FFmpeg  2.1.1
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 
29 #include "libavutil/channel_layout.h"
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  float *samples_flt[2];
55 } LAMEContext;
56 
57 
59 {
60  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
61  uint8_t *tmp;
62  int new_size = s->buffer_index + 2 * BUFFER_SIZE;
63 
64  av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
65  new_size);
66  tmp = av_realloc(s->buffer, new_size);
67  if (!tmp) {
68  av_freep(&s->buffer);
69  s->buffer_size = s->buffer_index = 0;
70  return AVERROR(ENOMEM);
71  }
72  s->buffer = tmp;
73  s->buffer_size = new_size;
74  }
75  return 0;
76 }
77 
79 {
80  LAMEContext *s = avctx->priv_data;
81 
82  av_freep(&s->samples_flt[0]);
83  av_freep(&s->samples_flt[1]);
84  av_freep(&s->buffer);
85 
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()) == NULL)
101  return AVERROR(ENOMEM);
102 
103 
104  lame_set_num_channels(s->gfp, avctx->channels);
105  lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
106 
107  /* sample rate */
108  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
109  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
110 
111  /* algorithmic quality */
113  lame_set_quality(s->gfp, 5);
114  else
115  lame_set_quality(s->gfp, avctx->compression_level);
116 
117  /* rate control */
118  if (avctx->flags & CODEC_FLAG_QSCALE) { // VBR
119  lame_set_VBR(s->gfp, vbr_default);
120  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
121  } else {
122  if (avctx->bit_rate) // CBR
123  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
124  }
125 
126  /* do not get a Xing VBR header frame from LAME */
127  lame_set_bWriteVbrTag(s->gfp,0);
128 
129  /* bit reservoir usage */
130  lame_set_disable_reservoir(s->gfp, !s->reservoir);
131 
132  /* set specified parameters */
133  if (lame_init_params(s->gfp) < 0) {
134  ret = -1;
135  goto error;
136  }
137 
138  /* get encoder delay */
139  avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
140  ff_af_queue_init(avctx, &s->afq);
141 
142  avctx->frame_size = lame_get_framesize(s->gfp);
143 
144  /* allocate float sample buffers */
145  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
146  int ch;
147  for (ch = 0; ch < avctx->channels; ch++) {
148  s->samples_flt[ch] = av_malloc(avctx->frame_size *
149  sizeof(*s->samples_flt[ch]));
150  if (!s->samples_flt[ch]) {
151  ret = AVERROR(ENOMEM);
152  goto error;
153  }
154  }
155  }
156 
157  ret = realloc_buffer(s);
158  if (ret < 0)
159  goto error;
160 
162 
163  return 0;
164 error:
165  mp3lame_encode_close(avctx);
166  return ret;
167 }
168 
169 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
170  lame_result = func(s->gfp, \
171  (const buf_type *)buf_name[0], \
172  (const buf_type *)buf_name[1], frame->nb_samples, \
173  s->buffer + s->buffer_index, \
174  s->buffer_size - s->buffer_index); \
175 } while (0)
176 
177 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
178  const AVFrame *frame, int *got_packet_ptr)
179 {
180  LAMEContext *s = avctx->priv_data;
181  MPADecodeHeader hdr;
182  int len, ret, ch;
183  int lame_result;
184 
185  if (frame) {
186  switch (avctx->sample_fmt) {
187  case AV_SAMPLE_FMT_S16P:
188  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
189  break;
190  case AV_SAMPLE_FMT_S32P:
191  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
192  break;
193  case AV_SAMPLE_FMT_FLTP:
194  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
195  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
196  return AVERROR(EINVAL);
197  }
198  for (ch = 0; ch < avctx->channels; ch++) {
200  (const float *)frame->data[ch],
201  32768.0f,
202  FFALIGN(frame->nb_samples, 8));
203  }
204  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
205  break;
206  default:
207  return AVERROR_BUG;
208  }
209  } else {
210  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
211  s->buffer_size - s->buffer_index);
212  }
213  if (lame_result < 0) {
214  if (lame_result == -1) {
215  av_log(avctx, AV_LOG_ERROR,
216  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
218  }
219  return -1;
220  }
221  s->buffer_index += lame_result;
222  ret = realloc_buffer(s);
223  if (ret < 0) {
224  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
225  return ret;
226  }
227 
228  /* add current frame to the queue */
229  if (frame) {
230  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
231  return ret;
232  }
233 
234  /* Move 1 frame from the LAME buffer to the output packet, if available.
235  We have to parse the first frame header in the output buffer to
236  determine the frame size. */
237  if (s->buffer_index < 4)
238  return 0;
240  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
241  return -1;
242  }
243  len = hdr.frame_size;
244  av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
245  s->buffer_index);
246  if (len <= s->buffer_index) {
247  if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
248  return ret;
249  memcpy(avpkt->data, s->buffer, len);
250  s->buffer_index -= len;
251  memmove(s->buffer, s->buffer + len, s->buffer_index);
252 
253  /* Get the next frame pts/duration */
254  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
255  &avpkt->duration);
256 
257  avpkt->size = len;
258  *got_packet_ptr = 1;
259  }
260  return 0;
261 }
262 
263 #define OFFSET(x) offsetof(LAMEContext, x)
264 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
265 static const AVOption options[] = {
266  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
267  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
268  { NULL },
269 };
270 
271 static const AVClass libmp3lame_class = {
272  .class_name = "libmp3lame encoder",
273  .item_name = av_default_item_name,
274  .option = options,
275  .version = LIBAVUTIL_VERSION_INT,
276 };
277 
279  { "b", "0" },
280  { NULL },
281 };
282 
283 static const int libmp3lame_sample_rates[] = {
284  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
285 };
286 
288  .name = "libmp3lame",
289  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
290  .type = AVMEDIA_TYPE_AUDIO,
291  .id = AV_CODEC_ID_MP3,
292  .priv_data_size = sizeof(LAMEContext),
294  .encode2 = mp3lame_encode_frame,
297  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
301  .supported_samplerates = libmp3lame_sample_rates,
302  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
304  0 },
305  .priv_class = &libmp3lame_class,
306  .defaults = libmp3lame_defaults,
307 };
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:177
static const AVClass libmp3lame_class
Definition: libmp3lame.c:271
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1500
const char * s
Definition: avisynth_c.h:668
This structure describes decoded (raw) audio or video data.
Definition: frame.h:96
AVOption.
Definition: opt.h:253
#define JOINT_STEREO
Definition: atrac3.c:52
#define LIBAVUTIL_VERSION_INT
Definition: avcodec.h:820
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:92
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AudioFrameQueue afq
Definition: libmp3lame.c:53
int size
Definition: avcodec.h:1064
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:283
void av_log(void *avcl, int level, const char *fmt,...) av_printf_format(3
Send the specified message to the log if the level is less than or equal to the current av_log_level...
AVCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:287
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:2922
#define av_cold
Definition: avcodec.h:653
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
Definition: libmp3lame.c:78
void av_freep(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:234
const char * av_default_item_name(void *ctx)
Return the context name.
Definition: log.c:145
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1881
uint8_t
signed 32 bits, planar
Definition: samplefmt.h:59
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:55
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
const char * name
Name of the codec implementation.
Definition: avcodec.h:2929
#define AE
Definition: libmp3lame.c:264
int reservoir
Definition: libmp3lame.c:50
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
#define CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:714
int duration
Duration of this packet in AVStream-&gt;time_base units, 0 if unknown.
Definition: avcodec.h:1085
const OptionDef options[]
Definition: ffserver.c:4682
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:1227
static AVFrame * frame
Definition: demuxing.c:51
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: avcodec.h:4147
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:769
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:774
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:151
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:398
int flags
CODEC_FLAG_*.
Definition: avcodec.h:1234
#define CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:692
static const AVCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:278
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:53
static int realloc_buffer(LAMEContext *s)
Definition: libmp3lame.c:58
int bit_rate
the average bitrate
Definition: avcodec.h:1204
void * av_realloc(void *ptr, size_t size) 1(2)
Allocate or reallocate a block of memory.
Definition: mem.c:141
ret
Definition: avfilter.c:961
void * av_malloc(size_t size) av_malloc_attrib 1(1)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:73
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
AVFloatDSPContext fdsp
Definition: libmp3lame.c:54
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1893
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
int compression_level
Definition: avcodec.h:1226
int sample_rate
samples per second
Definition: avcodec.h:1873
main external API structure.
Definition: avcodec.h:1146
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:538
Describe the class of an AVClass context structure.
Definition: log.h:50
#define MONO
Definition: cook.c:58
uint8_t * data
Definition: avcodec.h:1063
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:124
void * priv_data
Definition: avcodec.h:1182
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1220
av_cold void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
Initialize a float DSP context.
Definition: float_dsp.c:118
MPEG Audio header decoder.
common internal api header.
#define FFALIGN(x, a)
Definition: avcodec.h:930
mpeg audio declarations for both encoder and decoder.
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:169
#define AV_RB32(x)
Definition: intreadwrite.h:258
#define AVERROR_BUG
float * samples_flt[2]
Definition: libmp3lame.c:52
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:1874
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avcodec.h:2257
#define AVERROR(e)
float, planar
Definition: samplefmt.h:60
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define OFFSET(x)
Definition: libmp3lame.c:263
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:700
signed 16 bits, planar
Definition: samplefmt.h:58
lame_global_flags * gfp
Definition: libmp3lame.c:46
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1040
int delay
Codec delay.
Definition: avcodec.h:1302
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:150
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:107
int64_t pts
Presentation timestamp in AVStream-&gt;time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1056