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 "encode.h"
38 #include "internal.h"
39 #include "mpegaudio.h"
40 #include "mpegaudiodecheader.h"
41 
42 #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.
43 
44 typedef struct LAMEContext {
45  AVClass *class;
47  lame_global_flags *gfp;
48  uint8_t *buffer;
51  int reservoir;
53  int abr;
55  float *samples_flt[2];
58 } LAMEContext;
59 
60 
62 {
63  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
64  int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
65 
66  ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
67  new_size);
68  if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
69  s->buffer_size = s->buffer_index = 0;
70  return err;
71  }
72  s->buffer_size = new_size;
73  }
74  return 0;
75 }
76 
78 {
79  LAMEContext *s = avctx->priv_data;
80 
81  av_freep(&s->samples_flt[0]);
82  av_freep(&s->samples_flt[1]);
83  av_freep(&s->buffer);
84  av_freep(&s->fdsp);
85 
86  ff_af_queue_close(&s->afq);
87 
88  lame_close(s->gfp);
89  return 0;
90 }
91 
93 {
94  LAMEContext *s = avctx->priv_data;
95  int ret;
96 
97  s->avctx = avctx;
98 
99  /* initialize LAME and get defaults */
100  if (!(s->gfp = lame_init()))
101  return AVERROR(ENOMEM);
102 
103 
104  lame_set_num_channels(s->gfp, avctx->channels);
105  lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
106 
107  /* sample rate */
108  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
109  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
110 
111  /* algorithmic quality */
113  lame_set_quality(s->gfp, avctx->compression_level);
114 
115  /* rate control */
116  if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
117  lame_set_VBR(s->gfp, vbr_default);
118  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
119  } else {
120  if (avctx->bit_rate) {
121  if (s->abr) { // ABR
122  lame_set_VBR(s->gfp, vbr_abr);
123  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
124  } else // CBR
125  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
126  }
127  }
128 
129  /* lowpass cutoff frequency */
130  if (avctx->cutoff)
131  lame_set_lowpassfreq(s->gfp, avctx->cutoff);
132 
133  /* do not get a Xing VBR header frame from LAME */
134  lame_set_bWriteVbrTag(s->gfp,0);
135 
136  /* bit reservoir usage */
137  lame_set_disable_reservoir(s->gfp, !s->reservoir);
138 
139  /* set specified parameters */
140  if (lame_init_params(s->gfp) < 0) {
141  ret = -1;
142  goto error;
143  }
144 
145  /* get encoder delay */
146  avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
147  ff_af_queue_init(avctx, &s->afq);
148 
149  avctx->frame_size = lame_get_framesize(s->gfp);
150 
151  /* allocate float sample buffers */
152  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
153  int ch;
154  for (ch = 0; ch < avctx->channels; ch++) {
155  s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
156  sizeof(*s->samples_flt[ch]));
157  if (!s->samples_flt[ch]) {
158  ret = AVERROR(ENOMEM);
159  goto error;
160  }
161  }
162  }
163 
164  ret = realloc_buffer(s);
165  if (ret < 0)
166  goto error;
167 
169  if (!s->fdsp) {
170  ret = AVERROR(ENOMEM);
171  goto error;
172  }
173 
174 
175  return 0;
176 error:
177  mp3lame_encode_close(avctx);
178  return ret;
179 }
180 
181 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
182  lame_result = func(s->gfp, \
183  (const buf_type *)buf_name[0], \
184  (const buf_type *)buf_name[1], frame->nb_samples, \
185  s->buffer + s->buffer_index, \
186  s->buffer_size - s->buffer_index); \
187 } while (0)
188 
189 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
190  const AVFrame *frame, int *got_packet_ptr)
191 {
192  LAMEContext *s = avctx->priv_data;
193  MPADecodeHeader hdr;
194  int len, ret, ch, discard_padding;
195  int lame_result;
196  uint32_t h;
197 
198  if (frame) {
199  switch (avctx->sample_fmt) {
200  case AV_SAMPLE_FMT_S16P:
201  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
202  break;
203  case AV_SAMPLE_FMT_S32P:
204  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
205  break;
206  case AV_SAMPLE_FMT_FLTP:
207  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
208  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
209  return AVERROR(EINVAL);
210  }
211  for (ch = 0; ch < avctx->channels; ch++) {
212  s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
213  (const float *)frame->data[ch],
214  32768.0f,
215  FFALIGN(frame->nb_samples, 8));
216  }
217  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
218  break;
219  default:
220  return AVERROR_BUG;
221  }
222  } else if (!s->afq.frame_alloc) {
223  lame_result = 0;
224  } else {
225  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
226  s->buffer_size - s->buffer_index);
227  }
228  if (lame_result < 0) {
229  if (lame_result == -1) {
230  av_log(avctx, AV_LOG_ERROR,
231  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
232  s->buffer_index, s->buffer_size - s->buffer_index);
233  }
234  return -1;
235  }
236  s->buffer_index += lame_result;
237  ret = realloc_buffer(s);
238  if (ret < 0) {
239  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
240  return ret;
241  }
242 
243  /* add current frame to the queue */
244  if (frame) {
245  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
246  return ret;
247  }
248 
249  /* Move 1 frame from the LAME buffer to the output packet, if available.
250  We have to parse the first frame header in the output buffer to
251  determine the frame size. */
252  if (s->buffer_index < 4)
253  return 0;
254  h = AV_RB32(s->buffer);
255 
257  if (ret < 0) {
258  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
259  return AVERROR_BUG;
260  } else if (ret) {
261  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
262  return -1;
263  }
264  len = hdr.frame_size;
265  ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
266  s->buffer_index);
267  if (len <= s->buffer_index) {
268  if ((ret = ff_get_encode_buffer(avctx, avpkt, len, 0)) < 0)
269  return ret;
270  memcpy(avpkt->data, s->buffer, len);
271  s->buffer_index -= len;
272  memmove(s->buffer, s->buffer + len, s->buffer_index);
273 
274  /* Get the next frame pts/duration */
275  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
276  &avpkt->duration);
277 
278  discard_padding = avctx->frame_size - avpkt->duration;
279  // Check if subtraction resulted in an overflow
280  if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
281  av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
282  av_packet_unref(avpkt);
283  return AVERROR(EINVAL);
284  }
285  if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
286  uint8_t* side_data = av_packet_new_side_data(avpkt,
288  10);
289  if(!side_data) {
290  av_packet_unref(avpkt);
291  return AVERROR(ENOMEM);
292  }
293  if (!s->delay_sent) {
294  AV_WL32(side_data, avctx->initial_padding);
295  s->delay_sent = 1;
296  }
297  AV_WL32(side_data + 4, discard_padding);
298  }
299 
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  .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
337  .priv_data_size = sizeof(LAMEContext),
339  .encode2 = mp3lame_encode_frame,
340  .close = mp3lame_encode_close,
341  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
345  .supported_samplerates = libmp3lame_sample_rates,
346  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
348  0 },
349  .priv_class = &libmp3lame_class,
350  .defaults = libmp3lame_defaults,
351  .wrapper_name = "libmp3lame",
352 };
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:1012
av_packet_unref
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:417
AVCodec
AVCodec.
Definition: codec.h:202
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:50
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:42
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:992
ff_af_queue_close
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
Definition: audio_frame_queue.c:36
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:948
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:210
LAMEContext
Definition: libmp3lame.c:44
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:303
mpegaudiodecheader.h
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:373
AVOption
AVOption.
Definition: opt.h:247
encode.h
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
MPADecodeHeader
Definition: mpegaudiodecheader.h:47
ff_libmp3lame_encoder
const AVCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:330
LAMEContext::gfp
lame_global_flags * gfp
Definition: libmp3lame.c:47
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:391
STEREO
#define STEREO
Definition: cook.c:63
FF_COMPRESSION_DEFAULT
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:456
audio_frame_queue.h
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:1701
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:463
AV_CODEC_ID_MP3
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: codec_id.h:423
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:55
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
avpriv_mpegaudio_decode_header
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
Definition: mpegaudiodecheader.c:34
av_cold
#define av_cold
Definition: attributes.h:90
LAMEContext::delay_sent
int delay_sent
Definition: libmp3lame.c:54
mp3lame_encode_init
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:92
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:449
frame_size
int frame_size
Definition: mxfenc.c:2207
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:61
LAMEContext::buffer
uint8_t * buffer
Definition: libmp3lame.c:48
LAMEContext::fdsp
AVFloatDSPContext * fdsp
Definition: libmp3lame.c:57
libmp3lame_defaults
static const AVCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:321
AVCodecDefault
Definition: internal.h:206
LAMEContext::abr
int abr
Definition: libmp3lame.c:53
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
LAMEContext::buffer_index
int buffer_index
Definition: libmp3lame.c:49
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:433
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
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
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:46
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1000
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
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:185
ENCODE_BUFFER
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:181
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:51
LAMEContext::afq
AudioFrameQueue afq
Definition: libmp3lame.c:56
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:993
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:366
mp3lame_encode_frame
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:189
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:1036
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:209
AV_PKT_DATA_SKIP_SAMPLES
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: packet.h:156
len
int len
Definition: vorbis_enc_data.h:426
mpegaudio.h
avcodec.h
MONO
#define MONO
Definition: cook.c:62
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
AVCodecContext
main external API structure.
Definition: avcodec.h:383
channel_layout.h
av_packet_new_side_data
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, size_t size)
Allocate new information of a packet.
Definition: avpacket.c:225
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:78
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:82
mp3lame_encode_close
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
Definition: libmp3lame.c:77
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:272
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
AVPacket
This structure stores compressed data.
Definition: packet.h:350
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:410
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:241
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
int32_t
int32_t
Definition: audioconvert.c:56
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:28
LAMEContext::joint_stereo
int joint_stereo
Definition: libmp3lame.c:52
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:87
AVCodecContext::compression_level
int compression_level
Definition: avcodec.h:455