libmp3lame.c
Go to the documentation of this file.
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;
51  float *samples_flt[2];
54 } LAMEContext;
55 
56 
58 {
59  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
60  uint8_t *tmp;
61  int new_size = s->buffer_index + 2 * BUFFER_SIZE;
62 
63  av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
64  new_size);
65  tmp = av_realloc(s->buffer, new_size);
66  if (!tmp) {
67  av_freep(&s->buffer);
68  s->buffer_size = s->buffer_index = 0;
69  return AVERROR(ENOMEM);
70  }
71  s->buffer = tmp;
72  s->buffer_size = new_size;
73  }
74  return 0;
75 }
76 
78 {
79  LAMEContext *s = avctx->priv_data;
80 
81 #if FF_API_OLD_ENCODE_AUDIO
82  av_freep(&avctx->coded_frame);
83 #endif
84  av_freep(&s->samples_flt[0]);
85  av_freep(&s->samples_flt[1]);
86  av_freep(&s->buffer);
87 
89 
90  lame_close(s->gfp);
91  return 0;
92 }
93 
95 {
96  LAMEContext *s = avctx->priv_data;
97  int ret;
98 
99  s->avctx = avctx;
100 
101  /* initialize LAME and get defaults */
102  if ((s->gfp = lame_init()) == NULL)
103  return AVERROR(ENOMEM);
104 
105  lame_set_num_channels(s->gfp, avctx->channels);
106  lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_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, 5);
115  else
116  lame_set_quality(s->gfp, avctx->compression_level);
117 
118  /* rate control */
119  if (avctx->flags & CODEC_FLAG_QSCALE) {
120  lame_set_VBR(s->gfp, vbr_default);
121  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
122  } else {
123  if (avctx->bit_rate)
124  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
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 #if FF_API_OLD_ENCODE_AUDIO
146  avctx->coded_frame = avcodec_alloc_frame();
147  if (!avctx->coded_frame) {
148  ret = AVERROR(ENOMEM);
149  goto error;
150  }
151 #endif
152 
153  /* allocate float sample buffers */
154  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
155  int ch;
156  for (ch = 0; ch < avctx->channels; ch++) {
157  s->samples_flt[ch] = av_malloc(avctx->frame_size *
158  sizeof(*s->samples_flt[ch]));
159  if (!s->samples_flt[ch]) {
160  ret = AVERROR(ENOMEM);
161  goto error;
162  }
163  }
164  }
165 
166  ret = realloc_buffer(s);
167  if (ret < 0)
168  goto error;
169 
171 
172  return 0;
173 error:
174  mp3lame_encode_close(avctx);
175  return ret;
176 }
177 
178 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
179  lame_result = func(s->gfp, \
180  (const buf_type *)buf_name[0], \
181  (const buf_type *)buf_name[1], frame->nb_samples, \
182  s->buffer + s->buffer_index, \
183  s->buffer_size - s->buffer_index); \
184 } while (0)
185 
186 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
187  const AVFrame *frame, int *got_packet_ptr)
188 {
189  LAMEContext *s = avctx->priv_data;
190  MPADecodeHeader hdr;
191  int len, ret, ch;
192  int lame_result;
193 
194  if (frame) {
195  switch (avctx->sample_fmt) {
196  case AV_SAMPLE_FMT_S16P:
197  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
198  break;
199  case AV_SAMPLE_FMT_S32P:
200  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
201  break;
202  case AV_SAMPLE_FMT_FLTP:
203  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
204  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
205  return AVERROR(EINVAL);
206  }
207  for (ch = 0; ch < avctx->channels; ch++) {
209  (const float *)frame->data[ch],
210  32768.0f,
211  FFALIGN(frame->nb_samples, 8));
212  }
213  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
214  break;
215  default:
216  return AVERROR_BUG;
217  }
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;
249  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
250  return -1;
251  }
252  len = hdr.frame_size;
253  av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
254  s->buffer_index);
255  if (len <= s->buffer_index) {
256  if ((ret = ff_alloc_packet(avpkt, len))) {
257  av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
258  return ret;
259  }
260  memcpy(avpkt->data, s->buffer, len);
261  s->buffer_index -= len;
262  memmove(s->buffer, s->buffer + len, s->buffer_index);
263 
264  /* Get the next frame pts/duration */
265  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
266  &avpkt->duration);
267 
268  avpkt->size = len;
269  *got_packet_ptr = 1;
270  }
271  return 0;
272 }
273 
274 #define OFFSET(x) offsetof(LAMEContext, x)
275 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
276 static const AVOption options[] = {
277  { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 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  .type = AVMEDIA_TYPE_AUDIO,
300  .id = AV_CODEC_ID_MP3,
301  .priv_data_size = sizeof(LAMEContext),
303  .encode2 = mp3lame_encode_frame,
306  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
310  .supported_samplerates = libmp3lame_sample_rates,
311  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
313  0 },
314  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
315  .priv_class = &libmp3lame_class,
316  .defaults = libmp3lame_defaults,
317 };