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"
24 #include "libavutil/fifo.h"
25 #include "libavutil/opt.h"
26 #include "avcodec.h"
27 #include "audio_frame_queue.h"
28 #include "internal.h"
29 #include "vorbis.h"
30 #include "vorbis_parser.h"
31 
32 
33 /* Number of samples the user should send in each call.
34  * This value is used because it is the LCD of all possible frame sizes, so
35  * an output packet will always start at the same point as one of the input
36  * packets.
37  */
38 #define LIBVORBIS_FRAME_SIZE 64
39 
40 #define BUFFER_SIZE (1024 * 64)
41 
42 typedef struct LibvorbisEncContext {
43  AVClass *av_class; /**< class for AVOptions */
44  vorbis_info vi; /**< vorbis_info used during init */
45  vorbis_dsp_state vd; /**< DSP state used for analysis */
46  vorbis_block vb; /**< vorbis_block used for analysis */
47  AVFifoBuffer *pkt_fifo; /**< output packet buffer */
48  int eof; /**< end-of-file flag */
49  int dsp_initialized; /**< vd has been initialized */
50  vorbis_comment vc; /**< VorbisComment info */
51  double iblock; /**< impulse block bias option */
52  AVVorbisParseContext *vp; /**< parse context to get durations */
53  AudioFrameQueue afq; /**< frame queue for timestamps */
55 
56 static const AVOption options[] = {
57  { "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 },
58  { NULL }
59 };
60 
61 static const AVCodecDefault defaults[] = {
62  { "b", "0" },
63  { NULL },
64 };
65 
66 static const AVClass vorbis_class = {
67  .class_name = "libvorbis",
68  .item_name = av_default_item_name,
69  .option = options,
70  .version = LIBAVUTIL_VERSION_INT,
71 };
72 
73 static int vorbis_error_to_averror(int ov_err)
74 {
75  switch (ov_err) {
76  case OV_EFAULT: return AVERROR_BUG;
77  case OV_EINVAL: return AVERROR(EINVAL);
78  case OV_EIMPL: return AVERROR(EINVAL);
79  default: return AVERROR_UNKNOWN;
80  }
81 }
82 
83 static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
84 {
86  double cfreq;
87  int ret;
88 
89  if (avctx->flags & AV_CODEC_FLAG_QSCALE || !avctx->bit_rate) {
90  /* variable bitrate
91  * NOTE: we use the oggenc range of -1 to 10 for global_quality for
92  * user convenience, but libvorbis uses -0.1 to 1.0.
93  */
94  float q = avctx->global_quality / (float)FF_QP2LAMBDA;
95  /* default to 3 if the user did not set quality or bitrate */
96  if (!(avctx->flags & AV_CODEC_FLAG_QSCALE))
97  q = 3.0;
98  if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
99  avctx->sample_rate,
100  q / 10.0)))
101  goto error;
102  } else {
103  int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
104  int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
105 
106  /* average bitrate */
107  if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
108  avctx->sample_rate, maxrate,
109  avctx->bit_rate, minrate)))
110  goto error;
111 
112  /* variable bitrate by estimate, disable slow rate management */
113  if (minrate == -1 && maxrate == -1)
114  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
115  goto error; /* should not happen */
116  }
117 
118  /* cutoff frequency */
119  if (avctx->cutoff > 0) {
120  cfreq = avctx->cutoff / 1000.0;
121  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
122  goto error; /* should not happen */
123  }
124 
125  /* impulse block bias */
126  if (s->iblock) {
127  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
128  goto error;
129  }
130 
131  if (avctx->channels == 3 &&
133  avctx->channels == 4 &&
134  avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
135  avctx->channel_layout != AV_CH_LAYOUT_QUAD ||
136  avctx->channels == 5 &&
139  avctx->channels == 6 &&
142  avctx->channels == 7 &&
144  avctx->channels == 8 &&
146  if (avctx->channel_layout) {
147  char name[32];
148  av_get_channel_layout_string(name, sizeof(name), avctx->channels,
149  avctx->channel_layout);
150  av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
151  "output stream will have incorrect "
152  "channel layout.\n", name);
153  } else {
154  av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
155  "will use Vorbis channel layout for "
156  "%d channels.\n", avctx->channels);
157  }
158  }
159 
160  if ((ret = vorbis_encode_setup_init(vi)))
161  goto error;
162 
163  return 0;
164 error:
165  return vorbis_error_to_averror(ret);
166 }
167 
168 /* How many bytes are needed for a buffer of length 'l' */
169 static int xiph_len(int l)
170 {
171  return 1 + l / 255 + l;
172 }
173 
175 {
176  LibvorbisEncContext *s = avctx->priv_data;
177 
178  /* notify vorbisenc this is EOF */
179  if (s->dsp_initialized)
180  vorbis_analysis_wrote(&s->vd, 0);
181 
182  vorbis_block_clear(&s->vb);
183  vorbis_dsp_clear(&s->vd);
184  vorbis_info_clear(&s->vi);
185 
186  av_fifo_freep(&s->pkt_fifo);
187  ff_af_queue_close(&s->afq);
188  av_freep(&avctx->extradata);
189 
191 
192  return 0;
193 }
194 
196 {
197  LibvorbisEncContext *s = avctx->priv_data;
198  ogg_packet header, header_comm, header_code;
199  uint8_t *p;
200  unsigned int offset;
201  int ret;
202 
203  vorbis_info_init(&s->vi);
204  if ((ret = libvorbis_setup(&s->vi, avctx))) {
205  av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
206  goto error;
207  }
208  if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
209  av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
210  ret = vorbis_error_to_averror(ret);
211  goto error;
212  }
213  s->dsp_initialized = 1;
214  if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
215  av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
216  ret = vorbis_error_to_averror(ret);
217  goto error;
218  }
219 
220  vorbis_comment_init(&s->vc);
221  if (!(avctx->flags & AV_CODEC_FLAG_BITEXACT))
222  vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
223 
224  if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
225  &header_code))) {
226  ret = vorbis_error_to_averror(ret);
227  goto error;
228  }
229 
230  avctx->extradata_size = 1 + xiph_len(header.bytes) +
231  xiph_len(header_comm.bytes) +
232  header_code.bytes;
233  p = avctx->extradata = av_malloc(avctx->extradata_size +
235  if (!p) {
236  ret = AVERROR(ENOMEM);
237  goto error;
238  }
239  p[0] = 2;
240  offset = 1;
241  offset += av_xiphlacing(&p[offset], header.bytes);
242  offset += av_xiphlacing(&p[offset], header_comm.bytes);
243  memcpy(&p[offset], header.packet, header.bytes);
244  offset += header.bytes;
245  memcpy(&p[offset], header_comm.packet, header_comm.bytes);
246  offset += header_comm.bytes;
247  memcpy(&p[offset], header_code.packet, header_code.bytes);
248  offset += header_code.bytes;
249  av_assert0(offset == avctx->extradata_size);
250 
251  s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size);
252  if (!s->vp) {
253  av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
254  return ret;
255  }
256 
257  vorbis_comment_clear(&s->vc);
258 
260  ff_af_queue_init(avctx, &s->afq);
261 
263  if (!s->pkt_fifo) {
264  ret = AVERROR(ENOMEM);
265  goto error;
266  }
267 
268  return 0;
269 error:
270  libvorbis_encode_close(avctx);
271  return ret;
272 }
273 
275  const AVFrame *frame, int *got_packet_ptr)
276 {
277  LibvorbisEncContext *s = avctx->priv_data;
278  ogg_packet op;
279  int ret, duration;
280 
281  /* send samples to libvorbis */
282  if (frame) {
283  const int samples = frame->nb_samples;
284  float **buffer;
285  int c, channels = s->vi.channels;
286 
287  buffer = vorbis_analysis_buffer(&s->vd, samples);
288  for (c = 0; c < channels; c++) {
289  int co = (channels > 8) ? c :
291  memcpy(buffer[c], frame->extended_data[co],
292  samples * sizeof(*buffer[c]));
293  }
294  if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
295  av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
296  return vorbis_error_to_averror(ret);
297  }
298  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
299  return ret;
300  } else {
301  if (!s->eof && s->afq.frame_alloc)
302  if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
303  av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
304  return vorbis_error_to_averror(ret);
305  }
306  s->eof = 1;
307  }
308 
309  /* retrieve available packets from libvorbis */
310  while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
311  if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
312  break;
313  if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
314  break;
315 
316  /* add any available packets to the output packet buffer */
317  while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
318  if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
319  av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
320  return AVERROR_BUG;
321  }
322  av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
323  av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
324  }
325  if (ret < 0) {
326  av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
327  break;
328  }
329  }
330  if (ret < 0) {
331  av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
332  return vorbis_error_to_averror(ret);
333  }
334 
335  /* check for available packets */
336  if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
337  return 0;
338 
339  av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
340 
341  if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes, 0)) < 0)
342  return ret;
343  av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
344 
345  avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
346 
347  duration = av_vorbis_parse_frame(s->vp, avpkt->data, avpkt->size);
348  if (duration > 0) {
349  /* we do not know encoder delay until we get the first packet from
350  * libvorbis, so we have to update the AudioFrameQueue counts */
351  if (!avctx->initial_padding && s->afq.frames) {
352  avctx->initial_padding = duration;
354  s->afq.frames->duration += duration;
355  if (s->afq.frames->pts != AV_NOPTS_VALUE)
356  s->afq.frames->pts -= duration;
358  }
359  ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
360  }
361 
362  *got_packet_ptr = 1;
363  return 0;
364 }
365 
367  .name = "libvorbis",
368  .long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
369  .type = AVMEDIA_TYPE_AUDIO,
370  .id = AV_CODEC_ID_VORBIS,
371  .priv_data_size = sizeof(LibvorbisEncContext),
373  .encode2 = libvorbis_encode_frame,
374  .close = libvorbis_encode_close,
376  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
378  .priv_class = &vorbis_class,
379  .defaults = defaults,
380  .wrapper_name = "libvorbis",
381 };
float, planar
Definition: samplefmt.h:69
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
#define NULL
Definition: coverity.c:32
#define AV_CH_LAYOUT_7POINT1
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AudioFrameQueue afq
frame queue for timestamps
Definition: libvorbisenc.c:53
AVOption.
Definition: opt.h:246
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1620
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
Definition: libvorbisenc.c:174
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
channels
Definition: aptx.c:30
static const AVOption options[]
Definition: libvorbisenc.c:56
#define AV_OPT_FLAG_AUDIO_PARAM
Definition: opt.h:278
AVVorbisParseContext * av_vorbis_parse_init(const uint8_t *extradata, int extradata_size)
Allocate and initialize the Vorbis parser using headers in the extradata.
int size
Definition: avcodec.h:1483
AudioFrame * frames
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
AVFifoBuffer * pkt_fifo
output packet buffer
Definition: libvorbisenc.c:47
#define AV_CH_LAYOUT_STEREO
static void error(const char *err)
AVCodec.
Definition: avcodec.h:3494
#define AV_CH_LAYOUT_5POINT0
static int xiph_len(int l)
Definition: libvorbisenc.c:169
int av_fifo_generic_write(AVFifoBuffer *f, void *src, int size, int(*func)(void *, void *, int))
Feed data from a user-supplied callback to an AVFifoBuffer.
Definition: fifo.c:122
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:1011
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:32
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
double iblock
impulse block bias option
Definition: libvorbisenc.c:51
AVOptions.
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
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1500
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
int av_fifo_space(const AVFifoBuffer *f)
Return the amount of space in bytes in the AVFifoBuffer, that is the amount of data you can write int...
Definition: fifo.c:82
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1671
int64_t duration
Definition: movenc.c:63
AVClass * av_class
class for AVOptions
Definition: libvorbisenc.c:43
uint8_t * data
Definition: avcodec.h:1482
A public API for Vorbis parsing.
static const AVCodecDefault defaults[]
Definition: libvorbisenc.c:61
static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
Definition: libvorbisenc.c:195
AVVorbisParseContext * vp
parse context to get durations
Definition: libvorbisenc.c:52
static const uint8_t header[24]
Definition: sdr2.c:67
#define av_log(a,...)
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
Definition: opt.h:276
#define BUFFER_SIZE
Definition: libvorbisenc.c:40
static const AVClass vorbis_class
Definition: libvorbisenc.c:66
#define AV_CH_LAYOUT_5POINT1
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
void av_vorbis_parse_free(AVVorbisParseContext **s)
Free the parser and everything associated with it.
vorbis_comment vc
VorbisComment info.
Definition: libvorbisenc.c:50
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
int initial_padding
Audio only.
Definition: avcodec.h:3101
int av_fifo_generic_read(AVFifoBuffer *f, void *dest, int buf_size, void(*func)(void *, void *, int))
Feed data from an AVFifoBuffer to a user-supplied callback.
Definition: fifo.c:213
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1650
simple assert() macros that are a bit more flexible than ISO C assert().
#define AV_CH_LAYOUT_QUAD
const char * name
Name of the codec implementation.
Definition: avcodec.h:3501
AVCodec ff_libvorbis_encoder
Definition: libvorbisenc.c:366
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2281
static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libvorbisenc.c:274
#define AV_CH_LAYOUT_2_2
int64_t rc_min_rate
minimum bitrate
Definition: avcodec.h:2455
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:913
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:855
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:1016
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
#define s(width, name)
Definition: cbs_vp9.c:257
#define AV_CH_FRONT_CENTER
#define AV_CH_LAYOUT_5POINT1_BACK
vorbis_info vi
vorbis_info used during init
Definition: libvorbisenc.c:44
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.
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2250
const uint8_t ff_vorbis_encoding_channel_layout_offsets[8][8]
Definition: vorbis_data.c:36
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int av_fifo_size(const AVFifoBuffer *f)
Return the amount of data in bytes in the AVFifoBuffer, that is the amount of data you can read from ...
Definition: fifo.c:77
int sample_rate
samples per second
Definition: avcodec.h:2230
static int ogg_packet(AVFormatContext *s, int *sid, int *dstart, int *dsize, int64_t *fpos)
find the next Ogg packet
Definition: oggdec.c:477
main external API structure.
Definition: avcodec.h:1570
int dsp_initialized
vd has been initialized
Definition: libvorbisenc.c:49
a very simple circular buffer FIFO implementation
int eof
end-of-file flag
Definition: libvorbisenc.c:48
static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
Definition: libvorbisenc.c:83
int extradata_size
Definition: avcodec.h:1672
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
unsigned int av_xiphlacing(unsigned char *s, unsigned int v)
Encode extradata length to a buffer.
Definition: utils.c:1763
Describe the class of an AVClass context structure.
Definition: log.h:67
#define LIBVORBIS_FRAME_SIZE
Definition: libvorbisenc.c:38
#define AV_CH_LAYOUT_5POINT0_BACK
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1636
#define AV_CH_BACK_CENTER
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:78
static int vorbis_error_to_averror(int ov_err)
Definition: libvorbisenc.c:73
common internal api header.
int av_vorbis_parse_frame(AVVorbisParseContext *s, const uint8_t *buf, int buf_size)
Get the duration for a Vorbis packet.
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:71
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:795
void * priv_data
Definition: avcodec.h:1597
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:2274
AVFifoBuffer * av_fifo_alloc(unsigned int size)
Initialize an AVFifoBuffer.
Definition: fifo.c:43
int channels
number of audio channels
Definition: avcodec.h:2231
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:227
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define LIBAVCODEC_IDENT
Definition: version.h:42
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
Filter the word “frame” indicates either a video frame or a group of audio samples
#define av_freep(p)
void av_fifo_freep(AVFifoBuffer **f)
Free an AVFifoBuffer and reset pointer to NULL.
Definition: fifo.c:63
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:288
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
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
vorbis_dsp_state vd
DSP state used for analysis.
Definition: libvorbisenc.c:45
This structure stores compressed data.
Definition: avcodec.h:1459
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1475
vorbis_block vb
vorbis_block used for analysis
Definition: libvorbisenc.c:46
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
GLuint buffer
Definition: opengl_enc.c:101
int64_t rc_max_rate
maximum bitrate
Definition: avcodec.h:2448
const char * name
Definition: opengl_enc.c:102