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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()))
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, avctx->compression_level);
111 
112  /* rate control */
113  if (avctx->flags & CODEC_FLAG_QSCALE) { // VBR
114  lame_set_VBR(s->gfp, vbr_default);
115  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
116  } else {
117  if (avctx->bit_rate) {
118  if (s->abr) { // ABR
119  lame_set_VBR(s->gfp, vbr_abr);
120  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
121  } else // CBR
122  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
123  }
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->initial_padding = 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  uint32_t h;
185 
186  if (frame) {
187  switch (avctx->sample_fmt) {
188  case AV_SAMPLE_FMT_S16P:
189  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
190  break;
191  case AV_SAMPLE_FMT_S32P:
192  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
193  break;
194  case AV_SAMPLE_FMT_FLTP:
195  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
196  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
197  return AVERROR(EINVAL);
198  }
199  for (ch = 0; ch < avctx->channels; ch++) {
201  (const float *)frame->data[ch],
202  32768.0f,
203  FFALIGN(frame->nb_samples, 8));
204  }
205  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
206  break;
207  default:
208  return AVERROR_BUG;
209  }
210  } else if (!s->afq.frame_alloc) {
211  lame_result = 0;
212  } else {
213  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
214  s->buffer_size - s->buffer_index);
215  }
216  if (lame_result < 0) {
217  if (lame_result == -1) {
218  av_log(avctx, AV_LOG_ERROR,
219  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
221  }
222  return -1;
223  }
224  s->buffer_index += lame_result;
225  ret = realloc_buffer(s);
226  if (ret < 0) {
227  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
228  return ret;
229  }
230 
231  /* add current frame to the queue */
232  if (frame) {
233  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
234  return ret;
235  }
236 
237  /* Move 1 frame from the LAME buffer to the output packet, if available.
238  We have to parse the first frame header in the output buffer to
239  determine the frame size. */
240  if (s->buffer_index < 4)
241  return 0;
242  h = AV_RB32(s->buffer);
243  if (ff_mpa_check_header(h) < 0) {
244  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
245  return AVERROR_BUG;
246  }
247  if (avpriv_mpegaudio_decode_header(&hdr, h)) {
248  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
249  return -1;
250  }
251  len = hdr.frame_size;
252  av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
253  s->buffer_index);
254  if (len <= s->buffer_index) {
255  if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
256  return ret;
257  memcpy(avpkt->data, s->buffer, len);
258  s->buffer_index -= len;
259  memmove(s->buffer, s->buffer + len, s->buffer_index);
260 
261  /* Get the next frame pts/duration */
262  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
263  &avpkt->duration);
264 
265  avpkt->size = len;
266  *got_packet_ptr = 1;
267  }
268  return 0;
269 }
270 
271 #define OFFSET(x) offsetof(LAMEContext, x)
272 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
273 static const AVOption options[] = {
274  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
275  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
276  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 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  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
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  .priv_class = &libmp3lame_class,
315  .defaults = libmp3lame_defaults,
316 };