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libvorbisenc.c
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1 /*
2  * Copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include <vorbis/vorbisenc.h>
22 
23 #include "libavutil/avassert.h"
24 #include "libavutil/fifo.h"
25 #include "libavutil/opt.h"
26 #include "avcodec.h"
27 #include "audio_frame_queue.h"
28 #include "internal.h"
29 #include "vorbis.h"
30 #include "vorbis_parser.h"
31 
32 
33 /* Number of samples the user should send in each call.
34  * This value is used because it is the LCD of all possible frame sizes, so
35  * an output packet will always start at the same point as one of the input
36  * packets.
37  */
38 #define LIBVORBIS_FRAME_SIZE 64
39 
40 #define BUFFER_SIZE (1024 * 64)
41 
42 typedef struct LibvorbisEncContext {
43  AVClass *av_class; /**< class for AVOptions */
44  vorbis_info vi; /**< vorbis_info used during init */
45  vorbis_dsp_state vd; /**< DSP state used for analysis */
46  vorbis_block vb; /**< vorbis_block used for analysis */
47  AVFifoBuffer *pkt_fifo; /**< output packet buffer */
48  int eof; /**< end-of-file flag */
49  int dsp_initialized; /**< vd has been initialized */
50  vorbis_comment vc; /**< VorbisComment info */
51  double iblock; /**< impulse block bias option */
52  VorbisParseContext vp; /**< parse context to get durations */
53  AudioFrameQueue afq; /**< frame queue for timestamps */
55 
56 static const AVOption options[] = {
57  { "iblock", "Sets the impulse block bias", offsetof(LibvorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
58  { NULL }
59 };
60 
61 static const AVCodecDefault defaults[] = {
62  { "b", "0" },
63  { NULL },
64 };
65 
66 static const AVClass vorbis_class = {
67  .class_name = "libvorbis",
68  .item_name = av_default_item_name,
69  .option = options,
70  .version = LIBAVUTIL_VERSION_INT,
71 };
72 
73 static int vorbis_error_to_averror(int ov_err)
74 {
75  switch (ov_err) {
76  case OV_EFAULT: return AVERROR_BUG;
77  case OV_EINVAL: return AVERROR(EINVAL);
78  case OV_EIMPL: return AVERROR(EINVAL);
79  default: return AVERROR_UNKNOWN;
80  }
81 }
82 
83 static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
84 {
86  double cfreq;
87  int ret;
88 
89  if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) {
90  /* variable bitrate
91  * NOTE: we use the oggenc range of -1 to 10 for global_quality for
92  * user convenience, but libvorbis uses -0.1 to 1.0.
93  */
94  float q = avctx->global_quality / (float)FF_QP2LAMBDA;
95  /* default to 3 if the user did not set quality or bitrate */
96  if (!(avctx->flags & CODEC_FLAG_QSCALE))
97  q = 3.0;
98  if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
99  avctx->sample_rate,
100  q / 10.0)))
101  goto error;
102  } else {
103  int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
104  int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
105 
106  /* average bitrate */
107  if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
108  avctx->sample_rate, maxrate,
109  avctx->bit_rate, minrate)))
110  goto error;
111 
112  /* variable bitrate by estimate, disable slow rate management */
113  if (minrate == -1 && maxrate == -1)
114  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
115  goto error; /* should not happen */
116  }
117 
118  /* cutoff frequency */
119  if (avctx->cutoff > 0) {
120  cfreq = avctx->cutoff / 1000.0;
121  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
122  goto error; /* should not happen */
123  }
124 
125  /* impulse block bias */
126  if (s->iblock) {
127  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
128  goto error;
129  }
130 
131  if (avctx->channels == 3 &&
133  avctx->channels == 4 &&
134  avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
135  avctx->channel_layout != AV_CH_LAYOUT_QUAD ||
136  avctx->channels == 5 &&
139  avctx->channels == 6 &&
142  avctx->channels == 7 &&
144  avctx->channels == 8 &&
146  if (avctx->channel_layout) {
147  char name[32];
148  av_get_channel_layout_string(name, sizeof(name), avctx->channels,
149  avctx->channel_layout);
150  av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
151  "output stream will have incorrect "
152  "channel layout.\n", name);
153  } else {
154  av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
155  "will use Vorbis channel layout for "
156  "%d channels.\n", avctx->channels);
157  }
158  }
159 
160  if ((ret = vorbis_encode_setup_init(vi)))
161  goto error;
162 
163  return 0;
164 error:
165  return vorbis_error_to_averror(ret);
166 }
167 
168 /* How many bytes are needed for a buffer of length 'l' */
169 static int xiph_len(int l)
170 {
171  return 1 + l / 255 + l;
172 }
173 
175 {
176  LibvorbisEncContext *s = avctx->priv_data;
177 
178  /* notify vorbisenc this is EOF */
179  if (s->dsp_initialized)
180  vorbis_analysis_wrote(&s->vd, 0);
181 
182  vorbis_block_clear(&s->vb);
183  vorbis_dsp_clear(&s->vd);
184  vorbis_info_clear(&s->vi);
185 
186  av_fifo_freep(&s->pkt_fifo);
187  ff_af_queue_close(&s->afq);
188  av_freep(&avctx->extradata);
189 
190  return 0;
191 }
192 
194 {
195  LibvorbisEncContext *s = avctx->priv_data;
196  ogg_packet header, header_comm, header_code;
197  uint8_t *p;
198  unsigned int offset;
199  int ret;
200 
201  vorbis_info_init(&s->vi);
202  if ((ret = libvorbis_setup(&s->vi, avctx))) {
203  av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
204  goto error;
205  }
206  if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
207  av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
208  ret = vorbis_error_to_averror(ret);
209  goto error;
210  }
211  s->dsp_initialized = 1;
212  if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
213  av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
214  ret = vorbis_error_to_averror(ret);
215  goto error;
216  }
217 
218  vorbis_comment_init(&s->vc);
219  if (!(avctx->flags & CODEC_FLAG_BITEXACT))
220  vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
221 
222  if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
223  &header_code))) {
224  ret = vorbis_error_to_averror(ret);
225  goto error;
226  }
227 
228  avctx->extradata_size = 1 + xiph_len(header.bytes) +
229  xiph_len(header_comm.bytes) +
230  header_code.bytes;
231  p = avctx->extradata = av_malloc(avctx->extradata_size +
233  if (!p) {
234  ret = AVERROR(ENOMEM);
235  goto error;
236  }
237  p[0] = 2;
238  offset = 1;
239  offset += av_xiphlacing(&p[offset], header.bytes);
240  offset += av_xiphlacing(&p[offset], header_comm.bytes);
241  memcpy(&p[offset], header.packet, header.bytes);
242  offset += header.bytes;
243  memcpy(&p[offset], header_comm.packet, header_comm.bytes);
244  offset += header_comm.bytes;
245  memcpy(&p[offset], header_code.packet, header_code.bytes);
246  offset += header_code.bytes;
247  av_assert0(offset == avctx->extradata_size);
248 
249  if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
250  av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
251  return ret;
252  }
253 
254  vorbis_comment_clear(&s->vc);
255 
257  ff_af_queue_init(avctx, &s->afq);
258 
260  if (!s->pkt_fifo) {
261  ret = AVERROR(ENOMEM);
262  goto error;
263  }
264 
265  return 0;
266 error:
267  libvorbis_encode_close(avctx);
268  return ret;
269 }
270 
272  const AVFrame *frame, int *got_packet_ptr)
273 {
274  LibvorbisEncContext *s = avctx->priv_data;
275  ogg_packet op;
276  int ret, duration;
277 
278  /* send samples to libvorbis */
279  if (frame) {
280  const int samples = frame->nb_samples;
281  float **buffer;
282  int c, channels = s->vi.channels;
283 
284  buffer = vorbis_analysis_buffer(&s->vd, samples);
285  for (c = 0; c < channels; c++) {
286  int co = (channels > 8) ? c :
288  memcpy(buffer[c], frame->extended_data[co],
289  samples * sizeof(*buffer[c]));
290  }
291  if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
292  av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
293  return vorbis_error_to_averror(ret);
294  }
295  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
296  return ret;
297  } else {
298  if (!s->eof && s->afq.frame_alloc)
299  if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
300  av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
301  return vorbis_error_to_averror(ret);
302  }
303  s->eof = 1;
304  }
305 
306  /* retrieve available packets from libvorbis */
307  while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
308  if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
309  break;
310  if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
311  break;
312 
313  /* add any available packets to the output packet buffer */
314  while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
315  if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
316  av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
317  return AVERROR_BUG;
318  }
319  av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
320  av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
321  }
322  if (ret < 0) {
323  av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
324  break;
325  }
326  }
327  if (ret < 0) {
328  av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
329  return vorbis_error_to_averror(ret);
330  }
331 
332  /* check for available packets */
333  if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
334  return 0;
335 
336  av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
337 
338  if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)) < 0)
339  return ret;
340  av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
341 
342  avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
343 
344  duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
345  if (duration > 0) {
346  /* we do not know encoder delay until we get the first packet from
347  * libvorbis, so we have to update the AudioFrameQueue counts */
348  if (!avctx->initial_padding && s->afq.frames) {
349  avctx->initial_padding = duration;
351  s->afq.frames->duration += duration;
352  if (s->afq.frames->pts != AV_NOPTS_VALUE)
353  s->afq.frames->pts -= duration;
355  }
356  ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
357  }
358 
359  *got_packet_ptr = 1;
360  return 0;
361 }
362 
364  .name = "libvorbis",
365  .long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
366  .type = AVMEDIA_TYPE_AUDIO,
367  .id = AV_CODEC_ID_VORBIS,
368  .priv_data_size = sizeof(LibvorbisEncContext),
370  .encode2 = libvorbis_encode_frame,
372  .capabilities = CODEC_CAP_DELAY,
373  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
375  .priv_class = &vorbis_class,
376  .defaults = defaults,
377 };