FFmpeg  4.4
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 
255  ret = avpriv_mpegaudio_decode_header(&hdr, h);
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 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
#define JOINT_STEREO
Definition: atrac3.c:58
#define av_cold
Definition: attributes.h:88
uint8_t
int32_t
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
Libavcodec external API header.
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:609
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, buffer_size_t size)
Definition: avpacket.c:343
#define AV_RB32
Definition: intreadwrite.h:130
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
#define s(width, name)
Definition: cbs_vp9.c:257
audio channel layout utility functions
common internal and external API header
#define STEREO
Definition: cook.c:62
#define MONO
Definition: cook.c:61
#define NULL
Definition: coverity.c:32
static AVFrame * frame
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
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_STEREO
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:333
#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
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:275
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: codec.h:82
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: codec_id.h:425
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:634
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: packet.h:156
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:227
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
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
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:91
#define AE
Definition: libmp3lame.c:306
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:180
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:326
static const AVOption options[]
Definition: libmp3lame.c:307
static const AVCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:321
#define BUFFER_SIZE
Definition: libmp3lame.c:41
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
Definition: libmp3lame.c:76
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:188
static int realloc_buffer(LAMEContext *s)
Definition: libmp3lame.c:60
#define OFFSET(x)
Definition: libmp3lame.c:305
static const AVClass libmp3lame_class
Definition: libmp3lame.c:314
AVCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:330
#define FFALIGN(x, a)
Definition: macros.h:48
mpeg audio declarations for both encoder and decoder.
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
MPEG Audio header decoder.
int frame_size
Definition: mxfenc.c:2206
AVOptions.
Describe the class of an AVClass context structure.
Definition: log.h:67
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
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:602
int64_t bit_rate
the average bitrate
Definition: avcodec.h:586
int initial_padding
Audio only.
Definition: avcodec.h:2062
int sample_rate
samples per second
Definition: avcodec.h:1196
int compression_level
Definition: avcodec.h:608
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:616
int channels
number of audio channels
Definition: avcodec.h:1197
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:1240
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1216
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:349
AVOption.
Definition: opt.h:248
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:387
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:362
uint8_t * data
Definition: packet.h:369
float * samples_flt[2]
Definition: libmp3lame.c:54
AVFloatDSPContext * fdsp
Definition: libmp3lame.c:56
lame_global_flags * gfp
Definition: libmp3lame.c:46
AudioFrameQueue afq
Definition: libmp3lame.c:55
AVCodecContext * avctx
Definition: libmp3lame.c:45
int delay_sent
Definition: libmp3lame.c:53
int reservoir
Definition: libmp3lame.c:50
int buffer_index
Definition: libmp3lame.c:48
int buffer_size
Definition: libmp3lame.c:49
int joint_stereo
Definition: libmp3lame.c:51
uint8_t * buffer
Definition: libmp3lame.c:47
#define av_malloc_array(a, b)
#define ff_dlog(a,...)
#define av_freep(p)
#define av_log(a,...)
static void error(const char *err)
int len