FFmpeg
libmp3lame.c
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1 /*
2  * Interface to libmp3lame for mp3 encoding
3  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
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 /**
23  * @file
24  * Interface to libmp3lame for mp3 encoding.
25  */
26 
27 #include <lame/lame.h>
28 
30 #include "libavutil/common.h"
31 #include "libavutil/float_dsp.h"
32 #include "libavutil/intreadwrite.h"
33 #include "libavutil/log.h"
34 #include "libavutil/opt.h"
35 #include "avcodec.h"
36 #include "audio_frame_queue.h"
37 #include "internal.h"
38 #include "mpegaudio.h"
39 #include "mpegaudiodecheader.h"
40 
41 #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
42 
43 typedef struct LAMEContext {
44  AVClass *class;
46  lame_global_flags *gfp;
50  int reservoir;
52  int abr;
54  float *samples_flt[2];
57 } LAMEContext;
58 
59 
61 {
62  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
63  int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
64 
65  ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
66  new_size);
67  if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
68  s->buffer_size = s->buffer_index = 0;
69  return err;
70  }
71  s->buffer_size = new_size;
72  }
73  return 0;
74 }
75 
77 {
78  LAMEContext *s = avctx->priv_data;
79 
80  av_freep(&s->samples_flt[0]);
81  av_freep(&s->samples_flt[1]);
82  av_freep(&s->buffer);
83  av_freep(&s->fdsp);
84 
85  ff_af_queue_close(&s->afq);
86 
87  lame_close(s->gfp);
88  return 0;
89 }
90 
92 {
93  LAMEContext *s = avctx->priv_data;
94  int ret;
95 
96  s->avctx = avctx;
97 
98  /* initialize LAME and get defaults */
99  if (!(s->gfp = lame_init()))
100  return AVERROR(ENOMEM);
101 
102 
103  lame_set_num_channels(s->gfp, avctx->channels);
104  lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
105 
106  /* sample rate */
107  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
108  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
109 
110  /* algorithmic quality */
112  lame_set_quality(s->gfp, avctx->compression_level);
113 
114  /* rate control */
115  if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
116  lame_set_VBR(s->gfp, vbr_default);
117  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
118  } else {
119  if (avctx->bit_rate) {
120  if (s->abr) { // ABR
121  lame_set_VBR(s->gfp, vbr_abr);
122  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
123  } else // CBR
124  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
125  }
126  }
127 
128  /* lowpass cutoff frequency */
129  if (avctx->cutoff)
130  lame_set_lowpassfreq(s->gfp, avctx->cutoff);
131 
132  /* do not get a Xing VBR header frame from LAME */
133  lame_set_bWriteVbrTag(s->gfp,0);
134 
135  /* bit reservoir usage */
136  lame_set_disable_reservoir(s->gfp, !s->reservoir);
137 
138  /* set specified parameters */
139  if (lame_init_params(s->gfp) < 0) {
140  ret = -1;
141  goto error;
142  }
143 
144  /* get encoder delay */
145  avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
146  ff_af_queue_init(avctx, &s->afq);
147 
148  avctx->frame_size = lame_get_framesize(s->gfp);
149 
150  /* allocate float sample buffers */
151  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
152  int ch;
153  for (ch = 0; ch < avctx->channels; ch++) {
154  s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
155  sizeof(*s->samples_flt[ch]));
156  if (!s->samples_flt[ch]) {
157  ret = AVERROR(ENOMEM);
158  goto error;
159  }
160  }
161  }
162 
163  ret = realloc_buffer(s);
164  if (ret < 0)
165  goto error;
166 
168  if (!s->fdsp) {
169  ret = AVERROR(ENOMEM);
170  goto error;
171  }
172 
173 
174  return 0;
175 error:
176  mp3lame_encode_close(avctx);
177  return ret;
178 }
179 
180 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
181  lame_result = func(s->gfp, \
182  (const buf_type *)buf_name[0], \
183  (const buf_type *)buf_name[1], frame->nb_samples, \
184  s->buffer + s->buffer_index, \
185  s->buffer_size - s->buffer_index); \
186 } while (0)
187 
188 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
189  const AVFrame *frame, int *got_packet_ptr)
190 {
191  LAMEContext *s = avctx->priv_data;
192  MPADecodeHeader hdr;
193  int len, ret, ch, discard_padding;
194  int lame_result;
195  uint32_t h;
196 
197  if (frame) {
198  switch (avctx->sample_fmt) {
199  case AV_SAMPLE_FMT_S16P:
200  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
201  break;
202  case AV_SAMPLE_FMT_S32P:
203  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
204  break;
205  case AV_SAMPLE_FMT_FLTP:
206  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
207  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
208  return AVERROR(EINVAL);
209  }
210  for (ch = 0; ch < avctx->channels; ch++) {
211  s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
212  (const float *)frame->data[ch],
213  32768.0f,
214  FFALIGN(frame->nb_samples, 8));
215  }
216  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
217  break;
218  default:
219  return AVERROR_BUG;
220  }
221  } else if (!s->afq.frame_alloc) {
222  lame_result = 0;
223  } else {
224  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
225  s->buffer_size - s->buffer_index);
226  }
227  if (lame_result < 0) {
228  if (lame_result == -1) {
229  av_log(avctx, AV_LOG_ERROR,
230  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
231  s->buffer_index, s->buffer_size - s->buffer_index);
232  }
233  return -1;
234  }
235  s->buffer_index += lame_result;
236  ret = realloc_buffer(s);
237  if (ret < 0) {
238  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
239  return ret;
240  }
241 
242  /* add current frame to the queue */
243  if (frame) {
244  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
245  return ret;
246  }
247 
248  /* Move 1 frame from the LAME buffer to the output packet, if available.
249  We have to parse the first frame header in the output buffer to
250  determine the frame size. */
251  if (s->buffer_index < 4)
252  return 0;
253  h = AV_RB32(s->buffer);
254 
256  if (ret < 0) {
257  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
258  return AVERROR_BUG;
259  } else if (ret) {
260  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
261  return -1;
262  }
263  len = hdr.frame_size;
264  ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
265  s->buffer_index);
266  if (len <= s->buffer_index) {
267  if ((ret = ff_alloc_packet2(avctx, avpkt, len, 0)) < 0)
268  return ret;
269  memcpy(avpkt->data, s->buffer, len);
270  s->buffer_index -= len;
271  memmove(s->buffer, s->buffer + len, s->buffer_index);
272 
273  /* Get the next frame pts/duration */
274  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
275  &avpkt->duration);
276 
277  discard_padding = avctx->frame_size - avpkt->duration;
278  // Check if subtraction resulted in an overflow
279  if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
280  av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
281  av_packet_unref(avpkt);
282  return AVERROR(EINVAL);
283  }
284  if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
285  uint8_t* side_data = av_packet_new_side_data(avpkt,
287  10);
288  if(!side_data) {
289  av_packet_unref(avpkt);
290  return AVERROR(ENOMEM);
291  }
292  if (!s->delay_sent) {
293  AV_WL32(side_data, avctx->initial_padding);
294  s->delay_sent = 1;
295  }
296  AV_WL32(side_data + 4, discard_padding);
297  }
298 
299  avpkt->size = len;
300  *got_packet_ptr = 1;
301  }
302  return 0;
303 }
304 
305 #define OFFSET(x) offsetof(LAMEContext, x)
306 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
307 static const AVOption options[] = {
308  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
309  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
310  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
311  { NULL },
312 };
313 
314 static const AVClass libmp3lame_class = {
315  .class_name = "libmp3lame encoder",
316  .item_name = av_default_item_name,
317  .option = options,
318  .version = LIBAVUTIL_VERSION_INT,
319 };
320 
322  { "b", "0" },
323  { NULL },
324 };
325 
326 static const int libmp3lame_sample_rates[] = {
327  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
328 };
329 
331  .name = "libmp3lame",
332  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
333  .type = AVMEDIA_TYPE_AUDIO,
334  .id = AV_CODEC_ID_MP3,
335  .priv_data_size = sizeof(LAMEContext),
337  .encode2 = mp3lame_encode_frame,
338  .close = mp3lame_encode_close,
340  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
344  .supported_samplerates = libmp3lame_sample_rates,
345  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
347  0 },
348  .priv_class = &libmp3lame_class,
349  .defaults = libmp3lame_defaults,
350  .wrapper_name = "libmp3lame",
351 };
error
static void error(const char *err)
Definition: target_bsf_fuzzer.c:30
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1216
av_packet_unref
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:634
AVCodec
AVCodec.
Definition: codec.h:197
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
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
LAMEContext::buffer_size
int buffer_size
Definition: libmp3lame.c:49
JOINT_STEREO
#define JOINT_STEREO
Definition: atrac3.c:58
AV_WL32
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
BUFFER_SIZE
#define BUFFER_SIZE
Definition: libmp3lame.c:41
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:1196
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:925
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:90
AV_CODEC_FLAG_QSCALE
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:275
ff_libmp3lame_encoder
AVCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:330
LAMEContext
Definition: libmp3lame.c:43
ff_af_queue_init
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
Definition: audio_frame_queue.c:28
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
mpegaudiodecheader.h
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:369
AVOption
AVOption.
Definition: opt.h:248
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
MPADecodeHeader
Definition: mpegaudiodecheader.h:46
LAMEContext::gfp
lame_global_flags * gfp
Definition: libmp3lame.c:46
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:387
STEREO
#define STEREO
Definition: cook.c:62
FF_COMPRESSION_DEFAULT
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:609
audio_frame_queue.h
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:2062
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:616
AV_CODEC_ID_MP3
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: codec_id.h:425
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_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:91
AE
#define AE
Definition: libmp3lame.c:306
LAMEContext::samples_flt
float * samples_flt[2]
Definition: libmp3lame.c:54
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
avpriv_mpegaudio_decode_header
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
Definition: mpegaudiodecheader.c:36
av_cold
#define av_cold
Definition: attributes.h:90
LAMEContext::delay_sent
int delay_sent
Definition: libmp3lame.c:53
mp3lame_encode_init
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:91
intreadwrite.h
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVCodecContext::global_quality
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:602
frame_size
int frame_size
Definition: mxfenc.c:2206
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AudioFrameQueue
Definition: audio_frame_queue.h:32
realloc_buffer
static int realloc_buffer(LAMEContext *s)
Definition: libmp3lame.c:60
LAMEContext::buffer
uint8_t * buffer
Definition: libmp3lame.c:47
LAMEContext::fdsp
AVFloatDSPContext * fdsp
Definition: libmp3lame.c:56
int32_t
int32_t
Definition: audio_convert.c:194
libmp3lame_defaults
static const AVCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:321
AVCodecDefault
Definition: internal.h:222
LAMEContext::abr
int abr
Definition: libmp3lame.c:52
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
LAMEContext::buffer_index
int buffer_index
Definition: libmp3lame.c:48
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:586
OFFSET
#define OFFSET(x)
Definition: libmp3lame.c:305
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
float_dsp.h
ff_dlog
#define ff_dlog(a,...)
Definition: tableprint_vlc.h:29
libmp3lame_class
static const AVClass libmp3lame_class
Definition: libmp3lame.c:314
AVPacket::size
int size
Definition: packet.h:370
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
LAMEContext::avctx
AVCodecContext * avctx
Definition: libmp3lame.c:45
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
av_reallocp
int av_reallocp(void *ptr, size_t size)
Allocate, reallocate, or free a block of memory through a pointer to a pointer.
Definition: mem.c:161
ENCODE_BUFFER
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:180
AV_RB32
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:96
AVFloatDSPContext
Definition: float_dsp.h:24
LAMEContext::reservoir
int reservoir
Definition: libmp3lame.c:50
LAMEContext::afq
AudioFrameQueue afq
Definition: libmp3lame.c:55
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1197
log.h
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:362
mp3lame_encode_frame
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:188
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:32
common.h
AVCodecContext::cutoff
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:1240
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uint8_t
uint8_t
Definition: audio_convert.c:194
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:204
AV_PKT_DATA_SKIP_SAMPLES
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: packet.h:156
len
int len
Definition: vorbis_enc_data.h:452
mpegaudio.h
av_packet_new_side_data
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, buffer_size_t size)
Definition: avpacket.c:343
avcodec.h
MONO
#define MONO
Definition: cook.c:61
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:72
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
AVCodecContext
main external API structure.
Definition: avcodec.h:536
channel_layout.h
options
static const AVOption options[]
Definition: libmp3lame.c:307
libmp3lame_sample_rates
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:326
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:77
mp3lame_encode_close
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
Definition: libmp3lame.c:76
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:333
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:48
AVPacket
This structure stores compressed data.
Definition: packet.h:346
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:563
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
LAMEContext::joint_stereo
int joint_stereo
Definition: libmp3lame.c:51
h
h
Definition: vp9dsp_template.c:2038
FF_QP2LAMBDA
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:227
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:82
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: encode.c:33
AVCodecContext::compression_level
int compression_level
Definition: avcodec.h:608