FFmpeg
libopusenc.c
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1 /*
2  * Opus encoder using libopus
3  * Copyright (c) 2012 Nathan Caldwell
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include <opus.h>
23 #include <opus_multistream.h>
24 
26 #include "libavutil/opt.h"
27 #include "avcodec.h"
28 #include "bytestream.h"
29 #include "encode.h"
30 #include "internal.h"
31 #include "libopus.h"
32 #include "vorbis.h"
33 #include "audio_frame_queue.h"
34 
35 typedef struct LibopusEncOpts {
36  int vbr;
39  int fec;
45 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
46  int apply_phase_inv;
47 #endif
49 
50 typedef struct LibopusEncContext {
51  AVClass *class;
52  OpusMSEncoder *enc;
54  uint8_t *samples;
57  const uint8_t *encoder_channel_map;
59 
60 static const uint8_t opus_coupled_streams[8] = {
61  0, 1, 1, 2, 2, 2, 2, 3
62 };
63 
64 /* Opus internal to Vorbis channel order mapping written in the header */
65 static const uint8_t opus_vorbis_channel_map[8][8] = {
66  { 0 },
67  { 0, 1 },
68  { 0, 2, 1 },
69  { 0, 1, 2, 3 },
70  { 0, 4, 1, 2, 3 },
71  { 0, 4, 1, 2, 3, 5 },
72  { 0, 4, 1, 2, 3, 5, 6 },
73  { 0, 6, 1, 2, 3, 4, 5, 7 },
74 };
75 
76 /* libavcodec to libopus channel order mapping, passed to libopus */
77 static const uint8_t libavcodec_libopus_channel_map[8][8] = {
78  { 0 },
79  { 0, 1 },
80  { 0, 1, 2 },
81  { 0, 1, 2, 3 },
82  { 0, 1, 3, 4, 2 },
83  { 0, 1, 4, 5, 2, 3 },
84  { 0, 1, 5, 6, 2, 4, 3 },
85  { 0, 1, 6, 7, 4, 5, 2, 3 },
86 };
87 
88 static void libopus_write_header(AVCodecContext *avctx, int stream_count,
89  int coupled_stream_count,
90  int mapping_family,
91  const uint8_t *channel_mapping)
92 {
93  uint8_t *p = avctx->extradata;
94  int channels = avctx->channels;
95 
96  bytestream_put_buffer(&p, "OpusHead", 8);
97  bytestream_put_byte(&p, 1); /* Version */
98  bytestream_put_byte(&p, channels);
99  bytestream_put_le16(&p, avctx->initial_padding * 48000 / avctx->sample_rate); /* Lookahead samples at 48kHz */
100  bytestream_put_le32(&p, avctx->sample_rate); /* Original sample rate */
101  bytestream_put_le16(&p, 0); /* Gain of 0dB is recommended. */
102 
103  /* Channel mapping */
104  bytestream_put_byte(&p, mapping_family);
105  if (mapping_family != 0) {
106  bytestream_put_byte(&p, stream_count);
107  bytestream_put_byte(&p, coupled_stream_count);
108  bytestream_put_buffer(&p, channel_mapping, channels);
109  }
110 }
111 
112 static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc,
114 {
115  int ret;
116 
117  if (avctx->global_quality) {
118  av_log(avctx, AV_LOG_ERROR,
119  "Quality-based encoding not supported, "
120  "please specify a bitrate and VBR setting.\n");
121  return AVERROR(EINVAL);
122  }
123 
124  ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->bit_rate));
125  if (ret != OPUS_OK) {
126  av_log(avctx, AV_LOG_ERROR,
127  "Failed to set bitrate: %s\n", opus_strerror(ret));
128  return ret;
129  }
130 
131  ret = opus_multistream_encoder_ctl(enc,
132  OPUS_SET_COMPLEXITY(opts->complexity));
133  if (ret != OPUS_OK)
134  av_log(avctx, AV_LOG_WARNING,
135  "Unable to set complexity: %s\n", opus_strerror(ret));
136 
137  ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->vbr));
138  if (ret != OPUS_OK)
139  av_log(avctx, AV_LOG_WARNING,
140  "Unable to set VBR: %s\n", opus_strerror(ret));
141 
142  ret = opus_multistream_encoder_ctl(enc,
143  OPUS_SET_VBR_CONSTRAINT(opts->vbr == 2));
144  if (ret != OPUS_OK)
145  av_log(avctx, AV_LOG_WARNING,
146  "Unable to set constrained VBR: %s\n", opus_strerror(ret));
147 
148  ret = opus_multistream_encoder_ctl(enc,
149  OPUS_SET_PACKET_LOSS_PERC(opts->packet_loss));
150  if (ret != OPUS_OK)
151  av_log(avctx, AV_LOG_WARNING,
152  "Unable to set expected packet loss percentage: %s\n",
153  opus_strerror(ret));
154 
155  ret = opus_multistream_encoder_ctl(enc,
156  OPUS_SET_INBAND_FEC(opts->fec));
157  if (ret != OPUS_OK)
158  av_log(avctx, AV_LOG_WARNING,
159  "Unable to set inband FEC: %s\n",
160  opus_strerror(ret));
161 
162  if (avctx->cutoff) {
163  ret = opus_multistream_encoder_ctl(enc,
164  OPUS_SET_MAX_BANDWIDTH(opts->max_bandwidth));
165  if (ret != OPUS_OK)
166  av_log(avctx, AV_LOG_WARNING,
167  "Unable to set maximum bandwidth: %s\n", opus_strerror(ret));
168  }
169 
170 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
171  ret = opus_multistream_encoder_ctl(enc,
172  OPUS_SET_PHASE_INVERSION_DISABLED(!opts->apply_phase_inv));
173  if (ret != OPUS_OK)
174  av_log(avctx, AV_LOG_WARNING,
175  "Unable to set phase inversion: %s\n",
176  opus_strerror(ret));
177 #endif
178  return OPUS_OK;
179 }
180 
182  int max_channels) {
183  if (avctx->channels > max_channels) {
184  av_log(avctx, AV_LOG_ERROR, "Opus mapping family undefined for %d channels.\n",
185  avctx->channels);
186  return AVERROR(EINVAL);
187  }
188 
189  return 0;
190 }
191 
192 static int libopus_check_vorbis_layout(AVCodecContext *avctx, int mapping_family) {
194 
195  if (!avctx->channel_layout) {
196  av_log(avctx, AV_LOG_WARNING,
197  "No channel layout specified. Opus encoder will use Vorbis "
198  "channel layout for %d channels.\n", avctx->channels);
199  } else if (avctx->channel_layout != ff_vorbis_channel_layouts[avctx->channels - 1]) {
200  char name[32];
202  avctx->channel_layout);
203  av_log(avctx, AV_LOG_ERROR,
204  "Invalid channel layout %s for specified mapping family %d.\n",
205  name, mapping_family);
206 
207  return AVERROR(EINVAL);
208  }
209 
210  return 0;
211 }
212 
214  AVCodecContext *avctx,
215  int mapping_family,
216  const uint8_t ** channel_map_result)
217 {
218  const uint8_t * channel_map = NULL;
219  int ret;
220 
221  switch (mapping_family) {
222  case -1:
223  ret = libopus_check_max_channels(avctx, 8);
224  if (ret == 0) {
225  ret = libopus_check_vorbis_layout(avctx, mapping_family);
226  /* Channels do not need to be reordered. */
227  }
228 
229  break;
230  case 0:
231  ret = libopus_check_max_channels(avctx, 2);
232  if (ret == 0) {
233  ret = libopus_check_vorbis_layout(avctx, mapping_family);
234  }
235  break;
236  case 1:
237  /* Opus expects channels to be in Vorbis order. */
238  ret = libopus_check_max_channels(avctx, 8);
239  if (ret == 0) {
240  ret = libopus_check_vorbis_layout(avctx, mapping_family);
241  channel_map = ff_vorbis_channel_layout_offsets[avctx->channels - 1];
242  }
243  break;
244  case 255:
245  ret = libopus_check_max_channels(avctx, 254);
246  break;
247  default:
248  av_log(avctx, AV_LOG_WARNING,
249  "Unknown channel mapping family %d. Output channel layout may be invalid.\n",
250  mapping_family);
251  ret = 0;
252  }
253 
254  *channel_map_result = channel_map;
255  return ret;
256 }
257 
259 {
260  LibopusEncContext *opus = avctx->priv_data;
261  OpusMSEncoder *enc;
262  uint8_t libopus_channel_mapping[255];
263  int ret = OPUS_OK;
264  int av_ret;
265  int coupled_stream_count, header_size, frame_size;
266  int mapping_family;
267 
268  frame_size = opus->opts.frame_duration * 48000 / 1000;
269  switch (frame_size) {
270  case 120:
271  case 240:
272  if (opus->opts.application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
273  av_log(avctx, AV_LOG_WARNING,
274  "LPC mode cannot be used with a frame duration of less "
275  "than 10ms. Enabling restricted low-delay mode.\n"
276  "Use a longer frame duration if this is not what you want.\n");
277  /* Frame sizes less than 10 ms can only use MDCT mode, so switching to
278  * RESTRICTED_LOWDELAY avoids an unnecessary extra 2.5ms lookahead. */
279  opus->opts.application = OPUS_APPLICATION_RESTRICTED_LOWDELAY;
280  case 480:
281  case 960:
282  case 1920:
283  case 2880:
284 #ifdef OPUS_FRAMESIZE_120_MS
285  case 3840:
286  case 4800:
287  case 5760:
288 #endif
289  opus->opts.packet_size =
290  avctx->frame_size = frame_size * avctx->sample_rate / 48000;
291  break;
292  default:
293  av_log(avctx, AV_LOG_ERROR, "Invalid frame duration: %g.\n"
294  "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40"
295 #ifdef OPUS_FRAMESIZE_120_MS
296  ", 60, 80, 100 or 120.\n",
297 #else
298  " or 60.\n",
299 #endif
300  opus->opts.frame_duration);
301  return AVERROR(EINVAL);
302  }
303 
304  if (avctx->compression_level < 0 || avctx->compression_level > 10) {
305  av_log(avctx, AV_LOG_WARNING,
306  "Compression level must be in the range 0 to 10. "
307  "Defaulting to 10.\n");
308  opus->opts.complexity = 10;
309  } else {
310  opus->opts.complexity = avctx->compression_level;
311  }
312 
313  if (avctx->cutoff) {
314  switch (avctx->cutoff) {
315  case 4000:
317  break;
318  case 6000:
320  break;
321  case 8000:
323  break;
324  case 12000:
326  break;
327  case 20000:
329  break;
330  default:
331  av_log(avctx, AV_LOG_WARNING,
332  "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n"
333  "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",
334  avctx->cutoff);
335  avctx->cutoff = 0;
336  }
337  }
338 
339  /* Channels may need to be reordered to match opus mapping. */
341  &opus->encoder_channel_map);
342  if (av_ret) {
343  return av_ret;
344  }
345 
346  if (opus->opts.mapping_family == -1) {
347  /* By default, use mapping family 1 for the header but use the older
348  * libopus multistream API to avoid surround masking. */
349 
350  /* Set the mapping family so that the value is correct in the header */
351  mapping_family = avctx->channels > 2 ? 1 : 0;
352  coupled_stream_count = opus_coupled_streams[avctx->channels - 1];
353  opus->stream_count = avctx->channels - coupled_stream_count;
354  memcpy(libopus_channel_mapping,
355  opus_vorbis_channel_map[avctx->channels - 1],
356  avctx->channels * sizeof(*libopus_channel_mapping));
357 
358  enc = opus_multistream_encoder_create(
359  avctx->sample_rate, avctx->channels, opus->stream_count,
360  coupled_stream_count,
362  opus->opts.application, &ret);
363  } else {
364  /* Use the newer multistream API. The encoder will set the channel
365  * mapping and coupled stream counts to its internal defaults and will
366  * use surround masking analysis to save bits. */
367  mapping_family = opus->opts.mapping_family;
368  enc = opus_multistream_surround_encoder_create(
369  avctx->sample_rate, avctx->channels, mapping_family,
370  &opus->stream_count, &coupled_stream_count, libopus_channel_mapping,
371  opus->opts.application, &ret);
372  }
373 
374  if (ret != OPUS_OK) {
375  av_log(avctx, AV_LOG_ERROR,
376  "Failed to create encoder: %s\n", opus_strerror(ret));
378  }
379 
380  if (!avctx->bit_rate) {
381  /* Sane default copied from opusenc */
382  avctx->bit_rate = 64000 * opus->stream_count +
383  32000 * coupled_stream_count;
384  av_log(avctx, AV_LOG_WARNING,
385  "No bit rate set. Defaulting to %"PRId64" bps.\n", avctx->bit_rate);
386  }
387 
388  if (avctx->bit_rate < 500 || avctx->bit_rate > 256000 * avctx->channels) {
389  av_log(avctx, AV_LOG_ERROR, "The bit rate %"PRId64" bps is unsupported. "
390  "Please choose a value between 500 and %d.\n", avctx->bit_rate,
391  256000 * avctx->channels);
392  ret = AVERROR(EINVAL);
393  goto fail;
394  }
395 
396  ret = libopus_configure_encoder(avctx, enc, &opus->opts);
397  if (ret != OPUS_OK) {
399  goto fail;
400  }
401 
402  /* Header includes channel mapping table if and only if mapping family is NOT 0 */
403  header_size = 19 + (mapping_family == 0 ? 0 : 2 + avctx->channels);
404  avctx->extradata = av_malloc(header_size + AV_INPUT_BUFFER_PADDING_SIZE);
405  if (!avctx->extradata) {
406  av_log(avctx, AV_LOG_ERROR, "Failed to allocate extradata.\n");
407  ret = AVERROR(ENOMEM);
408  goto fail;
409  }
410  avctx->extradata_size = header_size;
411 
412  opus->samples = av_calloc(frame_size, avctx->channels *
414  if (!opus->samples) {
415  av_log(avctx, AV_LOG_ERROR, "Failed to allocate samples buffer.\n");
416  ret = AVERROR(ENOMEM);
417  goto fail;
418  }
419 
420  ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->initial_padding));
421  if (ret != OPUS_OK)
422  av_log(avctx, AV_LOG_WARNING,
423  "Unable to get number of lookahead samples: %s\n",
424  opus_strerror(ret));
425 
426  libopus_write_header(avctx, opus->stream_count, coupled_stream_count,
427  mapping_family, libopus_channel_mapping);
428 
429  ff_af_queue_init(avctx, &opus->afq);
430 
431  opus->enc = enc;
432 
433  return 0;
434 
435 fail:
436  opus_multistream_encoder_destroy(enc);
437  return ret;
438 }
439 
441  uint8_t *dst, const uint8_t *src, const uint8_t *channel_map,
442  int nb_channels, int nb_samples, int bytes_per_sample) {
443  int sample, channel;
444  for (sample = 0; sample < nb_samples; ++sample) {
445  for (channel = 0; channel < nb_channels; ++channel) {
446  const size_t src_pos = bytes_per_sample * (nb_channels * sample + channel);
447  const size_t dst_pos = bytes_per_sample * (nb_channels * sample + channel_map[channel]);
448 
449  memcpy(&dst[dst_pos], &src[src_pos], bytes_per_sample);
450  }
451  }
452 }
453 
454 static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt,
455  const AVFrame *frame, int *got_packet_ptr)
456 {
457  LibopusEncContext *opus = avctx->priv_data;
458  const int bytes_per_sample = av_get_bytes_per_sample(avctx->sample_fmt);
459  const int sample_size = avctx->channels * bytes_per_sample;
460  uint8_t *audio;
461  int ret;
462  int discard_padding;
463 
464  if (frame) {
465  ret = ff_af_queue_add(&opus->afq, frame);
466  if (ret < 0)
467  return ret;
468  if (opus->encoder_channel_map != NULL) {
469  audio = opus->samples;
471  audio, frame->data[0], opus->encoder_channel_map,
472  avctx->channels, frame->nb_samples, bytes_per_sample);
473  } else if (frame->nb_samples < opus->opts.packet_size) {
474  audio = opus->samples;
475  memcpy(audio, frame->data[0], frame->nb_samples * sample_size);
476  } else
477  audio = frame->data[0];
478  } else {
479  if (!opus->afq.remaining_samples || (!opus->afq.frame_alloc && !opus->afq.frame_count))
480  return 0;
481  audio = opus->samples;
482  memset(audio, 0, opus->opts.packet_size * sample_size);
483  }
484 
485  /* Maximum packet size taken from opusenc in opus-tools. 120ms packets
486  * consist of 6 frames in one packet. The maximum frame size is 1275
487  * bytes along with the largest possible packet header of 7 bytes. */
488  if ((ret = ff_alloc_packet(avctx, avpkt, (1275 * 6 + 7) * opus->stream_count)) < 0)
489  return ret;
490 
491  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
492  ret = opus_multistream_encode_float(opus->enc, (float *)audio,
493  opus->opts.packet_size,
494  avpkt->data, avpkt->size);
495  else
496  ret = opus_multistream_encode(opus->enc, (opus_int16 *)audio,
497  opus->opts.packet_size,
498  avpkt->data, avpkt->size);
499 
500  if (ret < 0) {
501  av_log(avctx, AV_LOG_ERROR,
502  "Error encoding frame: %s\n", opus_strerror(ret));
504  }
505 
506  av_shrink_packet(avpkt, ret);
507 
508  ff_af_queue_remove(&opus->afq, opus->opts.packet_size,
509  &avpkt->pts, &avpkt->duration);
510 
511  discard_padding = opus->opts.packet_size - avpkt->duration;
512  // Check if subtraction resulted in an overflow
513  if ((discard_padding < opus->opts.packet_size) != (avpkt->duration > 0)) {
514  av_packet_unref(avpkt);
515  return AVERROR(EINVAL);
516  }
517  if (discard_padding > 0) {
518  uint8_t* side_data = av_packet_new_side_data(avpkt,
520  10);
521  if(!side_data) {
522  av_packet_unref(avpkt);
523  return AVERROR(ENOMEM);
524  }
525  AV_WL32(side_data + 4, discard_padding);
526  }
527 
528  *got_packet_ptr = 1;
529 
530  return 0;
531 }
532 
534 {
535  LibopusEncContext *opus = avctx->priv_data;
536 
537  opus_multistream_encoder_destroy(opus->enc);
538 
539  ff_af_queue_close(&opus->afq);
540 
541  av_freep(&opus->samples);
542 
543  return 0;
544 }
545 
546 #define OFFSET(x) offsetof(LibopusEncContext, opts.x)
547 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
548 static const AVOption libopus_options[] = {
549  { "application", "Intended application type", OFFSET(application), AV_OPT_TYPE_INT, { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY, FLAGS, "application" },
550  { "voip", "Favor improved speech intelligibility", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP }, 0, 0, FLAGS, "application" },
551  { "audio", "Favor faithfulness to the input", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO }, 0, 0, FLAGS, "application" },
552  { "lowdelay", "Restrict to only the lowest delay modes", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0, FLAGS, "application" },
553  { "frame_duration", "Duration of a frame in milliseconds", OFFSET(frame_duration), AV_OPT_TYPE_FLOAT, { .dbl = 20.0 }, 2.5, 120.0, FLAGS },
554  { "packet_loss", "Expected packet loss percentage", OFFSET(packet_loss), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 100, FLAGS },
555  { "fec", "Enable inband FEC. Expected packet loss must be non-zero", OFFSET(fec), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS },
556  { "vbr", "Variable bit rate mode", OFFSET(vbr), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 2, FLAGS, "vbr" },
557  { "off", "Use constant bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, FLAGS, "vbr" },
558  { "on", "Use variable bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, FLAGS, "vbr" },
559  { "constrained", "Use constrained VBR", 0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, FLAGS, "vbr" },
560  { "mapping_family", "Channel Mapping Family", OFFSET(mapping_family), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, FLAGS, "mapping_family" },
561 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
562  { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, FLAGS },
563 #endif
564  { NULL },
565 };
566 
567 static const AVClass libopus_class = {
568  .class_name = "libopus",
569  .item_name = av_default_item_name,
570  .option = libopus_options,
571  .version = LIBAVUTIL_VERSION_INT,
572 };
573 
575  { "b", "0" },
576  { "compression_level", "10" },
577  { NULL },
578 };
579 
580 static const int libopus_sample_rates[] = {
581  48000, 24000, 16000, 12000, 8000, 0,
582 };
583 
585  .name = "libopus",
586  .long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
587  .type = AVMEDIA_TYPE_AUDIO,
588  .id = AV_CODEC_ID_OPUS,
589  .priv_data_size = sizeof(LibopusEncContext),
591  .encode2 = libopus_encode,
592  .close = libopus_encode_close,
594  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
597  .supported_samplerates = libopus_sample_rates,
598  .priv_class = &libopus_class,
599  .defaults = libopus_defaults,
600  .wrapper_name = "libopus",
601 };
libopus_class
static const AVClass libopus_class
Definition: libopusenc.c:567
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1012
av_packet_unref
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:417
AVCodec
AVCodec.
Definition: codec.h:202
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:186
libopus.h
LibopusEncOpts::application
int application
Definition: libopusenc.c:37
name
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default minimum maximum flags name is the option name
Definition: writing_filters.txt:88
libopus_encode
static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libopusenc.c:454
LibopusEncContext::opts
LibopusEncOpts opts
Definition: libopusenc.c:55
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
ff_libopus_encoder
const AVCodec ff_libopus_encoder
Definition: libopusenc.c:584
AV_WL32
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1043
ff_af_queue_remove
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
Definition: audio_frame_queue.c:75
OPUS_BANDWIDTH_NARROWBAND
@ OPUS_BANDWIDTH_NARROWBAND
Definition: opus.h:72
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:992
ff_af_queue_close
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
Definition: audio_frame_queue.c:36
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:948
ff_af_queue_init
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
Definition: audio_frame_queue.c:28
av_get_channel_layout_string
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
Definition: channel_layout.c:217
LibopusEncContext::enc
OpusMSEncoder * enc
Definition: libopusenc.c:52
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:303
LibopusEncOpts::complexity
int complexity
Definition: libopusenc.c:40
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:373
AVOption
AVOption.
Definition: opt.h:247
LibopusEncOpts::packet_size
int packet_size
Definition: libopusenc.c:42
encode.h
opus.h
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:391
OPUS_BANDWIDTH_FULLBAND
@ OPUS_BANDWIDTH_FULLBAND
Definition: opus.h:76
LibopusEncOpts::frame_duration
float frame_duration
Definition: libopusenc.c:41
opus_vorbis_channel_map
static const uint8_t opus_vorbis_channel_map[8][8]
Definition: libopusenc.c:65
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:31
libopus_encode_init
static av_cold int libopus_encode_init(AVCodecContext *avctx)
Definition: libopusenc.c:258
fail
#define fail()
Definition: checkasm.h:127
av_shrink_packet
void av_shrink_packet(AVPacket *pkt, int size)
Reduce packet size, correctly zeroing padding.
Definition: avpacket.c:114
audio_frame_queue.h
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:1701
ff_af_queue_add
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
Definition: audio_frame_queue.c:44
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen.c:29
av_cold
#define av_cold
Definition: attributes.h:90
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:485
libopus_encode_close
static av_cold int libopus_encode_close(AVCodecContext *avctx)
Definition: libopusenc.c:533
AVCodecContext::global_quality
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:449
frame_size
int frame_size
Definition: mxfenc.c:2205
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AudioFrameQueue::remaining_samples
int remaining_samples
Definition: audio_frame_queue.h:35
libopus_write_header
static void libopus_write_header(AVCodecContext *avctx, int stream_count, int coupled_stream_count, int mapping_family, const uint8_t *channel_mapping)
Definition: libopusenc.c:88
AudioFrameQueue
Definition: audio_frame_queue.h:32
channels
channels
Definition: aptx.h:33
ff_opus_error_to_averror
int ff_opus_error_to_averror(int err)
Definition: libopus.c:28
FLAGS
#define FLAGS
Definition: libopusenc.c:547
LibopusEncOpts::fec
int fec
Definition: libopusenc.c:39
OPUS_BANDWIDTH_WIDEBAND
@ OPUS_BANDWIDTH_WIDEBAND
Definition: opus.h:74
AVCodecDefault
Definition: internal.h:211
LibopusEncOpts::mapping_family
int mapping_family
Definition: libopusenc.c:44
opts
AVDictionary * opts
Definition: movenc.c:50
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
LibopusEncContext
Definition: libopusenc.c:50
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
NULL
#define NULL
Definition: coverity.c:32
libopus_configure_encoder
static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc, LibopusEncOpts *opts)
Definition: libopusenc.c:112
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:433
LibopusEncContext::samples
uint8_t * samples
Definition: libopusenc.c:54
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
src
#define src
Definition: vp8dsp.c:255
libopus_options
static const AVOption libopus_options[]
Definition: libopusenc.c:548
ff_vorbis_channel_layout_offsets
const uint8_t ff_vorbis_channel_layout_offsets[8][8]
Definition: vorbis_data.c:26
OPUS_BANDWIDTH_SUPERWIDEBAND
@ OPUS_BANDWIDTH_SUPERWIDEBAND
Definition: opus.h:75
LibopusEncContext::afq
AudioFrameQueue afq
Definition: libopusenc.c:56
AVPacket::size
int size
Definition: packet.h:374
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1000
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
sample
#define sample
Definition: flacdsp_template.c:44
LibopusEncOpts
Definition: libopusenc.c:35
libopus_defaults
static const AVCodecDefault libopus_defaults[]
Definition: libopusenc.c:574
AV_CODEC_ID_OPUS
@ AV_CODEC_ID_OPUS
Definition: codec_id.h:483
LibopusEncContext::encoder_channel_map
const uint8_t * encoder_channel_map
Definition: libopusenc.c:57
OFFSET
#define OFFSET(x)
Definition: libopusenc.c:546
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:993
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:227
av_assert2
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
bytestream_put_buffer
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
Definition: bytestream.h:372
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:366
vorbis.h
av_get_bytes_per_sample
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:484
AVCodecContext::cutoff
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:1036
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:209
AV_PKT_DATA_SKIP_SAMPLES
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: packet.h:156
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:271
avcodec.h
AudioFrameQueue::frame_count
unsigned frame_count
Definition: audio_frame_queue.h:37
libopus_check_max_channels
static int libopus_check_max_channels(AVCodecContext *avctx, int max_channels)
Definition: libopusenc.c:181
ret
ret
Definition: filter_design.txt:187
AVClass::class_name
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:71
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
LibopusEncContext::stream_count
int stream_count
Definition: libopusenc.c:53
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Definition: defs.h:40
libopus_check_vorbis_layout
static int libopus_check_vorbis_layout(AVCodecContext *avctx, int mapping_family)
Definition: libopusenc.c:192
AVCodecContext
main external API structure.
Definition: avcodec.h:383
LibopusEncOpts::vbr
int vbr
Definition: libopusenc.c:36
AudioFrameQueue::frame_alloc
unsigned frame_alloc
Definition: audio_frame_queue.h:38
channel_layout.h
av_packet_new_side_data
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, size_t size)
Allocate new information of a packet.
Definition: avpacket.c:225
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:224
AV_CODEC_CAP_DELAY
#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:82
OPUS_BANDWIDTH_MEDIUMBAND
@ OPUS_BANDWIDTH_MEDIUMBAND
Definition: opus.h:73
AVPacket
This structure stores compressed data.
Definition: packet.h:350
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:410
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:241
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
LibopusEncOpts::packet_loss
int packet_loss
Definition: libopusenc.c:38
ff_vorbis_channel_layouts
const uint64_t ff_vorbis_channel_layouts[9]
Definition: vorbis_data.c:37
bytestream.h
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
LibopusEncOpts::max_bandwidth
int max_bandwidth
Definition: libopusenc.c:43
AV_CODEC_CAP_SMALL_LAST_FRAME
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: codec.h:87
libopus_validate_layout_and_get_channel_map
static int libopus_validate_layout_and_get_channel_map(AVCodecContext *avctx, int mapping_family, const uint8_t **channel_map_result)
Definition: libopusenc.c:213
libavcodec_libopus_channel_map
static const uint8_t libavcodec_libopus_channel_map[8][8]
Definition: libopusenc.c:77
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:233
ff_alloc_packet
int ff_alloc_packet(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and allocate data.
Definition: encode.c:34
libopus_copy_samples_with_channel_map
static void libopus_copy_samples_with_channel_map(uint8_t *dst, const uint8_t *src, const uint8_t *channel_map, int nb_channels, int nb_samples, int bytes_per_sample)
Definition: libopusenc.c:440
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
channel
channel
Definition: ebur128.h:39
opus_coupled_streams
static const uint8_t opus_coupled_streams[8]
Definition: libopusenc.c:60
AVCodecContext::compression_level
int compression_level
Definition: avcodec.h:455
libopus_sample_rates
static const int libopus_sample_rates[]
Definition: libopusenc.c:580
nb_channels
int nb_channels
Definition: channel_layout.c:81