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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  int abr;
53  float *samples_flt[2];
56 } LAMEContext;
57 
58 
60 {
61  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
62  int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
63 
64  av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
65  new_size);
66  if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
67  s->buffer_size = s->buffer_index = 0;
68  return err;
69  }
70  s->buffer_size = new_size;
71  }
72  return 0;
73 }
74 
76 {
77  LAMEContext *s = avctx->priv_data;
78 
79  av_freep(&s->samples_flt[0]);
80  av_freep(&s->samples_flt[1]);
81  av_freep(&s->buffer);
82 
84 
85  lame_close(s->gfp);
86  return 0;
87 }
88 
90 {
91  LAMEContext *s = avctx->priv_data;
92  int ret;
93 
94  s->avctx = avctx;
95 
96  /* initialize LAME and get defaults */
97  if ((s->gfp = lame_init()) == NULL)
98  return AVERROR(ENOMEM);
99 
100 
101  lame_set_num_channels(s->gfp, avctx->channels);
102  lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
103 
104  /* sample rate */
105  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
106  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
107 
108  /* algorithmic quality */
110  lame_set_quality(s->gfp, 5);
111  else
112  lame_set_quality(s->gfp, avctx->compression_level);
113 
114  /* rate control */
115  if (avctx->flags & CODEC_FLAG_QSCALE) { // VBR
116  lame_set_VBR(s->gfp, vbr_default);
117  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
118  } else {
119  if (avctx->bit_rate) {
120  if (s->abr) { // ABR
121  lame_set_VBR(s->gfp, vbr_abr);
122  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
123  } else // CBR
124  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
125  }
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  /* allocate float sample buffers */
147  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
148  int ch;
149  for (ch = 0; ch < avctx->channels; ch++) {
150  s->samples_flt[ch] = av_malloc(avctx->frame_size *
151  sizeof(*s->samples_flt[ch]));
152  if (!s->samples_flt[ch]) {
153  ret = AVERROR(ENOMEM);
154  goto error;
155  }
156  }
157  }
158 
159  ret = realloc_buffer(s);
160  if (ret < 0)
161  goto error;
162 
164 
165  return 0;
166 error:
167  mp3lame_encode_close(avctx);
168  return ret;
169 }
170 
171 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
172  lame_result = func(s->gfp, \
173  (const buf_type *)buf_name[0], \
174  (const buf_type *)buf_name[1], frame->nb_samples, \
175  s->buffer + s->buffer_index, \
176  s->buffer_size - s->buffer_index); \
177 } while (0)
178 
179 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
180  const AVFrame *frame, int *got_packet_ptr)
181 {
182  LAMEContext *s = avctx->priv_data;
183  MPADecodeHeader hdr;
184  int len, ret, ch;
185  int lame_result;
186 
187  if (frame) {
188  switch (avctx->sample_fmt) {
189  case AV_SAMPLE_FMT_S16P:
190  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
191  break;
192  case AV_SAMPLE_FMT_S32P:
193  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
194  break;
195  case AV_SAMPLE_FMT_FLTP:
196  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
197  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
198  return AVERROR(EINVAL);
199  }
200  for (ch = 0; ch < avctx->channels; ch++) {
202  (const float *)frame->data[ch],
203  32768.0f,
204  FFALIGN(frame->nb_samples, 8));
205  }
206  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
207  break;
208  default:
209  return AVERROR_BUG;
210  }
211  } else {
212  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
213  s->buffer_size - s->buffer_index);
214  }
215  if (lame_result < 0) {
216  if (lame_result == -1) {
217  av_log(avctx, AV_LOG_ERROR,
218  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
220  }
221  return -1;
222  }
223  s->buffer_index += lame_result;
224  ret = realloc_buffer(s);
225  if (ret < 0) {
226  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
227  return ret;
228  }
229 
230  /* add current frame to the queue */
231  if (frame) {
232  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
233  return ret;
234  }
235 
236  /* Move 1 frame from the LAME buffer to the output packet, if available.
237  We have to parse the first frame header in the output buffer to
238  determine the frame size. */
239  if (s->buffer_index < 4)
240  return 0;
242  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
243  return -1;
244  }
245  len = hdr.frame_size;
246  av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
247  s->buffer_index);
248  if (len <= s->buffer_index) {
249  if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
250  return ret;
251  memcpy(avpkt->data, s->buffer, len);
252  s->buffer_index -= len;
253  memmove(s->buffer, s->buffer + len, s->buffer_index);
254 
255  /* Get the next frame pts/duration */
256  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
257  &avpkt->duration);
258 
259  avpkt->size = len;
260  *got_packet_ptr = 1;
261  }
262  return 0;
263 }
264 
265 #define OFFSET(x) offsetof(LAMEContext, x)
266 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
267 static const AVOption options[] = {
268  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
269  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
270  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
271  { NULL },
272 };
273 
274 static const AVClass libmp3lame_class = {
275  .class_name = "libmp3lame encoder",
276  .item_name = av_default_item_name,
277  .option = options,
278  .version = LIBAVUTIL_VERSION_INT,
279 };
280 
282  { "b", "0" },
283  { NULL },
284 };
285 
286 static const int libmp3lame_sample_rates[] = {
287  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
288 };
289 
291  .name = "libmp3lame",
292  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
293  .type = AVMEDIA_TYPE_AUDIO,
294  .id = AV_CODEC_ID_MP3,
295  .priv_data_size = sizeof(LAMEContext),
297  .encode2 = mp3lame_encode_frame,
300  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
304  .supported_samplerates = libmp3lame_sample_rates,
305  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
307  0 },
308  .priv_class = &libmp3lame_class,
309  .defaults = libmp3lame_defaults,
310 };