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;
51  float *samples_flt[2];
54 } LAMEContext;
55 
56 
58 {
59  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
60  int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
61 
62  av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
63  new_size);
64  if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
65  s->buffer_size = s->buffer_index = 0;
66  return err;
67  }
68  s->buffer_size = new_size;
69  }
70  return 0;
71 }
72 
74 {
75  LAMEContext *s = avctx->priv_data;
76 
77  av_freep(&s->samples_flt[0]);
78  av_freep(&s->samples_flt[1]);
79  av_freep(&s->buffer);
80 
82 
83  lame_close(s->gfp);
84  return 0;
85 }
86 
88 {
89  LAMEContext *s = avctx->priv_data;
90  int ret;
91 
92  s->avctx = avctx;
93 
94  /* initialize LAME and get defaults */
95  if ((s->gfp = lame_init()) == NULL)
96  return AVERROR(ENOMEM);
97 
98  lame_set_num_channels(s->gfp, avctx->channels);
99  lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
100 
101  /* sample rate */
102  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
103  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
104 
105  /* algorithmic quality */
107  lame_set_quality(s->gfp, 5);
108  else
109  lame_set_quality(s->gfp, avctx->compression_level);
110 
111  /* rate control */
112  if (avctx->flags & CODEC_FLAG_QSCALE) {
113  lame_set_VBR(s->gfp, vbr_default);
114  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
115  } else {
116  if (avctx->bit_rate)
117  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
118  }
119 
120  /* do not get a Xing VBR header frame from LAME */
121  lame_set_bWriteVbrTag(s->gfp,0);
122 
123  /* bit reservoir usage */
124  lame_set_disable_reservoir(s->gfp, !s->reservoir);
125 
126  /* set specified parameters */
127  if (lame_init_params(s->gfp) < 0) {
128  ret = -1;
129  goto error;
130  }
131 
132  /* get encoder delay */
133  avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
134  ff_af_queue_init(avctx, &s->afq);
135 
136  avctx->frame_size = lame_get_framesize(s->gfp);
137 
138  /* allocate float sample buffers */
139  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
140  int ch;
141  for (ch = 0; ch < avctx->channels; ch++) {
142  s->samples_flt[ch] = av_malloc(avctx->frame_size *
143  sizeof(*s->samples_flt[ch]));
144  if (!s->samples_flt[ch]) {
145  ret = AVERROR(ENOMEM);
146  goto error;
147  }
148  }
149  }
150 
151  ret = realloc_buffer(s);
152  if (ret < 0)
153  goto error;
154 
156 
157  return 0;
158 error:
159  mp3lame_encode_close(avctx);
160  return ret;
161 }
162 
163 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
164  lame_result = func(s->gfp, \
165  (const buf_type *)buf_name[0], \
166  (const buf_type *)buf_name[1], frame->nb_samples, \
167  s->buffer + s->buffer_index, \
168  s->buffer_size - s->buffer_index); \
169 } while (0)
170 
171 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
172  const AVFrame *frame, int *got_packet_ptr)
173 {
174  LAMEContext *s = avctx->priv_data;
175  MPADecodeHeader hdr;
176  int len, ret, ch;
177  int lame_result;
178 
179  if (frame) {
180  switch (avctx->sample_fmt) {
181  case AV_SAMPLE_FMT_S16P:
182  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
183  break;
184  case AV_SAMPLE_FMT_S32P:
185  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
186  break;
187  case AV_SAMPLE_FMT_FLTP:
188  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
189  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
190  return AVERROR(EINVAL);
191  }
192  for (ch = 0; ch < avctx->channels; ch++) {
194  (const float *)frame->data[ch],
195  32768.0f,
196  FFALIGN(frame->nb_samples, 8));
197  }
198  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
199  break;
200  default:
201  return AVERROR_BUG;
202  }
203  } else {
204  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
205  s->buffer_size - s->buffer_index);
206  }
207  if (lame_result < 0) {
208  if (lame_result == -1) {
209  av_log(avctx, AV_LOG_ERROR,
210  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
212  }
213  return -1;
214  }
215  s->buffer_index += lame_result;
216  ret = realloc_buffer(s);
217  if (ret < 0) {
218  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
219  return ret;
220  }
221 
222  /* add current frame to the queue */
223  if (frame) {
224  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
225  return ret;
226  }
227 
228  /* Move 1 frame from the LAME buffer to the output packet, if available.
229  We have to parse the first frame header in the output buffer to
230  determine the frame size. */
231  if (s->buffer_index < 4)
232  return 0;
234  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
235  return -1;
236  }
237  len = hdr.frame_size;
238  av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
239  s->buffer_index);
240  if (len <= s->buffer_index) {
241  if ((ret = ff_alloc_packet(avpkt, len))) {
242  av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
243  return ret;
244  }
245  memcpy(avpkt->data, s->buffer, len);
246  s->buffer_index -= len;
247  memmove(s->buffer, s->buffer + len, s->buffer_index);
248 
249  /* Get the next frame pts/duration */
250  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
251  &avpkt->duration);
252 
253  avpkt->size = len;
254  *got_packet_ptr = 1;
255  }
256  return 0;
257 }
258 
259 #define OFFSET(x) offsetof(LAMEContext, x)
260 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
261 static const AVOption options[] = {
262  { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
263  { NULL },
264 };
265 
266 static const AVClass libmp3lame_class = {
267  .class_name = "libmp3lame encoder",
268  .item_name = av_default_item_name,
269  .option = options,
270  .version = LIBAVUTIL_VERSION_INT,
271 };
272 
274  { "b", "0" },
275  { NULL },
276 };
277 
278 static const int libmp3lame_sample_rates[] = {
279  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
280 };
281 
283  .name = "libmp3lame",
284  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
285  .type = AVMEDIA_TYPE_AUDIO,
286  .id = AV_CODEC_ID_MP3,
287  .priv_data_size = sizeof(LAMEContext),
289  .encode2 = mp3lame_encode_frame,
292  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
296  .supported_samplerates = libmp3lame_sample_rates,
297  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
299  0 },
300  .priv_class = &libmp3lame_class,
301  .defaults = libmp3lame_defaults,
302 };