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/mem.h"
35 #include "libavutil/opt.h"
36 #include "avcodec.h"
37 #include "audio_frame_queue.h"
38 #include "codec_internal.h"
39 #include "encode.h"
40 #include "mpegaudio.h"
41 #include "mpegaudiodecheader.h"
42 
43 #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.
44 
45 typedef struct LAMEContext {
46  AVClass *class;
48  lame_global_flags *gfp;
49  uint8_t *buffer;
52  int reservoir;
54  int abr;
56  float *samples_flt[2];
59  int copyright;
60  int original;
61 } LAMEContext;
62 
63 
65 {
66  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
67  int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
68 
69  ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
70  new_size);
71  if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
72  s->buffer_size = s->buffer_index = 0;
73  return err;
74  }
75  s->buffer_size = new_size;
76  }
77  return 0;
78 }
79 
81 {
82  LAMEContext *s = avctx->priv_data;
83 
84  av_freep(&s->samples_flt[0]);
85  av_freep(&s->samples_flt[1]);
86  av_freep(&s->buffer);
87  av_freep(&s->fdsp);
88 
89  ff_af_queue_close(&s->afq);
90 
91  lame_close(s->gfp);
92  return 0;
93 }
94 
96 {
97  LAMEContext *s = avctx->priv_data;
98  int ret;
99 
100  s->avctx = avctx;
101 
102  /* initialize LAME and get defaults */
103  if (!(s->gfp = lame_init()))
104  return AVERROR(ENOMEM);
105 
106 
107  lame_set_num_channels(s->gfp, avctx->ch_layout.nb_channels);
108  lame_set_mode(s->gfp, avctx->ch_layout.nb_channels > 1 ?
109  s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
110 
111  /* sample rate */
112  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
113  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
114 
115  /* algorithmic quality */
117  lame_set_quality(s->gfp, avctx->compression_level);
118 
119  /* rate control */
120  if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
121  lame_set_VBR(s->gfp, vbr_default);
122  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
123  } else {
124  if (avctx->bit_rate) {
125  if (s->abr) { // ABR
126  lame_set_VBR(s->gfp, vbr_abr);
127  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
128  } else // CBR
129  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
130  }
131  }
132 
133  /* lowpass cutoff frequency */
134  if (avctx->cutoff)
135  lame_set_lowpassfreq(s->gfp, avctx->cutoff);
136 
137  /* do not get a Xing VBR header frame from LAME */
138  lame_set_bWriteVbrTag(s->gfp,0);
139 
140  /* bit reservoir usage */
141  lame_set_disable_reservoir(s->gfp, !s->reservoir);
142 
143  /* copyright flag */
144  lame_set_copyright(s->gfp, s->copyright);
145 
146  /* original flag */
147  lame_set_original(s->gfp, s->original);
148 
149  /* set specified parameters */
150  if (lame_init_params(s->gfp) < 0) {
152  goto error;
153  }
154 
155  /* get encoder delay */
156  avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
157  ff_af_queue_init(avctx, &s->afq);
158 
159  avctx->frame_size = lame_get_framesize(s->gfp);
160 
161  /* allocate float sample buffers */
162  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
163  int ch;
164  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
165  s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
166  sizeof(*s->samples_flt[ch]));
167  if (!s->samples_flt[ch]) {
168  ret = AVERROR(ENOMEM);
169  goto error;
170  }
171  }
172  }
173 
174  ret = realloc_buffer(s);
175  if (ret < 0)
176  goto error;
177 
179  if (!s->fdsp) {
180  ret = AVERROR(ENOMEM);
181  goto error;
182  }
183 
184 
185  return 0;
186 error:
187  mp3lame_encode_close(avctx);
188  return ret;
189 }
190 
191 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
192  lame_result = func(s->gfp, \
193  (const buf_type *)buf_name[0], \
194  (const buf_type *)buf_name[1], frame->nb_samples, \
195  s->buffer + s->buffer_index, \
196  s->buffer_size - s->buffer_index); \
197 } while (0)
198 
199 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
200  const AVFrame *frame, int *got_packet_ptr)
201 {
202  LAMEContext *s = avctx->priv_data;
203  MPADecodeHeader hdr;
204  int len, ret, ch, discard_padding;
205  int lame_result;
206  uint32_t h;
207 
208  if (frame) {
209  switch (avctx->sample_fmt) {
210  case AV_SAMPLE_FMT_S16P:
211  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
212  break;
213  case AV_SAMPLE_FMT_S32P:
214  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
215  break;
216  case AV_SAMPLE_FMT_FLTP:
217  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
218  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
219  return AVERROR(EINVAL);
220  }
221  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
222  s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
223  (const float *)frame->data[ch],
224  32768.0f,
225  FFALIGN(frame->nb_samples, 8));
226  }
227  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
228  break;
229  default:
230  return AVERROR_BUG;
231  }
232  } else if (!s->afq.frame_alloc) {
233  lame_result = 0;
234  } else {
235  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
236  s->buffer_size - s->buffer_index);
237  }
238  if (lame_result < 0) {
239  if (lame_result == -1) {
240  av_log(avctx, AV_LOG_ERROR,
241  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
242  s->buffer_index, s->buffer_size - s->buffer_index);
243  }
244  return AVERROR(ENOMEM);
245  }
246  s->buffer_index += lame_result;
247  ret = realloc_buffer(s);
248  if (ret < 0) {
249  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
250  return ret;
251  }
252 
253  /* add current frame to the queue */
254  if (frame) {
255  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
256  return ret;
257  }
258 
259  /* Move 1 frame from the LAME buffer to the output packet, if available.
260  We have to parse the first frame header in the output buffer to
261  determine the frame size. */
262  if (s->buffer_index < 4)
263  return 0;
264  h = AV_RB32(s->buffer);
265 
267  if (ret < 0) {
268  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
269  return AVERROR_BUG;
270  } else if (ret) {
271  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
272  return AVERROR_INVALIDDATA;
273  }
274  len = hdr.frame_size;
275  ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
276  s->buffer_index);
277  if (len <= s->buffer_index) {
278  if ((ret = ff_get_encode_buffer(avctx, avpkt, len, 0)) < 0)
279  return ret;
280  memcpy(avpkt->data, s->buffer, len);
281  s->buffer_index -= len;
282  memmove(s->buffer, s->buffer + len, s->buffer_index);
283 
284  /* Get the next frame pts/duration */
285  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
286  &avpkt->duration);
287 
288  discard_padding = avctx->frame_size - avpkt->duration;
289  // Check if subtraction resulted in an overflow
290  if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
291  av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
292  return AVERROR(EINVAL);
293  }
294  if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
295  uint8_t* side_data = av_packet_new_side_data(avpkt,
297  10);
298  if (!side_data)
299  return AVERROR(ENOMEM);
300  if (!s->delay_sent) {
301  AV_WL32(side_data, avctx->initial_padding);
302  s->delay_sent = 1;
303  }
304  AV_WL32(side_data + 4, discard_padding);
305  }
306 
307  *got_packet_ptr = 1;
308  }
309  return 0;
310 }
311 
312 #define OFFSET(x) offsetof(LAMEContext, x)
313 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
314 static const AVOption options[] = {
315  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
316  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
317  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
318  { "copyright", "set copyright flag", OFFSET(copyright), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE},
319  { "original", "set original flag", OFFSET(original), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE},
320  { NULL },
321 };
322 
323 static const AVClass libmp3lame_class = {
324  .class_name = "libmp3lame encoder",
325  .item_name = av_default_item_name,
326  .option = options,
327  .version = LIBAVUTIL_VERSION_INT,
328 };
329 
331  { "b", "0" },
332  { NULL },
333 };
334 
335 static const int libmp3lame_sample_rates[] = {
336  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
337 };
338 
340  .p.name = "libmp3lame",
341  CODEC_LONG_NAME("libmp3lame MP3 (MPEG audio layer 3)"),
342  .p.type = AVMEDIA_TYPE_AUDIO,
343  .p.id = AV_CODEC_ID_MP3,
344  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
346  .caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE,
347  .priv_data_size = sizeof(LAMEContext),
350  .close = mp3lame_encode_close,
351  .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
355  .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:32
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1077
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:51
JOINT_STEREO
#define JOINT_STEREO
Definition: atrac3.c:59
AV_WL32
#define AV_WL32(p, v)
Definition: intreadwrite.h:420
BUFFER_SIZE
#define BUFFER_SIZE
Definition: libmp3lame.c: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
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition: channel_layout.h:387
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1050
ff_af_queue_close
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
Definition: audio_frame_queue.c:36
AV_CODEC_FLAG_QSCALE
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:224
LAMEContext
Definition: libmp3lame.c:45
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:374
mpegaudiodecheader.h
AVPacket::data
uint8_t * data
Definition: packet.h:520
AVOption
AVOption.
Definition: opt.h:357
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:48
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:126
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:538
STEREO
#define STEREO
Definition: cook.c:65
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:321
FF_COMPRESSION_DEFAULT
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:1246
FFCodecDefault
Definition: codec_internal.h:96
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:130
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:1065
audio_frame_queue.h
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:1122
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:502
AV_CODEC_ID_MP3
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: codec_id.h:441
FF_CODEC_ENCODE_CB
#define FF_CODEC_ENCODE_CB(func)
Definition: codec_internal.h:295
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
AE
#define AE
Definition: libmp3lame.c:313
LAMEContext::samples_flt
float * samples_flt[2]
Definition: libmp3lame.c:56
LAMEContext::original
int original
Definition: libmp3lame.c:60
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:55
libmp3lame_defaults
static const FFCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:330
mp3lame_encode_init
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:95
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:1239
frame_size
int frame_size
Definition: mxfenc.c:2424
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:64
LAMEContext::buffer
uint8_t * buffer
Definition: libmp3lame.c:49
LAMEContext::fdsp
AVFloatDSPContext * fdsp
Definition: libmp3lame.c:58
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:271
LAMEContext::abr
int abr
Definition: libmp3lame.c:54
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:50
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:495
OFFSET
#define OFFSET(x)
Definition: libmp3lame.c:312
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:323
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts.c:366
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:311
codec_internal.h
LAMEContext::avctx
AVCodecContext * avctx
Definition: libmp3lame.c:47
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1057
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:188
ENCODE_BUFFER
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:191
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:59
LAMEContext::reservoir
int reservoir
Definition: libmp3lame.c:52
LAMEContext::afq
AudioFrameQueue afq
Definition: libmp3lame.c:57
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
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:513
mp3lame_encode_frame
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:199
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:1090
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:64
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:445
channel_layout.h
ff_libmp3lame_encoder
const FFCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:339
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: packet.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:106
options
static const AVOption options[]
Definition: libmp3lame.c:314
libmp3lame_sample_rates
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:335
AV_PKT_DATA_SKIP_SAMPLES
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: packet.h:153
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:80
mem.h
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:386
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
AVPacket
This structure stores compressed data.
Definition: packet.h:497
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:472
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:261
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:146
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:27
LAMEContext::joint_stereo
int joint_stereo
Definition: libmp3lame.c:53
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:1245