FFmpeg  4.4.5
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->cur_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->cur_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->cur_out[i], s->cur_out[i],
116  s->redundancy_output[i] + 120 + s->redundancy_idx,
117  ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
118  s->redundancy_idx = 0;
119  }
120 
121  s->cur_out[0] += nb_samples;
122  s->cur_out[1] += nb_samples;
123  s->remaining_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) {
145  av_log(s->avctx, AV_LOG_ERROR,
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 
160  ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
161  s->redundancy_output,
162  s->packet.stereo + 1, 240,
163  0, ff_celt_band_end[s->packet.bandwidth]);
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,
194  FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
195  s->packet.stereo + 1,
196  silk_frame_duration_ms[s->packet.config]);
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->cur_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->cur_out[0], s->cur_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 {
260  av_log(s->avctx, AV_LOG_WARNING,
261  "Spurious CELT delay samples present.\n");
262  av_audio_fifo_drain(s->celt_delay, delay_samples);
263  if (s->avctx->err_recognition & AV_EF_EXPLODE)
264  return AVERROR_BUG;
265  }
266  }
267 
269 
270  ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
271  s->packet.stereo + 1,
272  s->packet.frame_duration,
273  (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
274  ff_celt_band_end[s->packet.bandwidth]);
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->cur_out[i], s->cur_out[i],
299  s->redundancy_output[i] + 120 + s->redundancy_idx,
300  ff_celt_window2 + s->redundancy_idx, 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->cur_out[i] + samples - 120 + delayed_samples,
312  s->cur_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->cur_out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
321  opus_fade(s->cur_out[i] + 120 + delayed_samples,
322  s->redundancy_output[i] + 120,
323  s->cur_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  int nb_samples)
335 {
336  int output_samples = 0;
337  int flush_needed = 0;
338  int i, j, ret;
339 
340  s->cur_out[0] = s->out[0];
341  s->cur_out[1] = s->out[1];
342  s->remaining_out_size = s->out_size;
343 
344  /* check if we need to flush the resampler */
345  if (swr_is_initialized(s->swr)) {
346  if (buf) {
347  int64_t cur_samplerate;
348  av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
349  flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
350  } else {
351  flush_needed = !!s->delayed_samples;
352  }
353  }
354 
355  if (!buf && !flush_needed)
356  return 0;
357 
358  /* use dummy output buffers if the channel is not mapped to anything */
359  if (!s->cur_out[0] ||
360  (s->output_channels == 2 && !s->cur_out[1])) {
361  av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size,
362  s->remaining_out_size);
363  if (!s->out_dummy)
364  return AVERROR(ENOMEM);
365  if (!s->cur_out[0])
366  s->cur_out[0] = s->out_dummy;
367  if (!s->cur_out[1])
368  s->cur_out[1] = s->out_dummy;
369  }
370 
371  /* flush the resampler if necessary */
372  if (flush_needed) {
373  ret = opus_flush_resample(s, s->delayed_samples);
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");
393  if (s->avctx->err_recognition & AV_EF_EXPLODE)
394  return samples;
395 
396  for (j = 0; j < s->output_channels; j++)
397  memset(s->cur_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->cur_out[j] += samples;
404  s->remaining_out_size -= samples * sizeof(float);
405  }
406 
407 finish:
408  s->cur_out[0] = s->cur_out[1] = NULL;
409  s->remaining_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,
432  s->delayed_samples + av_audio_fifo_size(s->sync_buffer));
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;
444  c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
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  for (i = 0; i < avctx->channels; i++) {
462  ChannelMap *map = &c->channel_maps[i];
463  if (!map->copy)
464  c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
465  }
466 
467  /* read the data from the sync buffers */
468  for (i = 0; i < c->nb_streams; i++) {
469  OpusStreamContext *s = &c->streams[i];
470  float **out = s->out;
471  int sync_size = av_audio_fifo_size(s->sync_buffer);
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(s->sync_buffer, (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  s->out_size = 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 
515  s->silk_samplerate = get_silk_samplerate(s->packet.config);
516  }
517 
518  ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
519  coded_samples);
520  if (ret < 0)
521  return ret;
522  s->decoded_samples = 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  OpusStreamContext *s = &c->streams[i];
532  int buffer_samples = s->decoded_samples - decoded_samples;
533  if (buffer_samples) {
534  float *buf[2] = { s->out[0] ? s->out[0] : (float*)frame->extended_data[0],
535  s->out[1] ? s->out[1] : (float*)frame->extended_data[0] };
536  buf[0] += decoded_samples;
537  buf[1] += decoded_samples;
538  ret = av_audio_fifo_write(s->sync_buffer, (void**)buf, buffer_samples);
539  if (ret < 0)
540  return ret;
541  }
542  }
543 
544  for (i = 0; i < avctx->channels; i++) {
545  ChannelMap *map = &c->channel_maps[i];
546 
547  /* handle copied channels */
548  if (map->copy) {
549  memcpy(frame->extended_data[i],
550  frame->extended_data[map->copy_idx],
551  frame->linesize[0]);
552  } else if (map->silence) {
553  memset(frame->extended_data[i], 0, frame->linesize[0]);
554  }
555 
556  if (c->gain_i && decoded_samples > 0) {
557  c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i],
558  (float*)frame->extended_data[i],
559  c->gain, FFALIGN(decoded_samples, 8));
560  }
561  }
562 
563  frame->nb_samples = decoded_samples;
564  *got_frame_ptr = !!decoded_samples;
565 
566  return avpkt->size;
567 }
568 
570 {
572  int i;
573 
574  for (i = 0; i < c->nb_streams; i++) {
575  OpusStreamContext *s = &c->streams[i];
576 
577  memset(&s->packet, 0, sizeof(s->packet));
578  s->delayed_samples = 0;
579 
580  av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
581  swr_close(s->swr);
582 
583  av_audio_fifo_drain(s->sync_buffer, av_audio_fifo_size(s->sync_buffer));
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);
602  s->out_dummy_allocated_size = 0;
603 
604  av_audio_fifo_free(s->sync_buffer);
605  av_audio_fifo_free(s->celt_delay);
606  swr_free(&s->swr);
607  }
608 
609  av_freep(&c->streams);
610 
611  c->nb_streams = 0;
612 
613  av_freep(&c->channel_maps);
614  av_freep(&c->fdsp);
615 
616  return 0;
617 }
618 
620 {
621  OpusContext *c = avctx->priv_data;
622  int ret, i, j;
623 
625  avctx->sample_rate = 48000;
626 
627  c->fdsp = avpriv_float_dsp_alloc(0);
628  if (!c->fdsp)
629  return AVERROR(ENOMEM);
630 
631  /* find out the channel configuration */
632  ret = ff_opus_parse_extradata(avctx, c);
633  if (ret < 0)
634  return ret;
635 
636  /* allocate and init each independent decoder */
637  c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
638  if (!c->streams) {
639  c->nb_streams = 0;
640  return AVERROR(ENOMEM);
641  }
642 
643  for (i = 0; i < c->nb_streams; i++) {
644  OpusStreamContext *s = &c->streams[i];
645  uint64_t layout;
646 
647  s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
648 
649  s->avctx = avctx;
650 
651  for (j = 0; j < s->output_channels; j++) {
652  s->silk_output[j] = s->silk_buf[j];
653  s->celt_output[j] = s->celt_buf[j];
654  s->redundancy_output[j] = s->redundancy_buf[j];
655  }
656 
657  s->fdsp = c->fdsp;
658 
659  s->swr =swr_alloc();
660  if (!s->swr)
661  return AVERROR(ENOMEM);
662 
663  layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
664  av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
665  av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
666  av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
667  av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
668  av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
669  av_opt_set_int(s->swr, "filter_size", 16, 0);
670 
671  ret = ff_silk_init(avctx, &s->silk, s->output_channels);
672  if (ret < 0)
673  return ret;
674 
675  ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv);
676  if (ret < 0)
677  return ret;
678 
679  s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
680  s->output_channels, 1024);
681  if (!s->celt_delay)
682  return AVERROR(ENOMEM);
683 
684  s->sync_buffer = av_audio_fifo_alloc(avctx->sample_fmt,
685  s->output_channels, 32);
686  if (!s->sync_buffer)
687  return AVERROR(ENOMEM);
688  }
689 
690  return 0;
691 }
692 
693 #define OFFSET(x) offsetof(OpusContext, x)
694 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
695 static const AVOption opus_options[] = {
696  { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AD },
697  { NULL },
698 };
699 
700 static const AVClass opus_class = {
701  .class_name = "Opus Decoder",
702  .item_name = av_default_item_name,
703  .option = opus_options,
704  .version = LIBAVUTIL_VERSION_INT,
705 };
706 
708  .name = "opus",
709  .long_name = NULL_IF_CONFIG_SMALL("Opus"),
710  .priv_class = &opus_class,
711  .type = AVMEDIA_TYPE_AUDIO,
712  .id = AV_CODEC_ID_OPUS,
713  .priv_data_size = sizeof(OpusContext),
715  .close = opus_decode_close,
720 };
static void flush(AVCodecContext *avctx)
Macro definitions for various function/variable attributes.
#define av_cold
Definition: attributes.h:88
uint8_t
Audio FIFO Buffer.
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
Libavcodec external API header.
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:1656
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
#define s(width, name)
Definition: cbs_vp9.c:257
uint64_t layout
audio channel layout utility functions
#define fail()
Definition: checkasm.h:133
#define FFMIN(a, b)
Definition: common.h:105
#define FFMAX(a, b)
Definition: common.h:103
#define NULL
Definition: coverity.c:32
long long int64_t
Definition: coverity.c:34
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static AVFrame * frame
static SDL_Window * window
Definition: ffplay.c:366
bitstream reader API header.
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_STEREO
#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: codec.h:77
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:104
@ AV_CODEC_ID_OPUS
Definition: codec_id.h:484
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
Definition: audio_fifo.c:201
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
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:502
void * av_mallocz_array(size_t nmemb, size_t size)
Allocate a memory block for an array with av_mallocz().
Definition: mem.c:190
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:137
av_cold void swr_close(SwrContext *s)
Closes the context so that swr_is_initialized() returns 0.
Definition: swresample.c:148
int swr_is_initialized(struct SwrContext *s)
Check whether an swr context has been initialized or not.
Definition: swresample.c:710
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:152
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:149
int av_opt_get_int(void *obj, const char *name, int search_flags, int64_t *out_val)
Definition: opt.c:912
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:586
const VDPAUPixFmtMap * map
int i
Definition: input.c:407
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:41
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: internal.h:49
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define FFALIGN(x, a)
Definition: macros.h:48
const char data[16]
Definition: mxf.c:142
AVOptions.
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:91
av_cold int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s)
Definition: opus.c:292
@ OPUS_BANDWIDTH_WIDEBAND
Definition: opus.h:74
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:786
@ OPUS_MODE_SILK
Definition: opus.h:64
@ OPUS_MODE_HYBRID
Definition: opus.h:65
@ OPUS_MODE_CELT
Definition: opus.h:66
void ff_silk_flush(SilkContext *s)
Definition: opus_silk.c:867
void ff_silk_free(SilkContext **ps)
Definition: opus_silk.c:862
int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels)
Definition: opus_silk.c:875
void ff_celt_free(CeltFrame **f)
Definition: opus_celt.c:521
int ff_celt_init(AVCodecContext *avctx, CeltFrame **f, int output_channels, int apply_phase_inv)
Definition: opus_celt.c:538
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:320
void ff_celt_flush(CeltFrame *f)
Definition: opus_celt.c:490
uint32_t ff_opus_rc_dec_uint(OpusRangeCoder *rc, uint32_t size)
CELT: read a uniform distribution.
Definition: opus_rc.c:182
void ff_opus_rc_dec_raw_init(OpusRangeCoder *rc, const uint8_t *rightend, uint32_t bytes)
Definition: opus_rc.c:352
int ff_opus_rc_dec_init(OpusRangeCoder *rc, const uint8_t *data, int size)
Definition: opus_rc.c:338
uint32_t ff_opus_rc_dec_log(OpusRangeCoder *rc, uint32_t bits)
Definition: opus_rc.c:114
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
static int opus_decode_packet(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: opusdec.c:414
static int get_silk_samplerate(int config)
Definition: opusdec.c:67
static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
Definition: opusdec.c:153
static const AVClass opus_class
Definition: opusdec.c:700
static const int silk_resample_delay[]
Definition: opusdec.c:63
static const AVOption opus_options[]
Definition: opusdec.c:695
static int opus_init_resample(OpusStreamContext *s)
Definition: opusdec.c:128
static const uint16_t silk_frame_duration_ms[16]
Definition: opusdec.c:53
static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
Definition: opusdec.c:85
static av_cold int opus_decode_init(AVCodecContext *avctx)
Definition: opusdec.c:619
static void opus_fade(float *out, const float *in1, const float *in2, const float *window, int len)
Definition: opusdec.c:76
#define AD
Definition: opusdec.c:694
AVCodec ff_opus_decoder
Definition: opusdec.c:707
static av_cold void opus_decode_flush(AVCodecContext *ctx)
Definition: opusdec.c:569
static av_cold int opus_decode_close(AVCodecContext *avctx)
Definition: opusdec.c:590
#define OFFSET(x)
Definition: opusdec.c:693
static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
Definition: opusdec.c:173
static int opus_decode_subpacket(OpusStreamContext *s, const uint8_t *buf, int buf_size, int nb_samples)
Definition: opusdec.c:332
const uint8_t ff_celt_band_end[]
Definition: opustab.c:29
const float ff_celt_window2[120]
Definition: opustab.c:1138
#define FF_ARRAY_ELEMS(a)
Describe the class of an AVClass context structure.
Definition: log.h:67
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
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int sample_rate
samples per second
Definition: avcodec.h:1196
int channels
number of audio channels
Definition: avcodec.h:1197
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
void * priv_data
Format private data.
Definition: avformat.h:1260
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:349
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
AVOption.
Definition: opt.h:248
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
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:714
libswresample public header
#define av_freep(p)
#define av_log(a,...)
FILE * out
Definition: movenc.c:54
AVPacket * pkt
Definition: movenc.c:59
AVFormatContext * ctx
Definition: movenc.c:48
static void finish(void)
Definition: movenc.c:342
int size
int len
static double c[64]