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opusdec.c
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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"
41 #include "libavutil/opt.h"
42 
44 
45 #include "avcodec.h"
46 #include "get_bits.h"
47 #include "internal.h"
48 #include "mathops.h"
49 #include "opus.h"
50 #include "opustab.h"
51 #include "opus_celt.h"
52 
53 static 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 */
63 static const int silk_resample_delay[] = {
64  4, 8, 11, 11, 11
65 };
66 
67 static int get_silk_samplerate(int config)
68 {
69  if (config < 4)
70  return 8000;
71  else if (config < 8)
72  return 12000;
73  return 16000;
74 }
75 
76 static void opus_fade(float *out,
77  const float *in1, const float *in2,
78  const float *window, int len)
79 {
80  int i;
81  for (i = 0; i < len; i++)
82  out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
83 }
84 
85 static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
86 {
87  int celt_size = av_audio_fifo_size(s->celt_delay);
88  int ret, i;
89  ret = swr_convert(s->swr,
90  (uint8_t**)s->out, nb_samples,
91  NULL, 0);
92  if (ret < 0)
93  return ret;
94  else if (ret != nb_samples) {
95  av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
96  ret);
97  return AVERROR_BUG;
98  }
99 
100  if (celt_size) {
101  if (celt_size != nb_samples) {
102  av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
103  return AVERROR_BUG;
104  }
105  av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
106  for (i = 0; i < s->output_channels; i++) {
107  s->fdsp->vector_fmac_scalar(s->out[i],
108  s->celt_output[i], 1.0,
109  nb_samples);
110  }
111  }
112 
113  if (s->redundancy_idx) {
114  for (i = 0; i < s->output_channels; i++)
115  opus_fade(s->out[i], s->out[i],
116  s->redundancy_output[i] + 120 + s->redundancy_idx,
118  s->redundancy_idx = 0;
119  }
120 
121  s->out[0] += nb_samples;
122  s->out[1] += nb_samples;
123  s->out_size -= nb_samples * sizeof(float);
124 
125  return 0;
126 }
127 
129 {
130  static const float delay[16] = { 0.0 };
131  const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
132  int ret;
133 
134  av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
135  ret = swr_init(s->swr);
136  if (ret < 0) {
137  av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
138  return ret;
139  }
140 
141  ret = swr_convert(s->swr,
142  NULL, 0,
143  delayptr, silk_resample_delay[s->packet.bandwidth]);
144  if (ret < 0) {
146  "Error feeding initial silence to the resampler.\n");
147  return ret;
148  }
149 
150  return 0;
151 }
152 
154 {
155  int ret = ff_opus_rc_dec_init(&s->redundancy_rc, data, size);
156  if (ret < 0)
157  goto fail;
158  ff_opus_rc_dec_raw_init(&s->redundancy_rc, data + size, size);
159 
162  s->packet.stereo + 1, 240,
164  if (ret < 0)
165  goto fail;
166 
167  return 0;
168 fail:
169  av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
170  return ret;
171 }
172 
174 {
175  int samples = s->packet.frame_duration;
176  int redundancy = 0;
177  int redundancy_size, redundancy_pos;
178  int ret, i, consumed;
179  int delayed_samples = s->delayed_samples;
180 
181  ret = ff_opus_rc_dec_init(&s->rc, data, size);
182  if (ret < 0)
183  return ret;
184 
185  /* decode the silk frame */
186  if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
187  if (!swr_is_initialized(s->swr)) {
188  ret = opus_init_resample(s);
189  if (ret < 0)
190  return ret;
191  }
192 
193  samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
195  s->packet.stereo + 1,
197  if (samples < 0) {
198  av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
199  return samples;
200  }
201  samples = swr_convert(s->swr,
202  (uint8_t**)s->out, s->packet.frame_duration,
203  (const uint8_t**)s->silk_output, samples);
204  if (samples < 0) {
205  av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
206  return samples;
207  }
208  av_assert2((samples & 7) == 0);
209  s->delayed_samples += s->packet.frame_duration - samples;
210  } else
211  ff_silk_flush(s->silk);
212 
213  // decode redundancy information
214  consumed = opus_rc_tell(&s->rc);
215  if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
216  redundancy = ff_opus_rc_dec_log(&s->rc, 12);
217  else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
218  redundancy = 1;
219 
220  if (redundancy) {
221  redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);
222 
223  if (s->packet.mode == OPUS_MODE_HYBRID)
224  redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2;
225  else
226  redundancy_size = size - (consumed + 7) / 8;
227  size -= redundancy_size;
228  if (size < 0) {
229  av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
230  return AVERROR_INVALIDDATA;
231  }
232 
233  if (redundancy_pos) {
234  ret = opus_decode_redundancy(s, data + size, redundancy_size);
235  if (ret < 0)
236  return ret;
237  ff_celt_flush(s->celt);
238  }
239  }
240 
241  /* decode the CELT frame */
242  if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
243  float *out_tmp[2] = { s->out[0], s->out[1] };
244  float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
245  out_tmp : s->celt_output;
246  int celt_output_samples = samples;
247  int delay_samples = av_audio_fifo_size(s->celt_delay);
248 
249  if (delay_samples) {
250  if (s->packet.mode == OPUS_MODE_HYBRID) {
251  av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
252 
253  for (i = 0; i < s->output_channels; i++) {
254  s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
255  delay_samples);
256  out_tmp[i] += delay_samples;
257  }
258  celt_output_samples -= delay_samples;
259  } else {
261  "Spurious CELT delay samples present.\n");
262  av_audio_fifo_drain(s->celt_delay, delay_samples);
264  return AVERROR_BUG;
265  }
266  }
267 
268  ff_opus_rc_dec_raw_init(&s->rc, data + size, size);
269 
270  ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
271  s->packet.stereo + 1,
273  (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
275  if (ret < 0)
276  return ret;
277 
278  if (s->packet.mode == OPUS_MODE_HYBRID) {
279  int celt_delay = s->packet.frame_duration - celt_output_samples;
280  void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
281  s->celt_output[1] + celt_output_samples };
282 
283  for (i = 0; i < s->output_channels; i++) {
284  s->fdsp->vector_fmac_scalar(out_tmp[i],
285  s->celt_output[i], 1.0,
286  celt_output_samples);
287  }
288 
289  ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
290  if (ret < 0)
291  return ret;
292  }
293  } else
294  ff_celt_flush(s->celt);
295 
296  if (s->redundancy_idx) {
297  for (i = 0; i < s->output_channels; i++)
298  opus_fade(s->out[i], s->out[i],
299  s->redundancy_output[i] + 120 + s->redundancy_idx,
301  s->redundancy_idx = 0;
302  }
303  if (redundancy) {
304  if (!redundancy_pos) {
305  ff_celt_flush(s->celt);
306  ret = opus_decode_redundancy(s, data + size, redundancy_size);
307  if (ret < 0)
308  return ret;
309 
310  for (i = 0; i < s->output_channels; i++) {
311  opus_fade(s->out[i] + samples - 120 + delayed_samples,
312  s->out[i] + samples - 120 + delayed_samples,
313  s->redundancy_output[i] + 120,
314  ff_celt_window2, 120 - delayed_samples);
315  if (delayed_samples)
316  s->redundancy_idx = 120 - delayed_samples;
317  }
318  } else {
319  for (i = 0; i < s->output_channels; i++) {
320  memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
321  opus_fade(s->out[i] + 120 + delayed_samples,
322  s->redundancy_output[i] + 120,
323  s->out[i] + 120 + delayed_samples,
324  ff_celt_window2, 120);
325  }
326  }
327  }
328 
329  return samples;
330 }
331 
333  const uint8_t *buf, int buf_size,
334  float **out, int out_size,
335  int nb_samples)
336 {
337  int output_samples = 0;
338  int flush_needed = 0;
339  int i, j, ret;
340 
341  s->out[0] = out[0];
342  s->out[1] = out[1];
343  s->out_size = out_size;
344 
345  /* check if we need to flush the resampler */
346  if (swr_is_initialized(s->swr)) {
347  if (buf) {
348  int64_t cur_samplerate;
349  av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
350  flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
351  } else {
352  flush_needed = !!s->delayed_samples;
353  }
354  }
355 
356  if (!buf && !flush_needed)
357  return 0;
358 
359  /* use dummy output buffers if the channel is not mapped to anything */
360  if (!s->out[0] ||
361  (s->output_channels == 2 && !s->out[1])) {
363  if (!s->out_dummy)
364  return AVERROR(ENOMEM);
365  if (!s->out[0])
366  s->out[0] = s->out_dummy;
367  if (!s->out[1])
368  s->out[1] = s->out_dummy;
369  }
370 
371  /* flush the resampler if necessary */
372  if (flush_needed) {
374  if (ret < 0) {
375  av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
376  return ret;
377  }
378  swr_close(s->swr);
379  output_samples += s->delayed_samples;
380  s->delayed_samples = 0;
381 
382  if (!buf)
383  goto finish;
384  }
385 
386  /* decode all the frames in the packet */
387  for (i = 0; i < s->packet.frame_count; i++) {
388  int size = s->packet.frame_size[i];
389  int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
390 
391  if (samples < 0) {
392  av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
394  return samples;
395 
396  for (j = 0; j < s->output_channels; j++)
397  memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
398  samples = s->packet.frame_duration;
399  }
400  output_samples += samples;
401 
402  for (j = 0; j < s->output_channels; j++)
403  s->out[j] += samples;
404  s->out_size -= samples * sizeof(float);
405  }
406 
407 finish:
408  s->out[0] = s->out[1] = NULL;
409  s->out_size = 0;
410 
411  return output_samples;
412 }
413 
414 static int opus_decode_packet(AVCodecContext *avctx, void *data,
415  int *got_frame_ptr, AVPacket *avpkt)
416 {
417  OpusContext *c = avctx->priv_data;
418  AVFrame *frame = data;
419  const uint8_t *buf = avpkt->data;
420  int buf_size = avpkt->size;
421  int coded_samples = 0;
422  int decoded_samples = INT_MAX;
423  int delayed_samples = 0;
424  int i, ret;
425 
426  /* calculate the number of delayed samples */
427  for (i = 0; i < c->nb_streams; i++) {
428  OpusStreamContext *s = &c->streams[i];
429  s->out[0] =
430  s->out[1] = NULL;
431  delayed_samples = FFMAX(delayed_samples,
433  }
434 
435  /* decode the header of the first sub-packet to find out the sample count */
436  if (buf) {
437  OpusPacket *pkt = &c->streams[0].packet;
438  ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
439  if (ret < 0) {
440  av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
441  return ret;
442  }
443  coded_samples += pkt->frame_count * pkt->frame_duration;
445  }
446 
447  frame->nb_samples = coded_samples + delayed_samples;
448 
449  /* no input or buffered data => nothing to do */
450  if (!frame->nb_samples) {
451  *got_frame_ptr = 0;
452  return 0;
453  }
454 
455  /* setup the data buffers */
456  ret = ff_get_buffer(avctx, frame, 0);
457  if (ret < 0)
458  return ret;
459  frame->nb_samples = 0;
460 
461  memset(c->out, 0, c->nb_streams * 2 * sizeof(*c->out));
462  for (i = 0; i < avctx->channels; i++) {
463  ChannelMap *map = &c->channel_maps[i];
464  if (!map->copy)
465  c->out[2 * map->stream_idx + map->channel_idx] = (float*)frame->extended_data[i];
466  }
467 
468  /* read the data from the sync buffers */
469  for (i = 0; i < c->nb_streams; i++) {
470  float **out = c->out + 2 * i;
471  int sync_size = av_audio_fifo_size(c->sync_buffers[i]);
472 
473  float sync_dummy[32];
474  int out_dummy = (!out[0]) | ((!out[1]) << 1);
475 
476  if (!out[0])
477  out[0] = sync_dummy;
478  if (!out[1])
479  out[1] = sync_dummy;
480  if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
481  return AVERROR_BUG;
482 
483  ret = av_audio_fifo_read(c->sync_buffers[i], (void**)out, sync_size);
484  if (ret < 0)
485  return ret;
486 
487  if (out_dummy & 1)
488  out[0] = NULL;
489  else
490  out[0] += ret;
491  if (out_dummy & 2)
492  out[1] = NULL;
493  else
494  out[1] += ret;
495 
496  c->out_size[i] = frame->linesize[0] - ret * sizeof(float);
497  }
498 
499  /* decode each sub-packet */
500  for (i = 0; i < c->nb_streams; i++) {
501  OpusStreamContext *s = &c->streams[i];
502 
503  if (i && buf) {
504  ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
505  if (ret < 0) {
506  av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
507  return ret;
508  }
509  if (coded_samples != s->packet.frame_count * s->packet.frame_duration) {
510  av_log(avctx, AV_LOG_ERROR,
511  "Mismatching coded sample count in substream %d.\n", i);
512  return AVERROR_INVALIDDATA;
513  }
514 
516  }
517 
518  ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
519  c->out + 2 * i, c->out_size[i], coded_samples);
520  if (ret < 0)
521  return ret;
522  c->decoded_samples[i] = ret;
523  decoded_samples = FFMIN(decoded_samples, ret);
524 
525  buf += s->packet.packet_size;
526  buf_size -= s->packet.packet_size;
527  }
528 
529  /* buffer the extra samples */
530  for (i = 0; i < c->nb_streams; i++) {
531  int buffer_samples = c->decoded_samples[i] - decoded_samples;
532  if (buffer_samples) {
533  float *buf[2] = { c->out[2 * i + 0] ? c->out[2 * i + 0] : (float*)frame->extended_data[0],
534  c->out[2 * i + 1] ? c->out[2 * i + 1] : (float*)frame->extended_data[0] };
535  buf[0] += decoded_samples;
536  buf[1] += decoded_samples;
537  ret = av_audio_fifo_write(c->sync_buffers[i], (void**)buf, buffer_samples);
538  if (ret < 0)
539  return ret;
540  }
541  }
542 
543  for (i = 0; i < avctx->channels; i++) {
544  ChannelMap *map = &c->channel_maps[i];
545 
546  /* handle copied channels */
547  if (map->copy) {
548  memcpy(frame->extended_data[i],
549  frame->extended_data[map->copy_idx],
550  frame->linesize[0]);
551  } else if (map->silence) {
552  memset(frame->extended_data[i], 0, frame->linesize[0]);
553  }
554 
555  if (c->gain_i && decoded_samples > 0) {
556  c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i],
557  (float*)frame->extended_data[i],
558  c->gain, FFALIGN(decoded_samples, 8));
559  }
560  }
561 
562  frame->nb_samples = decoded_samples;
563  *got_frame_ptr = !!decoded_samples;
564 
565  return avpkt->size;
566 }
567 
569 {
570  OpusContext *c = ctx->priv_data;
571  int i;
572 
573  for (i = 0; i < c->nb_streams; i++) {
574  OpusStreamContext *s = &c->streams[i];
575 
576  memset(&s->packet, 0, sizeof(s->packet));
577  s->delayed_samples = 0;
578 
579  if (s->celt_delay)
581  swr_close(s->swr);
582 
584 
585  ff_silk_flush(s->silk);
586  ff_celt_flush(s->celt);
587  }
588 }
589 
591 {
592  OpusContext *c = avctx->priv_data;
593  int i;
594 
595  for (i = 0; i < c->nb_streams; i++) {
596  OpusStreamContext *s = &c->streams[i];
597 
598  ff_silk_free(&s->silk);
599  ff_celt_free(&s->celt);
600 
601  av_freep(&s->out_dummy);
603 
605  swr_free(&s->swr);
606  }
607 
608  av_freep(&c->streams);
609 
610  if (c->sync_buffers) {
611  for (i = 0; i < c->nb_streams; i++)
613  }
614  av_freep(&c->sync_buffers);
616  av_freep(&c->out);
617  av_freep(&c->out_size);
618 
619  c->nb_streams = 0;
620 
621  av_freep(&c->channel_maps);
622  av_freep(&c->fdsp);
623 
624  return 0;
625 }
626 
628 {
629  OpusContext *c = avctx->priv_data;
630  int ret, i, j;
631 
633  avctx->sample_rate = 48000;
634 
636  if (!c->fdsp)
637  return AVERROR(ENOMEM);
638 
639  /* find out the channel configuration */
640  ret = ff_opus_parse_extradata(avctx, c);
641  if (ret < 0) {
642  av_freep(&c->fdsp);
643  return ret;
644  }
645 
646  /* allocate and init each independent decoder */
647  c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
648  c->out = av_mallocz_array(c->nb_streams, 2 * sizeof(*c->out));
649  c->out_size = av_mallocz_array(c->nb_streams, sizeof(*c->out_size));
652  if (!c->streams || !c->sync_buffers || !c->decoded_samples || !c->out || !c->out_size) {
653  c->nb_streams = 0;
654  ret = AVERROR(ENOMEM);
655  goto fail;
656  }
657 
658  for (i = 0; i < c->nb_streams; i++) {
659  OpusStreamContext *s = &c->streams[i];
660  uint64_t layout;
661 
662  s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
663 
664  s->avctx = avctx;
665 
666  for (j = 0; j < s->output_channels; j++) {
667  s->silk_output[j] = s->silk_buf[j];
668  s->celt_output[j] = s->celt_buf[j];
669  s->redundancy_output[j] = s->redundancy_buf[j];
670  }
671 
672  s->fdsp = c->fdsp;
673 
674  s->swr =swr_alloc();
675  if (!s->swr)
676  goto fail;
677 
679  av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
680  av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
681  av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
682  av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
683  av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
684  av_opt_set_int(s->swr, "filter_size", 16, 0);
685 
686  ret = ff_silk_init(avctx, &s->silk, s->output_channels);
687  if (ret < 0)
688  goto fail;
689 
690  ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv);
691  if (ret < 0)
692  goto fail;
693 
695  s->output_channels, 1024);
696  if (!s->celt_delay) {
697  ret = AVERROR(ENOMEM);
698  goto fail;
699  }
700 
702  s->output_channels, 32);
703  if (!c->sync_buffers[i]) {
704  ret = AVERROR(ENOMEM);
705  goto fail;
706  }
707  }
708 
709  return 0;
710 fail:
711  opus_decode_close(avctx);
712  return ret;
713 }
714 
715 #define OFFSET(x) offsetof(OpusContext, x)
716 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
717 static const AVOption opus_options[] = {
718  { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AD },
719  { NULL },
720 };
721 
722 static const AVClass opus_class = {
723  .class_name = "Opus Decoder",
724  .item_name = av_default_item_name,
725  .option = opus_options,
726  .version = LIBAVUTIL_VERSION_INT,
727 };
728 
730  .name = "opus",
731  .long_name = NULL_IF_CONFIG_SMALL("Opus"),
732  .priv_class = &opus_class,
733  .type = AVMEDIA_TYPE_AUDIO,
734  .id = AV_CODEC_ID_OPUS,
735  .priv_data_size = sizeof(OpusContext),
737  .close = opus_decode_close,
740  .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
741 };
int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size, int self_delimiting)
Parse Opus packet info from raw packet data.
Definition: opus.c:90
static av_cold int opus_decode_close(AVCodecContext *avctx)
Definition: opusdec.c:590
float, planar
Definition: samplefmt.h:69
#define NULL
Definition: coverity.c:32
int ff_celt_decode_frame(CeltFrame *f, OpusRangeCoder *rc, float **output, int channels, int frame_size, int start_band, int end_band)
Definition: opus_celt.c:679
const char * s
Definition: avisynth_c.h:768
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
av_cold void swr_close(SwrContext *s)
Closes the context so that swr_is_initialized() returns 0.
Definition: swresample.c:148
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
This structure describes decoded (raw) audio or video data.
Definition: frame.h:201
AVOption.
Definition: opt.h:246
AVAudioFifo ** sync_buffers
Definition: opus.h:162
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
static void flush(AVCodecContext *avctx)
static const uint16_t silk_frame_duration_ms[16]
Definition: opusdec.c:53
int frame_count
frame count
Definition: opus.h:92
int nb_stereo_streams
Definition: opus.h:167
float redundancy_buf[2][960]
Definition: opus.h:115
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static FFServerConfig config
Definition: ffserver.c:193
static const AVOption opus_options[]
Definition: opusdec.c:717
int output_channels
Definition: opus.h:102
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
int delayed_samples
Definition: opus.h:129
float gain
Definition: opus.h:171
static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
Definition: opusdec.c:153
int size
Definition: avcodec.h:1415
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:193
#define OFFSET(x)
Definition: opusdec.c:715
int out_size
Definition: movenc.c:55
uint32_t ff_opus_rc_dec_log(OpusRangeCoder *rc, uint32_t bits)
Definition: opus_rc.c:114
static AVPacket pkt
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:3365
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
int16_t gain_i
Definition: opus.h:170
Macro definitions for various function/variable attributes.
const uint8_t ff_celt_band_end[]
Definition: opustab.c:27
void(* vector_fmac_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float and add to destination vector.
Definition: float_dsp.h:54
int apply_phase_inv
Definition: opus.h:155
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:981
int * decoded_samples
Definition: opus.h:164
static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
Definition: opusdec.c:85
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2165
uint8_t
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:149
#define av_cold
Definition: attributes.h:82
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
AVOptions.
static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
Definition: opusdec.c:173
int copy
Definition: opus.h:144
SilkContext * silk
Definition: opus.h:106
static AVFrame * frame
static void finish(void)
Definition: movenc.c:345
uint8_t * data
Definition: avcodec.h:1414
bitstream reader API header.
static void opus_fade(float *out, const float *in1, const float *in2, const float *window, int len)
Definition: opusdec.c:76
ptrdiff_t size
Definition: opengl_enc.c:101
float * silk_output[2]
Definition: opus.h:111
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
static av_cold int opus_decode_init(AVCodecContext *avctx)
Definition: opusdec.c:627
AVFloatDSPContext * fdsp
Definition: opus.h:108
ChannelMap * channel_maps
Definition: opus.h:173
libswresample public header
int nb_streams
Definition: opus.h:166
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:127
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void ff_celt_flush(CeltFrame *f)
Definition: opus_celt.c:854
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:558
AVFloatDSPContext * fdsp
Definition: opus.h:169
const char * name
Name of the codec implementation.
Definition: avcodec.h:3372
#define FFMAX(a, b)
Definition: common.h:94
#define fail()
Definition: checkasm.h:113
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
Definition: mem.c:483
static SDL_Window * window
Definition: ffplay.c:362
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
audio channel layout utility functions
float * out[2]
Definition: opus.h:119
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:2626
int frame_size[MAX_FRAMES]
frame sizes
Definition: opus.h:94
#define FFMIN(a, b)
Definition: common.h:96
int frame_duration
frame duration, in samples @ 48kHz
Definition: opus.h:95
float celt_buf[2][960]
Definition: opus.h:112
SwrContext * swr
Definition: opus.h:125
void ff_celt_free(CeltFrame **f)
Definition: opus_celt.c:881
AVFormatContext * ctx
Definition: movenc.c:48
int out_dummy_allocated_size
Definition: opus.h:123
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:85
float silk_buf[2][960]
Definition: opus.h:110
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:2637
static const AVClass opus_class
Definition: opusdec.c:722
#define FF_ARRAY_ELEMS(a)
int silence
Definition: opus.h:149
int av_opt_get_int(void *obj, const char *name, int search_flags, int64_t *out_val)
Definition: opt.c:875
static int get_silk_samplerate(int config)
Definition: opusdec.c:67
CeltFrame * celt
Definition: opus.h:107
float * out_dummy
Definition: opus.h:122
Libavcodec external API header.
OpusPacket packet
Definition: opus.h:131
int sample_rate
samples per second
Definition: avcodec.h:2157
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:232
AVCodec ff_opus_decoder
Definition: opusdec.c:729
void ff_silk_flush(SilkContext *s)
Definition: opus_silk.c:850
main external API structure.
Definition: avcodec.h:1502
const float ff_celt_window2[120]
Definition: opustab.c:1133
int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels)
Definition: opus_silk.c:858
static av_always_inline uint32_t opus_rc_tell(const OpusRangeCoder *rc)
CELT: estimate bits of entropy that have thus far been consumed for the current CELT frame...
Definition: opus_rc.h:61
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:137
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1886
void * buf
Definition: avisynth_c.h:690
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
int config
configuration: tells the audio mode, bandwidth, and frame duration
Definition: opus.h:90
void ff_silk_free(SilkContext **ps)
Definition: opus_silk.c:845
Describe the class of an AVClass context structure.
Definition: log.h:67
enum OpusMode mode
mode
Definition: opus.h:96
void ff_opus_rc_dec_raw_init(OpusRangeCoder *rc, const uint8_t *rightend, uint32_t bytes)
Definition: opus_rc.c:352
int copy_idx
Definition: opus.h:146
const VDPAUPixFmtMap * map
int stereo
whether this packet is mono or stereo
Definition: opus.h:88
AVCodecContext * avctx
Definition: opus.h:101
float ** out
Definition: opus.h:158
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg[SWR_CH_MAX], int in_count)
Definition: swresample.c:706
int data_size
size of the useful data – packet size - padding
Definition: opus.h:86
int channel_idx
Definition: opus.h:139
int redundancy_idx
Definition: opus.h:133
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
static int opus_decode_packet(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: opusdec.c:414
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
Definition: audio_fifo.c:201
static int opus_init_resample(OpusStreamContext *s)
Definition: opusdec.c:128
static int opus_decode_subpacket(OpusStreamContext *s, const uint8_t *buf, int buf_size, float **out, int out_size, int nb_samples)
Definition: opusdec.c:332
float * celt_output[2]
Definition: opus.h:113
common internal api header.
OpusRangeCoder rc
Definition: opus.h:104
int stream_idx
Definition: opus.h:138
int * out_size
Definition: opus.h:159
static double c[64]
static const int silk_resample_delay[]
Definition: opusdec.c:63
int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc, float *output[2], enum OpusBandwidth bandwidth, int coded_channels, int duration_ms)
Decode the LP layer of one Opus frame (which may correspond to several SILK frames).
Definition: opus_silk.c:781
OpusStreamContext * streams
Definition: opus.h:154
int packet_size
packet size
Definition: opus.h:85
OpusRangeCoder redundancy_rc
Definition: opus.h:105
void * priv_data
Definition: avcodec.h:1529
int ff_opus_rc_dec_init(OpusRangeCoder *rc, const uint8_t *data, int size)
Definition: opus_rc.c:338
Audio FIFO Buffer.
int len
int channels
number of audio channels
Definition: avcodec.h:2158
int frame_offset[MAX_FRAMES]
frame offsets
Definition: opus.h:93
enum OpusBandwidth bandwidth
bandwidth
Definition: opus.h:97
static av_cold void opus_decode_flush(AVCodecContext *ctx)
Definition: opusdec.c:568
float * redundancy_output[2]
Definition: opus.h:116
uint64_t layout
uint32_t ff_opus_rc_dec_uint(OpusRangeCoder *rc, uint32_t size)
CELT: read a uniform distribution.
Definition: opus_rc.c:182
AVAudioFifo * celt_delay
Definition: opus.h:126
av_cold int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s)
Definition: opus.c:291
FILE * out
Definition: movenc.c:54
#define av_freep(p)
int silk_samplerate
Definition: opus.h:127
#define AD
Definition: opusdec.c:716
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
#define AV_CH_LAYOUT_MONO
int swr_is_initialized(struct SwrContext *s)
Check whether an swr context has been initialized or not.
Definition: swresample.c:702
This structure stores compressed data.
Definition: avcodec.h:1391
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:956
int ff_celt_init(AVCodecContext *avctx, CeltFrame **f, int output_channels, int apply_phase_inv)
Definition: opus_celt.c:898
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:152
void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.c:191