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 "codec_internal.h"
38 #include "encode.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  int copyright;
59  int original;
60 } LAMEContext;
61 
62 
64 {
65  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
66  int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
67 
68  ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
69  new_size);
70  if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
71  s->buffer_size = s->buffer_index = 0;
72  return err;
73  }
74  s->buffer_size = new_size;
75  }
76  return 0;
77 }
78 
80 {
81  LAMEContext *s = avctx->priv_data;
82 
83  av_freep(&s->samples_flt[0]);
84  av_freep(&s->samples_flt[1]);
85  av_freep(&s->buffer);
86  av_freep(&s->fdsp);
87 
88  ff_af_queue_close(&s->afq);
89 
90  lame_close(s->gfp);
91  return 0;
92 }
93 
95 {
96  LAMEContext *s = avctx->priv_data;
97  int ret;
98 
99  s->avctx = avctx;
100 
101  /* initialize LAME and get defaults */
102  if (!(s->gfp = lame_init()))
103  return AVERROR(ENOMEM);
104 
105 
106  lame_set_num_channels(s->gfp, avctx->ch_layout.nb_channels);
107  lame_set_mode(s->gfp, avctx->ch_layout.nb_channels > 1 ?
108  s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
109 
110  /* sample rate */
111  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
112  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
113 
114  /* algorithmic quality */
116  lame_set_quality(s->gfp, avctx->compression_level);
117 
118  /* rate control */
119  if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
120  lame_set_VBR(s->gfp, vbr_default);
121  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
122  } else {
123  if (avctx->bit_rate) {
124  if (s->abr) { // ABR
125  lame_set_VBR(s->gfp, vbr_abr);
126  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
127  } else // CBR
128  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
129  }
130  }
131 
132  /* lowpass cutoff frequency */
133  if (avctx->cutoff)
134  lame_set_lowpassfreq(s->gfp, avctx->cutoff);
135 
136  /* do not get a Xing VBR header frame from LAME */
137  lame_set_bWriteVbrTag(s->gfp,0);
138 
139  /* bit reservoir usage */
140  lame_set_disable_reservoir(s->gfp, !s->reservoir);
141 
142  /* copyright flag */
143  lame_set_copyright(s->gfp, s->copyright);
144 
145  /* original flag */
146  lame_set_original(s->gfp, s->original);
147 
148  /* set specified parameters */
149  if (lame_init_params(s->gfp) < 0) {
151  goto error;
152  }
153 
154  /* get encoder delay */
155  avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
156  ff_af_queue_init(avctx, &s->afq);
157 
158  avctx->frame_size = lame_get_framesize(s->gfp);
159 
160  /* allocate float sample buffers */
161  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
162  int ch;
163  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
164  s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
165  sizeof(*s->samples_flt[ch]));
166  if (!s->samples_flt[ch]) {
167  ret = AVERROR(ENOMEM);
168  goto error;
169  }
170  }
171  }
172 
173  ret = realloc_buffer(s);
174  if (ret < 0)
175  goto error;
176 
178  if (!s->fdsp) {
179  ret = AVERROR(ENOMEM);
180  goto error;
181  }
182 
183 
184  return 0;
185 error:
186  mp3lame_encode_close(avctx);
187  return ret;
188 }
189 
190 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
191  lame_result = func(s->gfp, \
192  (const buf_type *)buf_name[0], \
193  (const buf_type *)buf_name[1], frame->nb_samples, \
194  s->buffer + s->buffer_index, \
195  s->buffer_size - s->buffer_index); \
196 } while (0)
197 
198 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
199  const AVFrame *frame, int *got_packet_ptr)
200 {
201  LAMEContext *s = avctx->priv_data;
202  MPADecodeHeader hdr;
203  int len, ret, ch, discard_padding;
204  int lame_result;
205  uint32_t h;
206 
207  if (frame) {
208  switch (avctx->sample_fmt) {
209  case AV_SAMPLE_FMT_S16P:
210  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
211  break;
212  case AV_SAMPLE_FMT_S32P:
213  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
214  break;
215  case AV_SAMPLE_FMT_FLTP:
216  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
217  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
218  return AVERROR(EINVAL);
219  }
220  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
221  s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
222  (const float *)frame->data[ch],
223  32768.0f,
224  FFALIGN(frame->nb_samples, 8));
225  }
226  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
227  break;
228  default:
229  return AVERROR_BUG;
230  }
231  } else if (!s->afq.frame_alloc) {
232  lame_result = 0;
233  } else {
234  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
235  s->buffer_size - s->buffer_index);
236  }
237  if (lame_result < 0) {
238  if (lame_result == -1) {
239  av_log(avctx, AV_LOG_ERROR,
240  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
241  s->buffer_index, s->buffer_size - s->buffer_index);
242  }
243  return AVERROR(ENOMEM);
244  }
245  s->buffer_index += lame_result;
246  ret = realloc_buffer(s);
247  if (ret < 0) {
248  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
249  return ret;
250  }
251 
252  /* add current frame to the queue */
253  if (frame) {
254  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
255  return ret;
256  }
257 
258  /* Move 1 frame from the LAME buffer to the output packet, if available.
259  We have to parse the first frame header in the output buffer to
260  determine the frame size. */
261  if (s->buffer_index < 4)
262  return 0;
263  h = AV_RB32(s->buffer);
264 
266  if (ret < 0) {
267  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
268  return AVERROR_BUG;
269  } else if (ret) {
270  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
271  return AVERROR_INVALIDDATA;
272  }
273  len = hdr.frame_size;
274  ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
275  s->buffer_index);
276  if (len <= s->buffer_index) {
277  if ((ret = ff_get_encode_buffer(avctx, avpkt, len, 0)) < 0)
278  return ret;
279  memcpy(avpkt->data, s->buffer, len);
280  s->buffer_index -= len;
281  memmove(s->buffer, s->buffer + len, s->buffer_index);
282 
283  /* Get the next frame pts/duration */
284  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
285  &avpkt->duration);
286 
287  discard_padding = avctx->frame_size - avpkt->duration;
288  // Check if subtraction resulted in an overflow
289  if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
290  av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
291  return AVERROR(EINVAL);
292  }
293  if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
294  uint8_t* side_data = av_packet_new_side_data(avpkt,
296  10);
297  if (!side_data)
298  return AVERROR(ENOMEM);
299  if (!s->delay_sent) {
300  AV_WL32(side_data, avctx->initial_padding);
301  s->delay_sent = 1;
302  }
303  AV_WL32(side_data + 4, discard_padding);
304  }
305 
306  *got_packet_ptr = 1;
307  }
308  return 0;
309 }
310 
311 #define OFFSET(x) offsetof(LAMEContext, x)
312 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
313 static const AVOption options[] = {
314  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
315  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
316  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
317  { "copyright", "set copyright flag", OFFSET(copyright), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE},
318  { "original", "set original flag", OFFSET(original), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE},
319  { NULL },
320 };
321 
322 static const AVClass libmp3lame_class = {
323  .class_name = "libmp3lame encoder",
324  .item_name = av_default_item_name,
325  .option = options,
326  .version = LIBAVUTIL_VERSION_INT,
327 };
328 
330  { "b", "0" },
331  { NULL },
332 };
333 
334 static const int libmp3lame_sample_rates[] = {
335  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
336 };
337 
339  .p.name = "libmp3lame",
340  CODEC_LONG_NAME("libmp3lame MP3 (MPEG audio layer 3)"),
341  .p.type = AVMEDIA_TYPE_AUDIO,
342  .p.id = AV_CODEC_ID_MP3,
343  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
345  .caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE,
346  .priv_data_size = sizeof(LAMEContext),
349  .close = mp3lame_encode_close,
350  .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
354  .p.supported_samplerates = libmp3lame_sample_rates,
356  .p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_MONO,
358  { 0 },
359  },
360  .p.priv_class = &libmp3lame_class,
361  .defaults = libmp3lame_defaults,
362  .p.wrapper_name = "libmp3lame",
363 };
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:1096
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
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:424
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
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition: channel_layout.h:389
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1068
ff_af_queue_close
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
Definition: audio_frame_queue.c:36
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:214
AV_CODEC_FLAG_QSCALE
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:224
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:340
mpegaudiodecheader.h
AVPacket::data
uint8_t * data
Definition: packet.h:522
AVOption
AVOption.
Definition: opt.h:251
encode.h
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:65
MPADecodeHeader
Definition: mpegaudiodecheader.h:47
LAMEContext::gfp
lame_global_flags * gfp
Definition: libmp3lame.c:47
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:34
FFCodec
Definition: codec_internal.h:127
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:540
STEREO
#define STEREO
Definition: cook.c:64
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:323
FF_COMPRESSION_DEFAULT
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:518
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:361
FFCodecDefault
Definition: codec_internal.h:97
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:131
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2111
audio_frame_queue.h
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:1798
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:525
AV_CODEC_ID_MP3
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: codec_id.h:444
FF_CODEC_ENCODE_CB
#define FF_CODEC_ENCODE_CB(func)
Definition: codec_internal.h:315
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:215
AE
#define AE
Definition: libmp3lame.c:312
LAMEContext::samples_flt
float * samples_flt[2]
Definition: libmp3lame.c:55
LAMEContext::original
int original
Definition: libmp3lame.c:59
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
libmp3lame_defaults
static const FFCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:329
mp3lame_encode_init
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:94
intreadwrite.h
s
#define s(width, name)
Definition: cbs_vp9.c:198
AVCodecContext::global_quality
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:511
frame_size
int frame_size
Definition: mxfenc.c:2422
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
CODEC_OLD_CHANNEL_LAYOUTS
#define CODEC_OLD_CHANNEL_LAYOUTS(...)
Definition: codec_internal.h:302
AudioFrameQueue
Definition: audio_frame_queue.h:32
realloc_buffer
static int realloc_buffer(LAMEContext *s)
Definition: libmp3lame.c:63
LAMEContext::buffer
uint8_t * buffer
Definition: libmp3lame.c:48
LAMEContext::fdsp
AVFloatDSPContext * fdsp
Definition: libmp3lame.c:57
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:272
frame
static AVFrame * frame
Definition: demux_decode.c:54
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:495
OFFSET
#define OFFSET(x)
Definition: libmp3lame.c:311
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:237
float_dsp.h
ff_dlog
#define ff_dlog(a,...)
Definition: tableprint_vlc.h:28
libmp3lame_class
static const AVClass libmp3lame_class
Definition: libmp3lame.c:322
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts.c:365
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
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:313
codec_internal.h
LAMEContext::avctx
AVCodecContext * avctx
Definition: libmp3lame.c:46
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1084
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
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:186
ENCODE_BUFFER
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:190
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::copyright
int copyright
Definition: libmp3lame.c:58
LAMEContext::reservoir
int reservoir
Definition: libmp3lame.c:51
LAMEContext::afq
AudioFrameQueue afq
Definition: libmp3lame.c:56
AVERROR_EXTERNAL
#define AVERROR_EXTERNAL
Generic error in an external library.
Definition: error.h:59
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:64
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:420
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:515
mp3lame_encode_frame
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:198
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:31
common.h
AVCodecContext::cutoff
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:1124
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:194
len
int len
Definition: vorbis_enc_data.h:426
mpegaudio.h
avcodec.h
MONO
#define MONO
Definition: cook.c:63
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
AVCodecContext
main external API structure.
Definition: avcodec.h:445
channel_layout.h
ff_libmp3lame_encoder
const FFCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:338
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:231
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:105
options
static const AVOption options[]
Definition: libmp3lame.c:313
libmp3lame_sample_rates
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:334
AV_PKT_DATA_SKIP_SAMPLES
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: packet.h:157
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:76
mp3lame_encode_close
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
Definition: libmp3lame.c:79
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:342
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:388
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
AVPacket
This structure stores compressed data.
Definition: packet.h:499
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:472
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:244
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
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
AVFrame::linesize
int linesize[AV_NUM_DATA_POINTERS]
For video, a positive or negative value, which is typically indicating the size in bytes of each pict...
Definition: frame.h:385
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
LAMEContext::joint_stereo
int joint_stereo
Definition: libmp3lame.c:52
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61
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:81
AVCodecContext::compression_level
int compression_level
Definition: avcodec.h:517