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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;
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  ff_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  av_freep(&s->fdsp);
83 
85 
86  lame_close(s->gfp);
87  return 0;
88 }
89 
91 {
92  LAMEContext *s = avctx->priv_data;
93  int ret;
94 
95  s->avctx = avctx;
96 
97  /* initialize LAME and get defaults */
98  if (!(s->gfp = lame_init()))
99  return AVERROR(ENOMEM);
100 
101 
102  lame_set_num_channels(s->gfp, avctx->channels);
103  lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
104 
105  /* sample rate */
106  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
107  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
108 
109  /* algorithmic quality */
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->initial_padding = 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_array(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  if (!s->fdsp) {
164  ret = AVERROR(ENOMEM);
165  goto error;
166  }
167 
168 
169  return 0;
170 error:
171  mp3lame_encode_close(avctx);
172  return ret;
173 }
174 
175 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
176  lame_result = func(s->gfp, \
177  (const buf_type *)buf_name[0], \
178  (const buf_type *)buf_name[1], frame->nb_samples, \
179  s->buffer + s->buffer_index, \
180  s->buffer_size - s->buffer_index); \
181 } while (0)
182 
183 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
184  const AVFrame *frame, int *got_packet_ptr)
185 {
186  LAMEContext *s = avctx->priv_data;
187  MPADecodeHeader hdr;
188  int len, ret, ch;
189  int lame_result;
190  uint32_t h;
191 
192  if (frame) {
193  switch (avctx->sample_fmt) {
194  case AV_SAMPLE_FMT_S16P:
195  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
196  break;
197  case AV_SAMPLE_FMT_S32P:
198  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
199  break;
200  case AV_SAMPLE_FMT_FLTP:
201  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
202  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
203  return AVERROR(EINVAL);
204  }
205  for (ch = 0; ch < avctx->channels; ch++) {
207  (const float *)frame->data[ch],
208  32768.0f,
209  FFALIGN(frame->nb_samples, 8));
210  }
211  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
212  break;
213  default:
214  return AVERROR_BUG;
215  }
216  } else if (!s->afq.frame_alloc) {
217  lame_result = 0;
218  } else {
219  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
220  s->buffer_size - s->buffer_index);
221  }
222  if (lame_result < 0) {
223  if (lame_result == -1) {
224  av_log(avctx, AV_LOG_ERROR,
225  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
227  }
228  return -1;
229  }
230  s->buffer_index += lame_result;
231  ret = realloc_buffer(s);
232  if (ret < 0) {
233  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
234  return ret;
235  }
236 
237  /* add current frame to the queue */
238  if (frame) {
239  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
240  return ret;
241  }
242 
243  /* Move 1 frame from the LAME buffer to the output packet, if available.
244  We have to parse the first frame header in the output buffer to
245  determine the frame size. */
246  if (s->buffer_index < 4)
247  return 0;
248  h = AV_RB32(s->buffer);
249  if (ff_mpa_check_header(h) < 0) {
250  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
251  return AVERROR_BUG;
252  }
253  if (avpriv_mpegaudio_decode_header(&hdr, h)) {
254  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
255  return -1;
256  }
257  len = hdr.frame_size;
258  ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
259  s->buffer_index);
260  if (len <= s->buffer_index) {
261  if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
262  return ret;
263  memcpy(avpkt->data, s->buffer, len);
264  s->buffer_index -= len;
265  memmove(s->buffer, s->buffer + len, s->buffer_index);
266 
267  /* Get the next frame pts/duration */
268  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
269  &avpkt->duration);
270 
271  avpkt->size = len;
272  *got_packet_ptr = 1;
273  }
274  return 0;
275 }
276 
277 #define OFFSET(x) offsetof(LAMEContext, x)
278 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
279 static const AVOption options[] = {
280  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
281  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
282  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
283  { NULL },
284 };
285 
286 static const AVClass libmp3lame_class = {
287  .class_name = "libmp3lame encoder",
288  .item_name = av_default_item_name,
289  .option = options,
290  .version = LIBAVUTIL_VERSION_INT,
291 };
292 
294  { "b", "0" },
295  { NULL },
296 };
297 
298 static const int libmp3lame_sample_rates[] = {
299  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
300 };
301 
303  .name = "libmp3lame",
304  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
305  .type = AVMEDIA_TYPE_AUDIO,
306  .id = AV_CODEC_ID_MP3,
307  .priv_data_size = sizeof(LAMEContext),
309  .encode2 = mp3lame_encode_frame,
310  .close = mp3lame_encode_close,
312  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
316  .supported_samplerates = libmp3lame_sample_rates,
317  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
319  0 },
320  .priv_class = &libmp3lame_class,
321  .defaults = libmp3lame_defaults,
322 };
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:183
float, planar
Definition: samplefmt.h:70
static const AVClass libmp3lame_class
Definition: libmp3lame.c:286
#define NULL
Definition: coverity.c:32
AVFloatDSPContext * fdsp
Definition: libmp3lame.c:55
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1743
const char * s
Definition: avisynth_c.h:631
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:1339
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
AVOption.
Definition: opt.h:255
#define JOINT_STEREO
Definition: atrac3.c:51
#define LIBAVUTIL_VERSION_INT
Definition: version.h:62
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:90
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AudioFrameQueue afq
Definition: libmp3lame.c:54
int size
Definition: avcodec.h:1174
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:298
AVCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:302
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:3208
#define FFALIGN(x, a)
Definition: common.h:71
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
Definition: libmp3lame.c:75
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
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2020
uint8_t
#define av_cold
Definition: attributes.h:74
AVOptions.
int buffer_size
Definition: libmp3lame.c:49
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
#define BUFFER_SIZE
Definition: libmp3lame.c:41
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:85
#define AE
Definition: libmp3lame.c:278
static AVFrame * frame
int reservoir
Definition: libmp3lame.c:50
uint8_t * data
Definition: avcodec.h:1173
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:185
#define CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:764
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1191
#define av_log(a,...)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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:829
av_default_item_name
#define AVERROR(e)
Definition: error.h:43
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:834
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:175
int initial_padding
Audio only.
Definition: avcodec.h:3042
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:425
int flags
CODEC_FLAG_*.
Definition: avcodec.h:1346
#define CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:717
const char * name
Name of the codec implementation.
Definition: avcodec.h:3215
static const AVOption options[]
Definition: libmp3lame.c:279
static int ff_mpa_check_header(uint32_t header)
static const AVCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:293
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
Libavcodec external API header.
#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:1316
audio channel layout utility functions
signed 32 bits, planar
Definition: samplefmt.h:69
ret
Definition: avfilter.c:974
int32_t
int joint_stereo
Definition: libmp3lame.c:51
int buffer_index
Definition: libmp3lame.c:48
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
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2032
#define ff_dlog(ctx,...)
Definition: internal.h:54
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
int compression_level
Definition: avcodec.h:1338
int sample_rate
samples per second
Definition: avcodec.h:2012
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:199
main external API structure.
Definition: avcodec.h:1252
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
Describe the class of an AVClass context structure.
Definition: log.h:67
#define MONO
Definition: cook.c:60
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:143
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1332
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:182
MPEG Audio header decoder.
common internal api header.
common internal and external API header
mpeg audio declarations for both encoder and decoder.
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:175
void * priv_data
Definition: avcodec.h:1294
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:2013
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:227
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define OFFSET(x)
Definition: libmp3lame.c:277
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:68
#define av_malloc_array(a, b)
lame_global_flags * gfp
Definition: libmp3lame.c:46
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1150
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:225
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1166