Libav
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 Libav.
6  *
7  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
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)
42 
43 typedef struct LAMEContext {
44  AVClass *class;
46  lame_global_flags *gfp;
50  int reservoir;
52  int abr;
53  float *samples_flt[2];
56 } LAMEContext;
57 
58 
60 {
61  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
62  int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
63 
64  av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
65  new_size);
66  if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
67  s->buffer_size = s->buffer_index = 0;
68  return err;
69  }
70  s->buffer_size = new_size;
71  }
72  return 0;
73 }
74 
76 {
77  LAMEContext *s = avctx->priv_data;
78 
79  av_freep(&s->samples_flt[0]);
80  av_freep(&s->samples_flt[1]);
81  av_freep(&s->buffer);
82 
84 
85  lame_close(s->gfp);
86  return 0;
87 }
88 
90 {
91  LAMEContext *s = avctx->priv_data;
92  int ret;
93 
94  s->avctx = avctx;
95 
96  /* initialize LAME and get defaults */
97  if (!(s->gfp = lame_init()))
98  return AVERROR(ENOMEM);
99 
100  lame_set_num_channels(s->gfp, avctx->channels);
101  lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
102 
103  /* sample rate */
104  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
105  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
106 
107  /* algorithmic quality */
109  lame_set_quality(s->gfp, 5);
110  else
111  lame_set_quality(s->gfp, avctx->compression_level);
112 
113  /* rate control */
114  if (avctx->flags & CODEC_FLAG_QSCALE) { // VBR
115  lame_set_VBR(s->gfp, vbr_default);
116  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
117  } else {
118  if (avctx->bit_rate) {
119  if (s->abr) { // ABR
120  lame_set_VBR(s->gfp, vbr_abr);
121  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
122  } else // CBR
123  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
124  }
125  }
126 
127  /* do not get a Xing VBR header frame from LAME */
128  lame_set_bWriteVbrTag(s->gfp,0);
129 
130  /* bit reservoir usage */
131  lame_set_disable_reservoir(s->gfp, !s->reservoir);
132 
133  /* set specified parameters */
134  if (lame_init_params(s->gfp) < 0) {
135  ret = -1;
136  goto error;
137  }
138 
139  /* get encoder delay */
140  avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
141  ff_af_queue_init(avctx, &s->afq);
142 
143  avctx->frame_size = lame_get_framesize(s->gfp);
144 
145  /* allocate float sample buffers */
146  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
147  int ch;
148  for (ch = 0; ch < avctx->channels; ch++) {
149  s->samples_flt[ch] = av_malloc(avctx->frame_size *
150  sizeof(*s->samples_flt[ch]));
151  if (!s->samples_flt[ch]) {
152  ret = AVERROR(ENOMEM);
153  goto error;
154  }
155  }
156  }
157 
158  ret = realloc_buffer(s);
159  if (ret < 0)
160  goto error;
161 
163 
164  return 0;
165 error:
166  mp3lame_encode_close(avctx);
167  return ret;
168 }
169 
170 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
171  lame_result = func(s->gfp, \
172  (const buf_type *)buf_name[0], \
173  (const buf_type *)buf_name[1], frame->nb_samples, \
174  s->buffer + s->buffer_index, \
175  s->buffer_size - s->buffer_index); \
176 } while (0)
177 
178 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
179  const AVFrame *frame, int *got_packet_ptr)
180 {
181  LAMEContext *s = avctx->priv_data;
182  MPADecodeHeader hdr;
183  int len, ret, ch;
184  int lame_result;
185  uint32_t h;
186 
187  if (frame) {
188  switch (avctx->sample_fmt) {
189  case AV_SAMPLE_FMT_S16P:
190  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
191  break;
192  case AV_SAMPLE_FMT_S32P:
193  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
194  break;
195  case AV_SAMPLE_FMT_FLTP:
196  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
197  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
198  return AVERROR(EINVAL);
199  }
200  for (ch = 0; ch < avctx->channels; ch++) {
202  (const float *)frame->data[ch],
203  32768.0f,
204  FFALIGN(frame->nb_samples, 8));
205  }
206  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
207  break;
208  default:
209  return AVERROR_BUG;
210  }
211  } else {
212  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
213  s->buffer_size - s->buffer_index);
214  }
215  if (lame_result < 0) {
216  if (lame_result == -1) {
217  av_log(avctx, AV_LOG_ERROR,
218  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
220  }
221  return -1;
222  }
223  s->buffer_index += lame_result;
224  ret = realloc_buffer(s);
225  if (ret < 0) {
226  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
227  return ret;
228  }
229 
230  /* add current frame to the queue */
231  if (frame) {
232  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
233  return ret;
234  }
235 
236  /* Move 1 frame from the LAME buffer to the output packet, if available.
237  We have to parse the first frame header in the output buffer to
238  determine the frame size. */
239  if (s->buffer_index < 4)
240  return 0;
241  h = AV_RB32(s->buffer);
242  if (ff_mpa_check_header(h) < 0) {
243  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
244  return AVERROR_BUG;
245  }
246  if (avpriv_mpegaudio_decode_header(&hdr, h)) {
247  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
248  return -1;
249  }
250  len = hdr.frame_size;
251  av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
252  s->buffer_index);
253  if (len <= s->buffer_index) {
254  if ((ret = ff_alloc_packet(avpkt, len))) {
255  av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
256  return ret;
257  }
258  memcpy(avpkt->data, s->buffer, len);
259  s->buffer_index -= len;
260  memmove(s->buffer, s->buffer + len, s->buffer_index);
261 
262  /* Get the next frame pts/duration */
263  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
264  &avpkt->duration);
265 
266  avpkt->size = len;
267  *got_packet_ptr = 1;
268  }
269  return 0;
270 }
271 
272 #define OFFSET(x) offsetof(LAMEContext, x)
273 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
274 static const AVOption options[] = {
275  { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
276  { "joint_stereo", "Use joint stereo.", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
277  { "abr", "Use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
278  { NULL },
279 };
280 
281 static const AVClass libmp3lame_class = {
282  .class_name = "libmp3lame encoder",
283  .item_name = av_default_item_name,
284  .option = options,
285  .version = LIBAVUTIL_VERSION_INT,
286 };
287 
289  { "b", "0" },
290  { NULL },
291 };
292 
293 static const int libmp3lame_sample_rates[] = {
294  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
295 };
296 
298  .name = "libmp3lame",
299  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
300  .type = AVMEDIA_TYPE_AUDIO,
301  .id = AV_CODEC_ID_MP3,
302  .priv_data_size = sizeof(LAMEContext),
304  .encode2 = mp3lame_encode_frame,
307  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
311  .supported_samplerates = libmp3lame_sample_rates,
312  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
314  0 },
315  .priv_class = &libmp3lame_class,
316  .defaults = libmp3lame_defaults,
317 };
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:178
float, planar
Definition: samplefmt.h:72
static const AVClass libmp3lame_class
Definition: libmp3lame.c:281
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:62
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:1137
This structure describes decoded (raw) audio or video data.
Definition: frame.h:135
AVOption.
Definition: opt.h:234
#define JOINT_STEREO
Definition: atrac3.c:51
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:89
AudioFrameQueue afq
Definition: libmp3lame.c:54
int size
Definition: avcodec.h:974
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:293
AVCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:297
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:2796
#define FFALIGN(x, a)
Definition: common.h:62
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
Definition: libmp3lame.c:75
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:198
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:38
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1799
uint8_t
#define av_cold
Definition: attributes.h:66
AVOptions.
int buffer_size
Definition: libmp3lame.c:49
#define AV_RB32
Definition: intreadwrite.h:130
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
#define BUFFER_SIZE
Definition: libmp3lame.c:41
#define AE
Definition: libmp3lame.c:273
int reservoir
Definition: libmp3lame.c:50
uint8_t * data
Definition: avcodec.h:973
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
int av_reallocp(void *ptr, size_t size)
Allocate or reallocate a block of memory.
Definition: mem.c:140
#define CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:658
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:991
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
uint8_t * buffer
Definition: libmp3lame.c:47
#define CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:713
#define AVERROR(e)
Definition: error.h:43
sample_fmts
Definition: avconv_filter.c:68
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:718
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:150
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:379
int flags
CODEC_FLAG_*.
Definition: avcodec.h:1144
#define CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:611
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:168
const char * name
Name of the codec implementation.
Definition: avcodec.h:2803
static const AVOption options[]
Definition: libmp3lame.c:274
static int ff_mpa_check_header(uint32_t header)
static const AVCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:288
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
#define STEREO
Definition: atrac3.c:52
static int realloc_buffer(LAMEContext *s)
Definition: libmp3lame.c:59
int bit_rate
the average bitrate
Definition: avcodec.h:1114
audio channel layout utility functions
signed 32 bits, planar
Definition: samplefmt.h:71
int32_t
int joint_stereo
Definition: libmp3lame.c:51
int buffer_index
Definition: libmp3lame.c:48
int ff_alloc_packet(AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1245
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:69
AVCodecContext * avctx
Definition: libmp3lame.c:45
AVFloatDSPContext fdsp
Definition: libmp3lame.c:55
LIBAVUTIL_VERSION_INT
Definition: eval.c:55
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1811
NULL
Definition: eval.c:55
Libavcodec external API header.
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:61
int compression_level
Definition: avcodec.h:1136
AV_SAMPLE_FMT_NONE
Definition: avconv_filter.c:68
int sample_rate
samples per second
Definition: avcodec.h:1791
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:153
av_default_item_name
Definition: dnxhdenc.c:52
main external API structure.
Definition: avcodec.h:1050
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:490
#define AVERROR_BUG
Bug detected, please report the issue.
Definition: error.h:60
Describe the class of an AVClass context structure.
Definition: log.h:33
#define MONO
Definition: cook.c:59
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1130
av_cold void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
Initialize a float DSP context.
Definition: float_dsp.c:115
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:141
MPEG Audio header decoder.
common internal api header.
common internal and external API header
mpeg audio declarations for both encoder and decoder.
static av_cold int init(AVCodecParserContext *s)
Definition: h264_parser.c:499
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:170
void * priv_data
Definition: avcodec.h:1092
float * samples_flt[2]
Definition: libmp3lame.c:53
int len
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int *duration)
Remove frame(s) from the queue.
int channels
number of audio channels
Definition: avcodec.h:1792
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:207
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define OFFSET(x)
Definition: libmp3lame.c:272
signed 16 bits, planar
Definition: samplefmt.h:70
lame_global_flags * gfp
Definition: libmp3lame.c:46
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:950
int delay
Codec delay.
Definition: avcodec.h:1212
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:179
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:966