FFmpeg: libavcodec/libopusenc.c Source File

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 
25 #include "libavutil/opt.h"
26 #include "avcodec.h"
27 #include "bytestream.h"
28 #include "internal.h"
29 #include "libopus.h"
30 #include "vorbis.h"
31 #include "audio_frame_queue.h"
32 
33  typedef struct LibopusEncOpts {
34   int vbr;
35   int application;
36   int packet_loss;
37   int complexity;
38   float frame_duration;
39   int packet_size;
40   int max_bandwidth;
41 } LibopusEncOpts;
42 
43  typedef struct LibopusEncContext {
44   AVClass *class;
45   OpusMSEncoder *enc;
46   int stream_count;
47   uint8_t *samples;
48   LibopusEncOpts opts;
49   AudioFrameQueue afq;
50 } LibopusEncContext;
51 
52  static const uint8_t opus_coupled_streams[8] = {
53  0, 1, 1, 2, 2, 2, 2, 3
54 };
55 
56 /* Opus internal to Vorbis channel order mapping written in the header */
57  static const uint8_t opus_vorbis_channel_map[8][8] = {
58  { 0 },
59  { 0, 1 },
60  { 0, 2, 1 },
61  { 0, 1, 2, 3 },
62  { 0, 4, 1, 2, 3 },
63  { 0, 4, 1, 2, 3, 5 },
64  { 0, 4, 1, 2, 3, 5, 6 },
65  { 0, 6, 1, 2, 3, 4, 5, 7 },
66 };
67 
68 /* libavcodec to libopus channel order mapping, passed to libopus */
69  static const uint8_t libavcodec_libopus_channel_map[8][8] = {
70  { 0 },
71  { 0, 1 },
72  { 0, 1, 2 },
73  { 0, 1, 2, 3 },
74  { 0, 1, 3, 4, 2 },
75  { 0, 1, 4, 5, 2, 3 },
76  { 0, 1, 5, 6, 2, 4, 3 },
77  { 0, 1, 6, 7, 4, 5, 2, 3 },
78 };
79 
80  static void libopus_write_header(AVCodecContext *avctx, int stream_count,
81  int coupled_stream_count,
82  const uint8_t *channel_mapping)
83 {
84  uint8_t *p = avctx->extradata;
85  int channels = avctx->channels;
86 
87  bytestream_put_buffer(&p, "OpusHead", 8);
88  bytestream_put_byte(&p, 1); /* Version */
89  bytestream_put_byte(&p, channels);
90  bytestream_put_le16(&p, avctx->initial_padding); /* Lookahead samples at 48kHz */
91  bytestream_put_le32(&p, avctx->sample_rate); /* Original sample rate */
92  bytestream_put_le16(&p, 0); /* Gain of 0dB is recommended. */
93 
94  /* Channel mapping */
95  if (channels > 2) {
96  bytestream_put_byte(&p, channels <= 8 ? 1 : 255);
97  bytestream_put_byte(&p, stream_count);
98  bytestream_put_byte(&p, coupled_stream_count);
99  bytestream_put_buffer(&p, channel_mapping, channels);
100  } else {
101  bytestream_put_byte(&p, 0);
102  }
103 }
104 
105  static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc,
106  LibopusEncOpts *opts)
107 {
108  int ret;
109 
110  if (avctx->global_quality) {
111  av_log(avctx, AV_LOG_ERROR,
112  "Quality-based encoding not supported, "
113  "please specify a bitrate and VBR setting.\n");
114  return AVERROR(EINVAL);
115  }
116 
117  ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->bit_rate));
118  if (ret != OPUS_OK) {
119  av_log(avctx, AV_LOG_ERROR,
120  "Failed to set bitrate: %s\n", opus_strerror(ret));
121  return ret;
122  }
123 
124  ret = opus_multistream_encoder_ctl(enc,
125  OPUS_SET_COMPLEXITY(opts->complexity));
126  if (ret != OPUS_OK)
127  av_log(avctx, AV_LOG_WARNING,
128  "Unable to set complexity: %s\n", opus_strerror(ret));
129 
130  ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->vbr));
131  if (ret != OPUS_OK)
132  av_log(avctx, AV_LOG_WARNING,
133  "Unable to set VBR: %s\n", opus_strerror(ret));
134 
135  ret = opus_multistream_encoder_ctl(enc,
136  OPUS_SET_VBR_CONSTRAINT(opts->vbr == 2));
137  if (ret != OPUS_OK)
138  av_log(avctx, AV_LOG_WARNING,
139  "Unable to set constrained VBR: %s\n", opus_strerror(ret));
140 
141  ret = opus_multistream_encoder_ctl(enc,
142  OPUS_SET_PACKET_LOSS_PERC(opts->packet_loss));
143  if (ret != OPUS_OK)
144  av_log(avctx, AV_LOG_WARNING,
145  "Unable to set expected packet loss percentage: %s\n",
146  opus_strerror(ret));
147 
148  if (avctx->cutoff) {
149  ret = opus_multistream_encoder_ctl(enc,
150  OPUS_SET_MAX_BANDWIDTH(opts->max_bandwidth));
151  if (ret != OPUS_OK)
152  av_log(avctx, AV_LOG_WARNING,
153  "Unable to set maximum bandwidth: %s\n", opus_strerror(ret));
154  }
155 
156  return OPUS_OK;
157 }
158 
159  static av_cold int libopus_encode_init(AVCodecContext *avctx)
160 {
161  LibopusEncContext *opus = avctx->priv_data;
162  const uint8_t *channel_mapping;
163  OpusMSEncoder *enc;
164  int ret = OPUS_OK;
165  int coupled_stream_count, header_size, frame_size;
166 
167  coupled_stream_count = opus_coupled_streams[avctx->channels - 1];
168  opus->stream_count = avctx->channels - coupled_stream_count;
169  channel_mapping = libavcodec_libopus_channel_map[avctx->channels - 1];
170 
171  /* FIXME: Opus can handle up to 255 channels. However, the mapping for
172  * anything greater than 8 is undefined. */
173  if (avctx->channels > 8) {
174  av_log(avctx, AV_LOG_ERROR,
175  "Channel layout undefined for %d channels.\n", avctx->channels);
176  return AVERROR_PATCHWELCOME;
177  }
178  if (!avctx->bit_rate) {
179  /* Sane default copied from opusenc */
180  avctx->bit_rate = 64000 * opus->stream_count +
181  32000 * coupled_stream_count;
182  av_log(avctx, AV_LOG_WARNING,
183  "No bit rate set. Defaulting to %"PRId64" bps.\n", (int64_t)avctx->bit_rate);
184  }
185 
186  if (avctx->bit_rate < 500 || avctx->bit_rate > 256000 * avctx->channels) {
187  av_log(avctx, AV_LOG_ERROR, "The bit rate %"PRId64" bps is unsupported. "
188  "Please choose a value between 500 and %d.\n", (int64_t)avctx->bit_rate,
189  256000 * avctx->channels);
190  return AVERROR(EINVAL);
191  }
192 
193  frame_size = opus->opts.frame_duration * 48000 / 1000;
194  switch (frame_size) {
195  case 120:
196  case 240:
197  if (opus->opts.application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
198  av_log(avctx, AV_LOG_WARNING,
199  "LPC mode cannot be used with a frame duration of less "
200  "than 10ms. Enabling restricted low-delay mode.\n"
201  "Use a longer frame duration if this is not what you want.\n");
202  /* Frame sizes less than 10 ms can only use MDCT mode, so switching to
203  * RESTRICTED_LOWDELAY avoids an unnecessary extra 2.5ms lookahead. */
204  opus->opts.application = OPUS_APPLICATION_RESTRICTED_LOWDELAY;
205  case 480:
206  case 960:
207  case 1920:
208  case 2880:
209  opus->opts.packet_size =
210  avctx->frame_size = frame_size * avctx->sample_rate / 48000;
211  break;
212  default:
213  av_log(avctx, AV_LOG_ERROR, "Invalid frame duration: %g.\n"
214  "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40 or 60.\n",
215  opus->opts.frame_duration);
216  return AVERROR(EINVAL);
217  }
218 
219  if (avctx->compression_level < 0 || avctx->compression_level > 10) {
220  av_log(avctx, AV_LOG_WARNING,
221  "Compression level must be in the range 0 to 10. "
222  "Defaulting to 10.\n");
223  opus->opts.complexity = 10;
224  } else {
225  opus->opts.complexity = avctx->compression_level;
226  }
227 
228  if (avctx->cutoff) {
229  switch (avctx->cutoff) {
230  case 4000:
231  opus->opts.max_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
232  break;
233  case 6000:
234  opus->opts.max_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
235  break;
236  case 8000:
237  opus->opts.max_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
238  break;
239  case 12000:
240  opus->opts.max_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
241  break;
242  case 20000:
243  opus->opts.max_bandwidth = OPUS_BANDWIDTH_FULLBAND;
244  break;
245  default:
246  av_log(avctx, AV_LOG_WARNING,
247  "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n"
248  "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",
249  avctx->cutoff);
250  avctx->cutoff = 0;
251  }
252  }
253 
254  enc = opus_multistream_encoder_create(avctx->sample_rate, avctx->channels,
255  opus->stream_count,
256  coupled_stream_count,
257  channel_mapping,
258  opus->opts.application, &ret);
259  if (ret != OPUS_OK) {
260  av_log(avctx, AV_LOG_ERROR,
261  "Failed to create encoder: %s\n", opus_strerror(ret));
262  return ff_opus_error_to_averror(ret);
263  }
264 
265  ret = libopus_configure_encoder(avctx, enc, &opus->opts);
266  if (ret != OPUS_OK) {
267  ret = ff_opus_error_to_averror(ret);
268  goto fail;
269  }
270 
271  header_size = 19 + (avctx->channels > 2 ? 2 + avctx->channels : 0);
272  avctx->extradata = av_malloc(header_size + AV_INPUT_BUFFER_PADDING_SIZE);
273  if (!avctx->extradata) {
274  av_log(avctx, AV_LOG_ERROR, "Failed to allocate extradata.\n");
275  ret = AVERROR(ENOMEM);
276  goto fail;
277  }
278  avctx->extradata_size = header_size;
279 
280  opus->samples = av_mallocz_array(frame_size, avctx->channels *
281  av_get_bytes_per_sample(avctx->sample_fmt));
282  if (!opus->samples) {
283  av_log(avctx, AV_LOG_ERROR, "Failed to allocate samples buffer.\n");
284  ret = AVERROR(ENOMEM);
285  goto fail;
286  }
287 
288  ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->initial_padding));
289  if (ret != OPUS_OK)
290  av_log(avctx, AV_LOG_WARNING,
291  "Unable to get number of lookahead samples: %s\n",
292  opus_strerror(ret));
293 
294  libopus_write_header(avctx, opus->stream_count, coupled_stream_count,
295  opus_vorbis_channel_map[avctx->channels - 1]);
296 
297  ff_af_queue_init(avctx, &opus->afq);
298 
299  opus->enc = enc;
300 
301  return 0;
302 
303 fail:
304  opus_multistream_encoder_destroy(enc);
305  av_freep(&avctx->extradata);
306  return ret;
307 }
308 
309  static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt,
310  const AVFrame *frame, int *got_packet_ptr)
311 {
312  LibopusEncContext *opus = avctx->priv_data;
313  const int sample_size = avctx->channels *
314  av_get_bytes_per_sample(avctx->sample_fmt);
315  uint8_t *audio;
316  int ret;
317  int discard_padding;
318 
319  if (frame) {
320  ret = ff_af_queue_add(&opus->afq, frame);
321  if (ret < 0)
322  return ret;
323  if (frame->nb_samples < opus->opts.packet_size) {
324  audio = opus->samples;
325  memcpy(audio, frame->data[0], frame->nb_samples * sample_size);
326  } else
327  audio = frame->data[0];
328  } else {
329  if (!opus->afq.remaining_samples || (!opus->afq.frame_alloc && !opus->afq.frame_count))
330  return 0;
331  audio = opus->samples;
332  memset(audio, 0, opus->opts.packet_size * sample_size);
333  }
334 
335  /* Maximum packet size taken from opusenc in opus-tools. 60ms packets
336  * consist of 3 frames in one packet. The maximum frame size is 1275
337  * bytes along with the largest possible packet header of 7 bytes. */
338  if ((ret = ff_alloc_packet2(avctx, avpkt, (1275 * 3 + 7) * opus->stream_count, 0)) < 0)
339  return ret;
340 
341  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
342  ret = opus_multistream_encode_float(opus->enc, (float *)audio,
343  opus->opts.packet_size,
344  avpkt->data, avpkt->size);
345  else
346  ret = opus_multistream_encode(opus->enc, (opus_int16 *)audio,
347  opus->opts.packet_size,
348  avpkt->data, avpkt->size);
349 
350  if (ret < 0) {
351  av_log(avctx, AV_LOG_ERROR,
352  "Error encoding frame: %s\n", opus_strerror(ret));
353  return ff_opus_error_to_averror(ret);
354  }
355 
356  av_shrink_packet(avpkt, ret);
357 
358  ff_af_queue_remove(&opus->afq, opus->opts.packet_size,
359  &avpkt->pts, &avpkt->duration);
360 
361  discard_padding = opus->opts.packet_size - avpkt->duration;
362  // Check if subtraction resulted in an overflow
363  if ((discard_padding < opus->opts.packet_size) != (avpkt->duration > 0)) {
364  av_packet_unref(avpkt);
365  av_free(avpkt);
366  return AVERROR(EINVAL);
367  }
368  if (discard_padding > 0) {
369  uint8_t* side_data = av_packet_new_side_data(avpkt,
370  AV_PKT_DATA_SKIP_SAMPLES,
371  10);
372  if(!side_data) {
373  av_packet_unref(avpkt);
374  av_free(avpkt);
375  return AVERROR(ENOMEM);
376  }
377  AV_WL32(side_data + 4, discard_padding);
378  }
379 
380  *got_packet_ptr = 1;
381 
382  return 0;
383 }
384 
385  static av_cold int libopus_encode_close(AVCodecContext *avctx)
386 {
387  LibopusEncContext *opus = avctx->priv_data;
388 
389  opus_multistream_encoder_destroy(opus->enc);
390 
391  ff_af_queue_close(&opus->afq);
392 
393  av_freep(&opus->samples);
394  av_freep(&avctx->extradata);
395 
396  return 0;
397 }
398 
399  #define OFFSET(x) offsetof(LibopusEncContext, opts.x)
400  #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
401  static const AVOption libopus_options[] = {
402  { "application", "Intended application type", OFFSET(application), AV_OPT_TYPE_INT, { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY, FLAGS, "application" },
403  { "voip", "Favor improved speech intelligibility", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP }, 0, 0, FLAGS, "application" },
404  { "audio", "Favor faithfulness to the input", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO }, 0, 0, FLAGS, "application" },
405  { "lowdelay", "Restrict to only the lowest delay modes", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0, FLAGS, "application" },
406  { "frame_duration", "Duration of a frame in milliseconds", OFFSET(frame_duration), AV_OPT_TYPE_FLOAT, { .dbl = 20.0 }, 2.5, 60.0, FLAGS },
407  { "packet_loss", "Expected packet loss percentage", OFFSET(packet_loss), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 100, FLAGS },
408  { "vbr", "Variable bit rate mode", OFFSET(vbr), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 2, FLAGS, "vbr" },
409  { "off", "Use constant bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, FLAGS, "vbr" },
410  { "on", "Use variable bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, FLAGS, "vbr" },
411  { "constrained", "Use constrained VBR", 0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, FLAGS, "vbr" },
412  { NULL },
413 };
414 
415  static const AVClass libopus_class = {
416  .class_name = "libopus",
417  .item_name = av_default_item_name,
418  .option = libopus_options,
419  .version = LIBAVUTIL_VERSION_INT,
420 };
421 
422  static const AVCodecDefault libopus_defaults[] = {
423  { "b", "0" },
424  { "compression_level", "10" },
425  { NULL },
426 };
427 
428  static const int libopus_sample_rates[] = {
429  48000, 24000, 16000, 12000, 8000, 0,
430 };
431 
432  AVCodec ff_libopus_encoder = {
433  .name = "libopus",
434  .long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
435  .type = AVMEDIA_TYPE_AUDIO,
436  .id = AV_CODEC_ID_OPUS,
437  .priv_data_size = sizeof(LibopusEncContext),
438  .init = libopus_encode_init,
439  .encode2 = libopus_encode,
440  .close = libopus_encode_close,
441  .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
442  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
443  AV_SAMPLE_FMT_FLT,
444  AV_SAMPLE_FMT_NONE },
445  .channel_layouts = ff_vorbis_channel_layouts,
446  .supported_samplerates = libopus_sample_rates,
447  .priv_class = &libopus_class,
448  .defaults = libopus_defaults,
449 };
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
NULL
#define NULL
Definition: coverity.c:32
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:181
libopus_configure_encoder
static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc, LibopusEncOpts *opts)
Definition: libopusenc.c:105
AVOption
AVOption.
Definition: opt.h:245
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1597
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:70
av_shrink_packet
void av_shrink_packet(AVPacket *pkt, int size)
Reduce packet size, correctly zeroing padding.
Definition: avpacket.c:101
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AV_OPT_TYPE_INT
Definition: opt.h:223
OFFSET
#define OFFSET(x)
Definition: libopusenc.c:399
AVPacket::size
int size
Definition: avcodec.h:1468
AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:60
LibopusEncContext::samples
uint8_t * samples
Definition: libopusenc.c:47
AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:64
AVCodec
AVCodec.
Definition: avcodec.h:3392
LibopusEncOpts::packet_size
int packet_size
Definition: libopusenc.c:39
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:72
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: avcodec.h:881
ff_opus_error_to_averror
int ff_opus_error_to_averror(int err)
Definition: libopus.c:28
AV_OPT_TYPE_CONST
Definition: opt.h:231
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2295
uint8_t
uint8_t
Definition: audio_convert.c:194
av_cold
#define av_cold
Definition: attributes.h:82
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:31
opt.h
AVOptions.
ff_af_queue_init
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
Definition: audio_frame_queue.c:28
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1485
libopus_class
static const AVClass libopus_class
Definition: libopusenc.c:415
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1647
frame
static AVFrame * frame
Definition: demuxing_decoding.c:53
AVPacket::data
uint8_t * data
Definition: avcodec.h:1467
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
LibopusEncContext::enc
OpusMSEncoder * enc
Definition: libopusenc.c:45
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_default_item_name
av_default_item_name
Definition: libopenh264enc.c:68
AVERROR
#define AVERROR(e)
Definition: error.h:43
AV_OPT_TYPE_FLOAT
Definition: opt.h:226
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:176
libavcodec_libopus_channel_map
static const uint8_t libavcodec_libopus_channel_map[8][8]
Definition: libopusenc.c:69
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:3204
AVMEDIA_TYPE_AUDIO
Definition: avutil.h:194
AVCodec::name
const char * name
Name of the codec implementation.
Definition: avcodec.h:3399
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
fail
#define fail()
Definition: checkasm.h:80
opus_coupled_streams
static const uint8_t opus_coupled_streams[8]
Definition: libopusenc.c:52
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: avcodec.h:886
libopus_sample_rates
static const int libopus_sample_rates[]
Definition: libopusenc.c:428
AV_CODEC_ID_OPUS
Definition: avcodec.h:482
AudioFrameQueue::frame_count
unsigned frame_count
Definition: audio_frame_queue.h:37
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2307
libopus_encode_init
static av_cold int libopus_encode_init(AVCodecContext *avctx)
Definition: libopusenc.c:159
frame_size
int frame_size
Definition: mxfenc.c:1821
avcodec.h
Libavcodec external API header.
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
AVCodecContext::compression_level
int compression_level
Definition: avcodec.h:1619
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:2287
AudioFrameQueue::frame_alloc
unsigned frame_alloc
Definition: audio_frame_queue.h:38
AVCodecContext
main external API structure.
Definition: avcodec.h:1532
libopus_encode
static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libopusenc.c:309
av_packet_unref
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:545
ff_vorbis_channel_layouts
const uint64_t ff_vorbis_channel_layouts[9]
Definition: vorbis_data.c:47
ff_libopus_encoder
AVCodec ff_libopus_encoder
Definition: libopusenc.c:432
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:1648
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: avcodec.h:1352
libopus_encode_close
static av_cold int libopus_encode_close(AVCodecContext *avctx)
Definition: libopusenc.c:385
ff_alloc_packet2
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1621
AVCodecContext::global_quality
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1613
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:192
av_get_bytes_per_sample
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:104
audio_frame_queue.h
opus_vorbis_channel_map
static const uint8_t opus_vorbis_channel_map[8][8]
Definition: libopusenc.c:57
LibopusEncContext::opts
LibopusEncOpts opts
Definition: libopusenc.c:48
internal.h
common internal api header.
LibopusEncContext::afq
AudioFrameQueue afq
Definition: libopusenc.c:49
AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:62
libopus_defaults
static const AVCodecDefault libopus_defaults[]
Definition: libopusenc.c:422
bytestream_put_buffer
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
Definition: bytestream.h:368
LibopusEncOpts::packet_loss
int packet_loss
Definition: libopusenc.c:36
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:635
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1574
AVCodecContext::cutoff
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:2331
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
LibopusEncOpts::application
int application
Definition: libopusenc.c:35
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:2288
LibopusEncOpts::max_bandwidth
int max_bandwidth
Definition: libopusenc.c:40
LibopusEncOpts::frame_duration
float frame_duration
Definition: libopusenc.c:38
libopus_options
static const AVOption libopus_options[]
Definition: libopusenc.c:401
opts
AVDictionary * opts
Definition: movenc-test.c:50
ff_af_queue_close
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
Definition: audio_frame_queue.c:36
av_mallocz_array
static void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.h:229
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
av_packet_new_side_data
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
Definition: avpacket.c:299
FLAGS
#define FLAGS
Definition: libopusenc.c:400
AVPacket
This structure stores compressed data.
Definition: avcodec.h:1444
LibopusEncOpts::complexity
int complexity
Definition: libopusenc.c:37
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:235
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1460
libopus_write_header
static void libopus_write_header(AVCodecContext *avctx, int stream_count, int coupled_stream_count, const uint8_t *channel_mapping)
Definition: libopusenc.c:80
AV_WL32
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
Generated on Mon Feb 15 2016 15:20:39 for FFmpeg by   doxygen 1.8.6

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