FFmpeg
libvorbisenc.c
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1 /*
2  * Copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include <vorbis/vorbisenc.h>
22 
23 #include "libavutil/avassert.h"
25 #include "libavutil/fifo.h"
26 #include "libavutil/opt.h"
27 #include "avcodec.h"
28 #include "audio_frame_queue.h"
29 #include "codec_internal.h"
30 #include "encode.h"
31 #include "internal.h"
32 #include "version.h"
33 #include "vorbis.h"
34 #include "vorbis_parser.h"
35 
36 
37 /* Number of samples the user should send in each call.
38  * This value is used because it is the LCD of all possible frame sizes, so
39  * an output packet will always start at the same point as one of the input
40  * packets.
41  */
42 #define LIBVORBIS_FRAME_SIZE 64
43 
44 #define BUFFER_SIZE (1024 * 64)
45 
46 typedef struct LibvorbisEncContext {
47  AVClass *av_class; /**< class for AVOptions */
48  vorbis_info vi; /**< vorbis_info used during init */
49  vorbis_dsp_state vd; /**< DSP state used for analysis */
50  vorbis_block vb; /**< vorbis_block used for analysis */
51  AVFifo *pkt_fifo; /**< output packet buffer */
52  int eof; /**< end-of-file flag */
53  int dsp_initialized; /**< vd has been initialized */
54  vorbis_comment vc; /**< VorbisComment info */
55  double iblock; /**< impulse block bias option */
56  AVVorbisParseContext *vp; /**< parse context to get durations */
57  AudioFrameQueue afq; /**< frame queue for timestamps */
59 
60 static const AVOption options[] = {
61  { "iblock", "Sets the impulse block bias", offsetof(LibvorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
62  { NULL }
63 };
64 
65 static const FFCodecDefault defaults[] = {
66  { "b", "0" },
67  { NULL },
68 };
69 
70 static const AVClass vorbis_class = {
71  .class_name = "libvorbis",
72  .item_name = av_default_item_name,
73  .option = options,
74  .version = LIBAVUTIL_VERSION_INT,
75 };
76 
77 static const uint8_t vorbis_encoding_channel_layout_offsets[8][8] = {
78  { 0 },
79  { 0, 1 },
80  { 0, 2, 1 },
81  { 0, 1, 2, 3 },
82  { 0, 2, 1, 3, 4 },
83  { 0, 2, 1, 4, 5, 3 },
84  { 0, 2, 1, 5, 6, 4, 3 },
85  { 0, 2, 1, 6, 7, 4, 5, 3 },
86 };
87 
88 static int vorbis_error_to_averror(int ov_err)
89 {
90  switch (ov_err) {
91  case OV_EFAULT: return AVERROR_BUG;
92  case OV_EINVAL: return AVERROR(EINVAL);
93  case OV_EIMPL: return AVERROR(EINVAL);
94  default: return AVERROR_UNKNOWN;
95  }
96 }
97 
98 static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
99 {
100  LibvorbisEncContext *s = avctx->priv_data;
101  int channels = avctx->ch_layout.nb_channels;
102  double cfreq;
103  int ret;
104 
105  if (avctx->flags & AV_CODEC_FLAG_QSCALE || !avctx->bit_rate) {
106  /* variable bitrate
107  * NOTE: we use the oggenc range of -1 to 10 for global_quality for
108  * user convenience, but libvorbis uses -0.1 to 1.0.
109  */
110  float q = avctx->global_quality / (float)FF_QP2LAMBDA;
111  /* default to 3 if the user did not set quality or bitrate */
112  if (!(avctx->flags & AV_CODEC_FLAG_QSCALE))
113  q = 3.0;
114  if ((ret = vorbis_encode_setup_vbr(vi, channels,
115  avctx->sample_rate,
116  q / 10.0)))
117  goto error;
118  } else {
119  int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
120  int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
121 
122  /* average bitrate */
123  if ((ret = vorbis_encode_setup_managed(vi, channels,
124  avctx->sample_rate, maxrate,
125  avctx->bit_rate, minrate)))
126  goto error;
127 
128  /* variable bitrate by estimate, disable slow rate management */
129  if (minrate == -1 && maxrate == -1)
130  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
131  goto error; /* should not happen */
132  }
133 
134  /* cutoff frequency */
135  if (avctx->cutoff > 0) {
136  cfreq = avctx->cutoff / 1000.0;
137  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
138  goto error; /* should not happen */
139  }
140 
141  /* impulse block bias */
142  if (s->iblock) {
143  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
144  goto error;
145  }
146 
147  if ((channels == 3 &&
149  (channels == 4 &&
152  (channels == 5 &&
155  (channels == 6 &&
158  (channels == 7 &&
160  (channels == 8 &&
162  if (avctx->ch_layout.order != AV_CHANNEL_ORDER_UNSPEC) {
163  char name[32];
164  av_channel_layout_describe(&avctx->ch_layout, name, sizeof(name));
165  av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
166  "output stream will have incorrect "
167  "channel layout.\n", name);
168  } else {
169  av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
170  "will use Vorbis channel layout for "
171  "%d channels.\n", channels);
172  }
173  }
174 
175  if ((ret = vorbis_encode_setup_init(vi)))
176  goto error;
177 
178  return 0;
179 error:
181 }
182 
183 /* How many bytes are needed for a buffer of length 'l' */
184 static int xiph_len(int l)
185 {
186  return 1 + l / 255 + l;
187 }
188 
190 {
191  LibvorbisEncContext *s = avctx->priv_data;
192 
193  /* notify vorbisenc this is EOF */
194  if (s->dsp_initialized)
195  vorbis_analysis_wrote(&s->vd, 0);
196 
197  vorbis_block_clear(&s->vb);
198  vorbis_dsp_clear(&s->vd);
199  vorbis_info_clear(&s->vi);
200 
201  av_fifo_freep2(&s->pkt_fifo);
202  ff_af_queue_close(&s->afq);
203 
204  av_vorbis_parse_free(&s->vp);
205 
206  return 0;
207 }
208 
210 {
211  LibvorbisEncContext *s = avctx->priv_data;
212  ogg_packet header, header_comm, header_code;
213  uint8_t *p;
214  unsigned int offset;
215  int ret;
216 
217  vorbis_info_init(&s->vi);
218  if ((ret = libvorbis_setup(&s->vi, avctx))) {
219  av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
220  goto error;
221  }
222  if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
223  av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
225  goto error;
226  }
227  s->dsp_initialized = 1;
228  if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
229  av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
231  goto error;
232  }
233 
234  vorbis_comment_init(&s->vc);
235  if (!(avctx->flags & AV_CODEC_FLAG_BITEXACT))
236  vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
237 
238  if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
239  &header_code))) {
241  goto error;
242  }
243 
244  avctx->extradata_size = 1 + xiph_len(header.bytes) +
245  xiph_len(header_comm.bytes) +
246  header_code.bytes;
247  p = avctx->extradata = av_malloc(avctx->extradata_size +
249  if (!p) {
250  ret = AVERROR(ENOMEM);
251  goto error;
252  }
253  p[0] = 2;
254  offset = 1;
255  offset += av_xiphlacing(&p[offset], header.bytes);
256  offset += av_xiphlacing(&p[offset], header_comm.bytes);
257  memcpy(&p[offset], header.packet, header.bytes);
258  offset += header.bytes;
259  memcpy(&p[offset], header_comm.packet, header_comm.bytes);
260  offset += header_comm.bytes;
261  memcpy(&p[offset], header_code.packet, header_code.bytes);
262  offset += header_code.bytes;
263  av_assert0(offset == avctx->extradata_size);
264 
265  s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size);
266  if (!s->vp) {
267  av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
268  return ret;
269  }
270 
271  vorbis_comment_clear(&s->vc);
272 
274  ff_af_queue_init(avctx, &s->afq);
275 
276  s->pkt_fifo = av_fifo_alloc2(BUFFER_SIZE, 1, 0);
277  if (!s->pkt_fifo) {
278  ret = AVERROR(ENOMEM);
279  goto error;
280  }
281 
282  return 0;
283 error:
284  libvorbis_encode_close(avctx);
285  return ret;
286 }
287 
289  const AVFrame *frame, int *got_packet_ptr)
290 {
291  LibvorbisEncContext *s = avctx->priv_data;
292  ogg_packet op;
293  int ret, duration;
294 
295  /* send samples to libvorbis */
296  if (frame) {
297  const int samples = frame->nb_samples;
298  float **buffer;
299  int c, channels = s->vi.channels;
300 
301  buffer = vorbis_analysis_buffer(&s->vd, samples);
302  for (c = 0; c < channels; c++) {
303  int co = (channels > 8) ? c :
305  memcpy(buffer[c], frame->extended_data[co],
306  samples * sizeof(*buffer[c]));
307  }
308  if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
309  av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
311  }
312  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
313  return ret;
314  } else {
315  if (!s->eof && s->afq.frame_alloc)
316  if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
317  av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
319  }
320  s->eof = 1;
321  }
322 
323  /* retrieve available packets from libvorbis */
324  while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
325  if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
326  break;
327  if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
328  break;
329 
330  /* add any available packets to the output packet buffer */
331  while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
332  if (av_fifo_can_write(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
333  av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
334  return AVERROR_BUG;
335  }
336  av_fifo_write(s->pkt_fifo, &op, sizeof(ogg_packet));
337  av_fifo_write(s->pkt_fifo, op.packet, op.bytes);
338  }
339  if (ret < 0) {
340  av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
341  break;
342  }
343  }
344  if (ret < 0) {
345  av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
347  }
348 
349  /* Read an available packet if possible */
350  if (av_fifo_read(s->pkt_fifo, &op, sizeof(ogg_packet)) < 0)
351  return 0;
352 
353  if ((ret = ff_get_encode_buffer(avctx, avpkt, op.bytes, 0)) < 0)
354  return ret;
355  av_fifo_read(s->pkt_fifo, avpkt->data, op.bytes);
356 
357  avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
358 
359  duration = av_vorbis_parse_frame(s->vp, avpkt->data, avpkt->size);
360  if (duration > 0) {
361  /* we do not know encoder delay until we get the first packet from
362  * libvorbis, so we have to update the AudioFrameQueue counts */
363  if (!avctx->initial_padding && s->afq.frames) {
364  avctx->initial_padding = duration;
365  av_assert0(!s->afq.remaining_delay);
366  s->afq.frames->duration += duration;
367  if (s->afq.frames->pts != AV_NOPTS_VALUE)
368  s->afq.frames->pts -= duration;
369  s->afq.remaining_samples += duration;
370  }
371  ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
372  }
373 
374  *got_packet_ptr = 1;
375  return 0;
376 }
377 
379  .p.name = "libvorbis",
380  .p.long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
381  .p.type = AVMEDIA_TYPE_AUDIO,
382  .p.id = AV_CODEC_ID_VORBIS,
383  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
385  .caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE,
386  .priv_data_size = sizeof(LibvorbisEncContext),
389  .close = libvorbis_encode_close,
390  .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
392  .p.priv_class = &vorbis_class,
393  .defaults = defaults,
394  .p.wrapper_name = "libvorbis",
395 };
error
static void error(const char *err)
Definition: target_bsf_fuzzer.c:31
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1035
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:186
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
LibvorbisEncContext::vc
vorbis_comment vc
VorbisComment info
Definition: libvorbisenc.c:54
av_vorbis_parse_free
void av_vorbis_parse_free(AVVorbisParseContext **s)
Free the parser and everything associated with it.
Definition: vorbis_parser.c:278
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
LIBAVCODEC_IDENT
#define LIBAVCODEC_IDENT
Definition: version.h:43
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
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1007
AVVorbisParseContext
Definition: vorbis_parser_internal.h:34
AVCodecContext::rc_min_rate
int64_t rc_min_rate
minimum bitrate
Definition: avcodec.h:1230
ff_af_queue_close
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
Definition: audio_frame_queue.c:36
AV_CHANNEL_LAYOUT_5POINT1
#define AV_CHANNEL_LAYOUT_5POINT1
Definition: channel_layout.h:364
AV_CODEC_FLAG_QSCALE
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:216
ff_af_queue_init
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
Definition: audio_frame_queue.c:28
AV_CHANNEL_LAYOUT_6POINT1
#define AV_CHANNEL_LAYOUT_6POINT1
Definition: channel_layout.h:370
vorbis_class
static const AVClass vorbis_class
Definition: libvorbisenc.c:70
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:325
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:374
AVOption
AVOption.
Definition: opt.h:251
encode.h
libvorbis_setup
static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
Definition: libvorbisenc.c:98
LIBVORBIS_FRAME_SIZE
#define LIBVORBIS_FRAME_SIZE
Definition: libvorbisenc.c:42
FF_CODEC_CAP_NOT_INIT_THREADSAFE
#define FF_CODEC_CAP_NOT_INIT_THREADSAFE
The codec is not known to be init-threadsafe (i.e.
Definition: codec_internal.h:33
FFCodec
Definition: codec_internal.h:118
version.h
av_vorbis_parse_frame
int av_vorbis_parse_frame(AVVorbisParseContext *s, const uint8_t *buf, int buf_size)
Get the duration for a Vorbis packet.
Definition: vorbis_parser.c:266
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:392
av_fifo_read
int av_fifo_read(AVFifo *f, void *buf, size_t nb_elems)
Read data from a FIFO.
Definition: fifo.c:240
AVChannelLayout::order
enum AVChannelOrder order
Channel order used in this layout.
Definition: channel_layout.h:295
AVERROR_UNKNOWN
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:73
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:300
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:30
AV_CHANNEL_LAYOUT_5POINT0
#define AV_CHANNEL_LAYOUT_5POINT0
Definition: channel_layout.h:363
init
static int init
Definition: av_tx.c:47
fifo.h
FFCodecDefault
Definition: codec_internal.h:88
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:122
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2065
defaults
static const FFCodecDefault defaults[]
Definition: libvorbisenc.c:65
audio_frame_queue.h
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:1732
options
static const AVOption options[]
Definition: libvorbisenc.c:60
AV_CHANNEL_LAYOUT_SURROUND
#define AV_CHANNEL_LAYOUT_SURROUND
Definition: channel_layout.h:357
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:478
FF_CODEC_ENCODE_CB
#define FF_CODEC_ENCODE_CB(func)
Definition: codec_internal.h:269
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
ogg_packet
static int ogg_packet(AVFormatContext *s, int *sid, int *dstart, int *dsize, int64_t *fpos)
find the next Ogg packet
Definition: oggdec.c:497
LibvorbisEncContext::av_class
AVClass * av_class
class for AVOptions
Definition: libvorbisenc.c:47
LibvorbisEncContext::vi
vorbis_info vi
vorbis_info used during init
Definition: libvorbisenc.c:48
libvorbis_encode_frame
static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libvorbisenc.c:288
avassert.h
ff_samples_to_time_base
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
Definition: internal.h:187
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
av_cold
#define av_cold
Definition: attributes.h:90
AV_CHANNEL_LAYOUT_2_2
#define AV_CHANNEL_LAYOUT_2_2
Definition: channel_layout.h:361
duration
int64_t duration
Definition: movenc.c:64
float
float
Definition: af_crystalizer.c:122
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:500
s
#define s(width, name)
Definition: cbs_vp9.c:256
AVCodecContext::global_quality
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:464
AV_OPT_FLAG_ENCODING_PARAM
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
Definition: opt.h:281
libvorbis_encode_close
static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
Definition: libvorbisenc.c:189
AV_CHANNEL_ORDER_UNSPEC
@ AV_CHANNEL_ORDER_UNSPEC
Only the channel count is specified, without any further information about the channel order.
Definition: channel_layout.h:106
AV_OPT_TYPE_DOUBLE
@ AV_OPT_TYPE_DOUBLE
Definition: opt.h:227
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
op
static int op(uint8_t **dst, const uint8_t *dst_end, GetByteContext *gb, int pixel, int count, int *x, int width, int linesize)
Perform decode operation.
Definition: anm.c:76
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
AudioFrameQueue
Definition: audio_frame_queue.h:32
channels
channels
Definition: aptx.h:32
av_fifo_can_write
size_t av_fifo_can_write(const AVFifo *f)
Definition: fifo.c:94
AVCodecContext::rc_max_rate
int64_t rc_max_rate
maximum bitrate
Definition: avcodec.h:1223
AV_OPT_FLAG_AUDIO_PARAM
#define AV_OPT_FLAG_AUDIO_PARAM
Definition: opt.h:283
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
NULL
#define NULL
Definition: coverity.c:32
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:448
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:237
av_channel_layout_compare
int av_channel_layout_compare(const AVChannelLayout *chl, const AVChannelLayout *chl1)
Check whether two channel layouts are semantically the same, i.e.
Definition: channel_layout.c:930
vorbis_parser.h
AV_CHANNEL_LAYOUT_7POINT1
#define AV_CHANNEL_LAYOUT_7POINT1
Definition: channel_layout.h:375
LibvorbisEncContext::afq
AudioFrameQueue afq
frame queue for timestamps
Definition: libvorbisenc.c:57
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AVPacket::size
int size
Definition: packet.h:375
AVFifo
Definition: fifo.c:35
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
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:290
codec_internal.h
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
av_xiphlacing
unsigned int av_xiphlacing(unsigned char *s, unsigned int v)
Encode extradata length to a buffer.
Definition: utils.c:858
LibvorbisEncContext::vb
vorbis_block vb
vorbis_block used for analysis
Definition: libvorbisenc.c:50
LibvorbisEncContext::vp
AVVorbisParseContext * vp
parse context to get durations
Definition: libvorbisenc.c:56
header
static const uint8_t header[24]
Definition: sdr2.c:67
BUFFER_SIZE
#define BUFFER_SIZE
Definition: libvorbisenc.c:44
LibvorbisEncContext
Definition: libvorbisenc.c:46
xiph_len
static int xiph_len(int l)
Definition: libvorbisenc.c:184
offset
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 offset
Definition: writing_filters.txt:86
vorbis_error_to_averror
static int vorbis_error_to_averror(int ov_err)
Definition: libvorbisenc.c:88
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:367
vorbis.h
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:499
LibvorbisEncContext::pkt_fifo
AVFifo * pkt_fifo
output packet buffer
Definition: libvorbisenc.c:51
av_fifo_alloc2
AVFifo * av_fifo_alloc2(size_t nb_elems, size_t elem_size, unsigned int flags)
Allocate and initialize an AVFifo with a given element size.
Definition: fifo.c:47
AVCodecContext::cutoff
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:1059
vorbis_encoding_channel_layout_offsets
static const uint8_t vorbis_encoding_channel_layout_offsets[8][8]
Definition: libvorbisenc.c:77
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
av_fifo_freep2
void av_fifo_freep2(AVFifo **f)
Free an AVFifo and reset pointer to NULL.
Definition: fifo.c:286
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:211
avcodec.h
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
av_vorbis_parse_init
AVVorbisParseContext * av_vorbis_parse_init(const uint8_t *extradata, int extradata_size)
Allocate and initialize the Vorbis parser using headers in the extradata.
Definition: vorbis_parser.c:283
LibvorbisEncContext::eof
int eof
end-of-file flag
Definition: libvorbisenc.c:52
libvorbis_encode_init
static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
Definition: libvorbisenc.c:209
av_fifo_write
int av_fifo_write(AVFifo *f, const void *buf, size_t nb_elems)
Write data into a FIFO.
Definition: fifo.c:188
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Definition: defs.h:40
AV_CHANNEL_LAYOUT_5POINT0_BACK
#define AV_CHANNEL_LAYOUT_5POINT0_BACK
Definition: channel_layout.h:365
AVCodecContext
main external API structure.
Definition: avcodec.h:398
channel_layout.h
LibvorbisEncContext::iblock
double iblock
impulse block bias option
Definition: libvorbisenc.c:55
buffer
the frame and frame reference mechanism is intended to as much as expensive copies of that data while still allowing the filters to produce correct results The data is stored in buffers represented by AVFrame structures Several references can point to the same frame buffer
Definition: filter_design.txt:49
ff_get_encode_buffer
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
Definition: encode.c:79
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
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
av_channel_layout_describe
int av_channel_layout_describe(const AVChannelLayout *channel_layout, char *buf, size_t buf_size)
Get a human-readable string describing the channel layout properties.
Definition: channel_layout.c:776
LibvorbisEncContext::dsp_initialized
int dsp_initialized
vd has been initialized
Definition: libvorbisenc.c:53
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:288
ff_libvorbis_encoder
const FFCodec ff_libvorbis_encoder
Definition: libvorbisenc.c:378
AVPacket
This structure stores compressed data.
Definition: packet.h:351
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:425
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:52
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AV_CODEC_ID_VORBIS
@ AV_CODEC_ID_VORBIS
Definition: codec_id.h:434
FF_QP2LAMBDA
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:227
AV_CHANNEL_LAYOUT_QUAD
#define AV_CHANNEL_LAYOUT_QUAD
Definition: channel_layout.h:362
AV_CHANNEL_LAYOUT_5POINT1_BACK
#define AV_CHANNEL_LAYOUT_5POINT1_BACK
Definition: channel_layout.h:366
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
LibvorbisEncContext::vd
vorbis_dsp_state vd
DSP state used for analysis
Definition: libvorbisenc.c:49