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
278 float *out_tmp[2] = { s->out[0], s->out[1] };
279 float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
280 out_tmp : s->celt_output;
281 int celt_output_samples = samples;
282 int delay_samples = av_audio_fifo_size(s->celt_delay);
283
284 if (delay_samples) {
285 if (s->packet.mode == OPUS_MODE_HYBRID) {
286 av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
287
288 for (i = 0; i < s->output_channels; i++) {
289 s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
290 delay_samples);
291 out_tmp[i] += delay_samples;
292 }
293 celt_output_samples -= delay_samples;
294 } else {
295 av_log(s->avctx, AV_LOG_WARNING,
296 "Spurious CELT delay samples present.\n");
297 av_audio_fifo_drain(s->celt_delay, delay_samples);
298 if (s->avctx->err_recognition & AV_EF_EXPLODE)
299 return AVERROR_BUG;
300 }
301 }
302
303 opus_raw_init(&s->rc, data + size, size);
304
305 ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
306 s->packet.stereo + 1,
307 s->packet.frame_duration,
308 (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
309 celt_band_end[s->packet.bandwidth]);
310 if (ret < 0)
311 return ret;
312
313 if (s->packet.mode == OPUS_MODE_HYBRID) {
314 int celt_delay = s->packet.frame_duration - celt_output_samples;
315 void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
316 s->celt_output[1] + celt_output_samples };
317
318 for (i = 0; i < s->output_channels; i++) {
319 s->fdsp->vector_fmac_scalar(out_tmp[i],
320 s->celt_output[i], 1.0,
321 celt_output_samples);
322 }
323
324 ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
325 if (ret < 0)
326 return ret;
327 }
328 } else
329 ff_celt_flush(s->celt);
330
331 if (s->redundancy_idx) {
332 for (i = 0; i < s->output_channels; i++)
333 opus_fade(s->out[i], s->out[i],
334 s->redundancy_output[i] + 120 + s->redundancy_idx,
335 ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
336 s->redundancy_idx = 0;
337 }
338 if (redundancy) {
339 if (!redundancy_pos) {
340 ff_celt_flush(s->celt);
341 ret = opus_decode_redundancy(s, data + size, redundancy_size);
342 if (ret < 0)
343 return ret;
344
345 for (i = 0; i < s->output_channels; i++) {
346 opus_fade(s->out[i] + samples - 120 + delayed_samples,
347 s->out[i] + samples - 120 + delayed_samples,
348 s->redundancy_output[i] + 120,
349 ff_celt_window2, 120 - delayed_samples);
350 if (delayed_samples)
351 s->redundancy_idx = 120 - delayed_samples;
352 }
353 } else {
354 for (i = 0; i < s->output_channels; i++) {
355 memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
356 opus_fade(s->out[i] + 120 + delayed_samples,
357 s->redundancy_output[i] + 120,
358 s->out[i] + 120 + delayed_samples,
359 ff_celt_window2, 120);
360 }
361 }
362 }
363
364 return samples;
365}
366
367static int opus_decode_subpacket(OpusStreamContext *s,
368 const uint8_t *buf, int buf_size,
369 int nb_samples)
370{
371 int output_samples = 0;
372 int flush_needed = 0;
373 int i, j, ret;
374
375 /* check if we need to flush the resampler */
376 if (swr_is_initialized(s->swr)) {
377 if (buf) {
378 int64_t cur_samplerate;
379 av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
380 flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
381 } else {
382 flush_needed = !!s->delayed_samples;
383 }
384 }
385
386 if (!buf && !flush_needed)
387 return 0;
388
389 /* use dummy output buffers if the channel is not mapped to anything */
390 if (!s->out[0] ||
391 (s->output_channels == 2 && !s->out[1])) {
392 av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
393 if (!s->out_dummy)
394 return AVERROR(ENOMEM);
395 if (!s->out[0])
396 s->out[0] = s->out_dummy;
397 if (!s->out[1])
398 s->out[1] = s->out_dummy;
399 }
400
401 /* flush the resampler if necessary */
402 if (flush_needed) {
403 ret = opus_flush_resample(s, s->delayed_samples);
404 if (ret < 0) {
405 av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
406 return ret;
407 }
408 swr_close(s->swr);
409 output_samples += s->delayed_samples;
410 s->delayed_samples = 0;
411
412 if (!buf)
413 goto finish;
414 }
415
416 /* decode all the frames in the packet */
417 for (i = 0; i < s->packet.frame_count; i++) {
418 int size = s->packet.frame_size[i];
419 int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
420
421 if (samples < 0) {
422 av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
423 if (s->avctx->err_recognition & AV_EF_EXPLODE)
424 return samples;
425
426 for (j = 0; j < s->output_channels; j++)
427 memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
428 samples = s->packet.frame_duration;
429 }
430 output_samples += samples;
431
432 for (j = 0; j < s->output_channels; j++)
433 s->out[j] += samples;
434 s->out_size -= samples * sizeof(float);
435 }
436
437finish:
438 s->out[0] = s->out[1] = NULL;
439 s->out_size = 0;
440
441 return output_samples;
442}
443
444static int opus_decode_packet(AVCodecContext *avctx, void *data,
445 int *got_frame_ptr, AVPacket *avpkt)
446{
447 OpusContext *c = avctx->priv_data;
448 AVFrame *frame = data;
449 const uint8_t *buf = avpkt->data;
450 int buf_size = avpkt->size;
451 int coded_samples = 0;
452 int decoded_samples = 0;
453 int i, ret;
454
455 /* decode the header of the first sub-packet to find out the sample count */
456 if (buf) {
457 OpusPacket *pkt = &c->streams[0].packet;
458 ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
459 if (ret < 0) {
460 av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
461 return ret;
462 }
463 coded_samples += pkt->frame_count * pkt->frame_duration;
464 c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
465 }
466
467 frame->nb_samples = coded_samples + c->streams[0].delayed_samples;
468
469 /* no input or buffered data => nothing to do */
470 if (!frame->nb_samples) {
471 *got_frame_ptr = 0;
472 return 0;
473 }
474
475 /* setup the data buffers */
476 ret = ff_get_buffer(avctx, frame, 0);
477 if (ret < 0) {
478 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
479 return ret;
480 }
481 frame->nb_samples = 0;
482
483 for (i = 0; i < avctx->channels; i++) {
484 ChannelMap *map = &c->channel_maps[i];
485 if (!map->copy)
486 c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
487 }
488
489 for (i = 0; i < c->nb_streams; i++)
490 c->streams[i].out_size = frame->linesize[0];
491
492 /* decode each sub-packet */
493 for (i = 0; i < c->nb_streams; i++) {
494 OpusStreamContext *s = &c->streams[i];
495
496 if (i && buf) {
497 ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
498 if (ret < 0) {
499 av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
500 return ret;
501 }
f6fa7814
DM
502 if (coded_samples != s->packet.frame_count * s->packet.frame_duration) {
503 av_log(avctx, AV_LOG_ERROR,
504 "Mismatching coded sample count in substream %d.\n", i);
505 return AVERROR_INVALIDDATA;
506 }
507
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DM
508 s->silk_samplerate = get_silk_samplerate(s->packet.config);
509 }
510
511 ret = opus_decode_subpacket(&c->streams[i], buf,
512 s->packet.data_size, coded_samples);
513 if (ret < 0)
514 return ret;
515 if (decoded_samples && ret != decoded_samples) {
516 av_log(avctx, AV_LOG_ERROR, "Different numbers of decoded samples "
517 "in a multi-channel stream\n");
518 return AVERROR_INVALIDDATA;
519 }
520 decoded_samples = ret;
521 buf += s->packet.packet_size;
522 buf_size -= s->packet.packet_size;
523 }
524
525 for (i = 0; i < avctx->channels; i++) {
526 ChannelMap *map = &c->channel_maps[i];
527
528 /* handle copied channels */
529 if (map->copy) {
530 memcpy(frame->extended_data[i],
531 frame->extended_data[map->copy_idx],
532 frame->linesize[0]);
533 } else if (map->silence) {
534 memset(frame->extended_data[i], 0, frame->linesize[0]);
535 }
536
537 if (c->gain_i) {
f6fa7814 538 c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i],
2ba45a60
DM
539 (float*)frame->extended_data[i],
540 c->gain, FFALIGN(decoded_samples, 8));
541 }
542 }
543
544 frame->nb_samples = decoded_samples;
545 *got_frame_ptr = !!decoded_samples;
546
547 return avpkt->size;
548}
549
550static av_cold void opus_decode_flush(AVCodecContext *ctx)
551{
552 OpusContext *c = ctx->priv_data;
553 int i;
554
555 for (i = 0; i < c->nb_streams; i++) {
556 OpusStreamContext *s = &c->streams[i];
557
558 memset(&s->packet, 0, sizeof(s->packet));
559 s->delayed_samples = 0;
560
561 if (s->celt_delay)
562 av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
563 swr_close(s->swr);
564
565 ff_silk_flush(s->silk);
566 ff_celt_flush(s->celt);
567 }
568}
569
570static av_cold int opus_decode_close(AVCodecContext *avctx)
571{
572 OpusContext *c = avctx->priv_data;
573 int i;
574
575 for (i = 0; i < c->nb_streams; i++) {
576 OpusStreamContext *s = &c->streams[i];
577
578 ff_silk_free(&s->silk);
579 ff_celt_free(&s->celt);
580
581 av_freep(&s->out_dummy);
582 s->out_dummy_allocated_size = 0;
583
584 av_audio_fifo_free(s->celt_delay);
585 swr_free(&s->swr);
586 }
587
588 av_freep(&c->streams);
589 c->nb_streams = 0;
590
591 av_freep(&c->channel_maps);
f6fa7814 592 av_freep(&c->fdsp);
2ba45a60
DM
593
594 return 0;
595}
596
597static av_cold int opus_decode_init(AVCodecContext *avctx)
598{
599 OpusContext *c = avctx->priv_data;
600 int ret, i, j;
601
602 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
603 avctx->sample_rate = 48000;
604
f6fa7814
DM
605 c->fdsp = avpriv_float_dsp_alloc(0);
606 if (!c->fdsp)
607 return AVERROR(ENOMEM);
2ba45a60
DM
608
609 /* find out the channel configuration */
610 ret = ff_opus_parse_extradata(avctx, c);
611 if (ret < 0)
612 return ret;
613
614 /* allocate and init each independent decoder */
615 c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
616 if (!c->streams) {
617 c->nb_streams = 0;
618 ret = AVERROR(ENOMEM);
619 goto fail;
620 }
621
622 for (i = 0; i < c->nb_streams; i++) {
623 OpusStreamContext *s = &c->streams[i];
624 uint64_t layout;
625
626 s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
627
628 s->avctx = avctx;
629
630 for (j = 0; j < s->output_channels; j++) {
631 s->silk_output[j] = s->silk_buf[j];
632 s->celt_output[j] = s->celt_buf[j];
633 s->redundancy_output[j] = s->redundancy_buf[j];
634 }
635
f6fa7814 636 s->fdsp = c->fdsp;
2ba45a60
DM
637
638 s->swr =swr_alloc();
639 if (!s->swr)
640 goto fail;
641
642 layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
643 av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
644 av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
645 av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
646 av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
647 av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
648 av_opt_set_int(s->swr, "filter_size", 16, 0);
649
650 ret = ff_silk_init(avctx, &s->silk, s->output_channels);
651 if (ret < 0)
652 goto fail;
653
654 ret = ff_celt_init(avctx, &s->celt, s->output_channels);
655 if (ret < 0)
656 goto fail;
657
658 s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
659 s->output_channels, 1024);
660 if (!s->celt_delay) {
661 ret = AVERROR(ENOMEM);
662 goto fail;
663 }
664 }
665
666 return 0;
667fail:
668 opus_decode_close(avctx);
669 return ret;
670}
671
672AVCodec ff_opus_decoder = {
673 .name = "opus",
674 .long_name = NULL_IF_CONFIG_SMALL("Opus"),
675 .type = AVMEDIA_TYPE_AUDIO,
676 .id = AV_CODEC_ID_OPUS,
677 .priv_data_size = sizeof(OpusContext),
678 .init = opus_decode_init,
679 .close = opus_decode_close,
680 .decode = opus_decode_packet,
681 .flush = opus_decode_flush,
682 .capabilities = CODEC_CAP_DR1 | CODEC_CAP_DELAY,
683};