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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;
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 
106  lame_set_num_channels(s->gfp, avctx->channels);
107  lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
108 
109  /* sample rate */
110  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
111  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
112 
113  /* algorithmic quality */
115  lame_set_quality(s->gfp, 5);
116  else
117  lame_set_quality(s->gfp, avctx->compression_level);
118 
119  /* rate control */
120  if (avctx->flags & CODEC_FLAG_QSCALE) {
121  lame_set_VBR(s->gfp, vbr_default);
122  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
123  } else {
124  if (avctx->bit_rate)
125  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
126  }
127 
128  /* do not get a Xing VBR header frame from LAME */
129  lame_set_bWriteVbrTag(s->gfp,0);
130 
131  /* bit reservoir usage */
132  lame_set_disable_reservoir(s->gfp, !s->reservoir);
133 
134  /* set specified parameters */
135  if (lame_init_params(s->gfp) < 0) {
136  ret = -1;
137  goto error;
138  }
139 
140  /* get encoder delay */
141  avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
142  ff_af_queue_init(avctx, &s->afq);
143 
144  avctx->frame_size = lame_get_framesize(s->gfp);
145 
146 #if FF_API_OLD_ENCODE_AUDIO
147  avctx->coded_frame = avcodec_alloc_frame();
148  if (!avctx->coded_frame) {
149  ret = AVERROR(ENOMEM);
150  goto error;
151  }
152 #endif
153 
154  /* allocate float sample buffers */
155  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
156  int ch;
157  for (ch = 0; ch < avctx->channels; ch++) {
158  s->samples_flt[ch] = av_malloc(avctx->frame_size *
159  sizeof(*s->samples_flt[ch]));
160  if (!s->samples_flt[ch]) {
161  ret = AVERROR(ENOMEM);
162  goto error;
163  }
164  }
165  }
166 
167  ret = realloc_buffer(s);
168  if (ret < 0)
169  goto error;
170 
172 
173  return 0;
174 error:
175  mp3lame_encode_close(avctx);
176  return ret;
177 }
178 
179 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
180  lame_result = func(s->gfp, \
181  (const buf_type *)buf_name[0], \
182  (const buf_type *)buf_name[1], frame->nb_samples, \
183  s->buffer + s->buffer_index, \
184  s->buffer_size - s->buffer_index); \
185 } while (0)
186 
187 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
188  const AVFrame *frame, int *got_packet_ptr)
189 {
190  LAMEContext *s = avctx->priv_data;
191  MPADecodeHeader hdr;
192  int len, ret, ch;
193  int lame_result;
194 
195  if (frame) {
196  switch (avctx->sample_fmt) {
197  case AV_SAMPLE_FMT_S16P:
198  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
199  break;
200  case AV_SAMPLE_FMT_S32P:
201  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
202  break;
203  case AV_SAMPLE_FMT_FLTP:
204  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
205  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
206  return AVERROR(EINVAL);
207  }
208  for (ch = 0; ch < avctx->channels; ch++) {
210  (const float *)frame->data[ch],
211  32768.0f,
212  FFALIGN(frame->nb_samples, 8));
213  }
214  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
215  break;
216  default:
217  return AVERROR_BUG;
218  }
219  } else {
220  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
222  }
223  if (lame_result < 0) {
224  if (lame_result == -1) {
225  av_log(avctx, AV_LOG_ERROR,
226  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
228  }
229  return -1;
230  }
231  s->buffer_index += lame_result;
232  ret = realloc_buffer(s);
233  if (ret < 0) {
234  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
235  return ret;
236  }
237 
238  /* add current frame to the queue */
239  if (frame) {
240  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
241  return ret;
242  }
243 
244  /* Move 1 frame from the LAME buffer to the output packet, if available.
245  We have to parse the first frame header in the output buffer to
246  determine the frame size. */
247  if (s->buffer_index < 4)
248  return 0;
250  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
251  return -1;
252  }
253  len = hdr.frame_size;
254  av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
255  s->buffer_index);
256  if (len <= s->buffer_index) {
257  if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
258  return ret;
259  memcpy(avpkt->data, s->buffer, len);
260  s->buffer_index -= len;
261  memmove(s->buffer, s->buffer + len, s->buffer_index);
262 
263  /* Get the next frame pts/duration */
264  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
265  &avpkt->duration);
266 
267  avpkt->size = len;
268  *got_packet_ptr = 1;
269  }
270  return 0;
271 }
272 
273 #define OFFSET(x) offsetof(LAMEContext, x)
274 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
275 static const AVOption options[] = {
276  { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
277  { NULL },
278 };
279 
280 static const AVClass libmp3lame_class = {
281  .class_name = "libmp3lame encoder",
282  .item_name = av_default_item_name,
283  .option = options,
284  .version = LIBAVUTIL_VERSION_INT,
285 };
286 
288  { "b", "0" },
289  { NULL },
290 };
291 
292 static const int libmp3lame_sample_rates[] = {
293  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
294 };
295 
297  .name = "libmp3lame",
298  .type = AVMEDIA_TYPE_AUDIO,
299  .id = AV_CODEC_ID_MP3,
300  .priv_data_size = sizeof(LAMEContext),
302  .encode2 = mp3lame_encode_frame,
305  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
309  .supported_samplerates = libmp3lame_sample_rates,
310  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
312  0 },
313  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
314  .priv_class = &libmp3lame_class,
315  .defaults = libmp3lame_defaults,
316 };