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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  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 };