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 } 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->ch_layout.nb_channels);
105  lame_set_mode(s->gfp, avctx->ch_layout.nb_channels > 1 ?
106  s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
107 
108  /* sample rate */
109  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
110  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
111 
112  /* algorithmic quality */
114  lame_set_quality(s->gfp, avctx->compression_level);
115 
116  /* rate control */
117  if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
118  lame_set_VBR(s->gfp, vbr_default);
119  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
120  } else {
121  if (avctx->bit_rate) {
122  if (s->abr) { // ABR
123  lame_set_VBR(s->gfp, vbr_abr);
124  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
125  } else // CBR
126  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
127  }
128  }
129 
130  /* lowpass cutoff frequency */
131  if (avctx->cutoff)
132  lame_set_lowpassfreq(s->gfp, avctx->cutoff);
133 
134  /* do not get a Xing VBR header frame from LAME */
135  lame_set_bWriteVbrTag(s->gfp,0);
136 
137  /* bit reservoir usage */
138  lame_set_disable_reservoir(s->gfp, !s->reservoir);
139 
140  /* set specified parameters */
141  if (lame_init_params(s->gfp) < 0) {
143  goto error;
144  }
145 
146  /* get encoder delay */
147  avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
148  ff_af_queue_init(avctx, &s->afq);
149 
150  avctx->frame_size = lame_get_framesize(s->gfp);
151 
152  /* allocate float sample buffers */
153  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
154  int ch;
155  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
156  s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
157  sizeof(*s->samples_flt[ch]));
158  if (!s->samples_flt[ch]) {
159  ret = AVERROR(ENOMEM);
160  goto error;
161  }
162  }
163  }
164 
165  ret = realloc_buffer(s);
166  if (ret < 0)
167  goto error;
168 
170  if (!s->fdsp) {
171  ret = AVERROR(ENOMEM);
172  goto error;
173  }
174 
175 
176  return 0;
177 error:
178  mp3lame_encode_close(avctx);
179  return ret;
180 }
181 
182 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
183  lame_result = func(s->gfp, \
184  (const buf_type *)buf_name[0], \
185  (const buf_type *)buf_name[1], frame->nb_samples, \
186  s->buffer + s->buffer_index, \
187  s->buffer_size - s->buffer_index); \
188 } while (0)
189 
190 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
191  const AVFrame *frame, int *got_packet_ptr)
192 {
193  LAMEContext *s = avctx->priv_data;
194  MPADecodeHeader hdr;
195  int len, ret, ch, discard_padding;
196  int lame_result;
197  uint32_t h;
198 
199  if (frame) {
200  switch (avctx->sample_fmt) {
201  case AV_SAMPLE_FMT_S16P:
202  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
203  break;
204  case AV_SAMPLE_FMT_S32P:
205  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
206  break;
207  case AV_SAMPLE_FMT_FLTP:
208  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
209  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
210  return AVERROR(EINVAL);
211  }
212  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
213  s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
214  (const float *)frame->data[ch],
215  32768.0f,
216  FFALIGN(frame->nb_samples, 8));
217  }
218  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
219  break;
220  default:
221  return AVERROR_BUG;
222  }
223  } else if (!s->afq.frame_alloc) {
224  lame_result = 0;
225  } else {
226  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
227  s->buffer_size - s->buffer_index);
228  }
229  if (lame_result < 0) {
230  if (lame_result == -1) {
231  av_log(avctx, AV_LOG_ERROR,
232  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
233  s->buffer_index, s->buffer_size - s->buffer_index);
234  }
235  return AVERROR(ENOMEM);
236  }
237  s->buffer_index += lame_result;
238  ret = realloc_buffer(s);
239  if (ret < 0) {
240  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
241  return ret;
242  }
243 
244  /* add current frame to the queue */
245  if (frame) {
246  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
247  return ret;
248  }
249 
250  /* Move 1 frame from the LAME buffer to the output packet, if available.
251  We have to parse the first frame header in the output buffer to
252  determine the frame size. */
253  if (s->buffer_index < 4)
254  return 0;
255  h = AV_RB32(s->buffer);
256 
258  if (ret < 0) {
259  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
260  return AVERROR_BUG;
261  } else if (ret) {
262  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
263  return AVERROR_INVALIDDATA;
264  }
265  len = hdr.frame_size;
266  ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
267  s->buffer_index);
268  if (len <= s->buffer_index) {
269  if ((ret = ff_get_encode_buffer(avctx, avpkt, len, 0)) < 0)
270  return ret;
271  memcpy(avpkt->data, s->buffer, len);
272  s->buffer_index -= len;
273  memmove(s->buffer, s->buffer + len, s->buffer_index);
274 
275  /* Get the next frame pts/duration */
276  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
277  &avpkt->duration);
278 
279  discard_padding = avctx->frame_size - avpkt->duration;
280  // Check if subtraction resulted in an overflow
281  if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
282  av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
283  av_packet_unref(avpkt);
284  return AVERROR(EINVAL);
285  }
286  if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
287  uint8_t* side_data = av_packet_new_side_data(avpkt,
289  10);
290  if(!side_data) {
291  av_packet_unref(avpkt);
292  return AVERROR(ENOMEM);
293  }
294  if (!s->delay_sent) {
295  AV_WL32(side_data, avctx->initial_padding);
296  s->delay_sent = 1;
297  }
298  AV_WL32(side_data + 4, discard_padding);
299  }
300 
301  *got_packet_ptr = 1;
302  }
303  return 0;
304 }
305 
306 #define OFFSET(x) offsetof(LAMEContext, x)
307 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
308 static const AVOption options[] = {
309  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
310  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
311  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
312  { NULL },
313 };
314 
315 static const AVClass libmp3lame_class = {
316  .class_name = "libmp3lame encoder",
317  .item_name = av_default_item_name,
318  .option = options,
319  .version = LIBAVUTIL_VERSION_INT,
320 };
321 
323  { "b", "0" },
324  { NULL },
325 };
326 
327 static const int libmp3lame_sample_rates[] = {
328  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
329 };
330 
332  .p.name = "libmp3lame",
333  CODEC_LONG_NAME("libmp3lame MP3 (MPEG audio layer 3)"),
334  .p.type = AVMEDIA_TYPE_AUDIO,
335  .p.id = AV_CODEC_ID_MP3,
336  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
338  .caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE,
339  .priv_data_size = sizeof(LAMEContext),
342  .close = mp3lame_encode_close,
343  .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
347  .p.supported_samplerates = libmp3lame_sample_rates,
349  .p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_MONO,
351  { 0 },
352  },
353  .p.priv_class = &libmp3lame_class,
354  .defaults = libmp3lame_defaults,
355  .p.wrapper_name = "libmp3lame",
356 };
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:1035
av_packet_unref
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:422
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: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
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition: channel_layout.h:369
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1007
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:210
AV_CODEC_FLAG_QSCALE
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:216
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:325
mpegaudiodecheader.h
AVPacket::data
uint8_t * data
Definition: packet.h:374
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:119
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:392
STEREO
#define STEREO
Definition: cook.c:64
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:311
FF_COMPRESSION_DEFAULT
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:471
FFCodecDefault
Definition: codec_internal.h:89
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:123
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2059
audio_frame_queue.h
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:1726
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:478
AV_CODEC_ID_MP3
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: codec_id.h:436
FF_CODEC_ENCODE_CB
#define FF_CODEC_ENCODE_CB(func)
Definition: codec_internal.h:307
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:211
AE
#define AE
Definition: libmp3lame.c:307
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
libmp3lame_defaults
static const FFCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:322
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:256
AVCodecContext::global_quality
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:464
frame_size
int frame_size
Definition: mxfenc.c:2202
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts_bsf.c:363
CODEC_OLD_CHANNEL_LAYOUTS
#define CODEC_OLD_CHANNEL_LAYOUTS(...)
Definition: codec_internal.h:294
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
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:264
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:448
OFFSET
#define OFFSET(x)
Definition: libmp3lame.c:306
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:315
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:301
codec_internal.h
LAMEContext::avctx
AVCodecContext * avctx
Definition: libmp3lame.c:46
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1023
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:182
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
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:367
mp3lame_encode_frame
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:190
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:1059
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:211
AV_PKT_DATA_SKIP_SAMPLES
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: packet.h:157
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
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:398
channel_layout.h
ff_libmp3lame_encoder
const FFCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:331
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:230
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:79
options
static const AVOption options[]
Definition: libmp3lame.c:308
libmp3lame_sample_rates
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:327
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:288
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:368
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
AVPacket
This structure stores compressed data.
Definition: packet.h:351
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:425
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
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:87
AVCodecContext::compression_level
int compression_level
Definition: avcodec.h:470