libvorbis.c
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1 /*
2  * copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
3  *
4  * This file is part of Libav.
5  *
6  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
27 #include <vorbis/vorbisenc.h>
28 
29 #include "libavutil/fifo.h"
30 #include "libavutil/opt.h"
31 #include "avcodec.h"
32 #include "audio_frame_queue.h"
33 #include "bytestream.h"
34 #include "internal.h"
35 #include "vorbis.h"
36 #include "vorbis_parser.h"
37 
38 #undef NDEBUG
39 #include <assert.h>
40 
41 /* Number of samples the user should send in each call.
42  * This value is used because it is the LCD of all possible frame sizes, so
43  * an output packet will always start at the same point as one of the input
44  * packets.
45  */
46 #define OGGVORBIS_FRAME_SIZE 64
47 
48 #define BUFFER_SIZE (1024 * 64)
49 
50 typedef struct OggVorbisContext {
52  vorbis_info vi;
53  vorbis_dsp_state vd;
54  vorbis_block vb;
56  int eof;
58  vorbis_comment vc;
60  double iblock;
64 
65 static const AVOption options[] = {
66  { "iblock", "Sets the impulse block bias", offsetof(OggVorbisContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
67  { NULL }
68 };
69 
70 static const AVCodecDefault defaults[] = {
71  { "b", "0" },
72  { NULL },
73 };
74 
75 static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
76 
77 
78 static int vorbis_error_to_averror(int ov_err)
79 {
80  switch (ov_err) {
81  case OV_EFAULT: return AVERROR_BUG;
82  case OV_EINVAL: return AVERROR(EINVAL);
83  case OV_EIMPL: return AVERROR(EINVAL);
84  default: return AVERROR_UNKNOWN;
85  }
86 }
87 
88 static av_cold int oggvorbis_init_encoder(vorbis_info *vi,
89  AVCodecContext *avctx)
90 {
91  OggVorbisContext *s = avctx->priv_data;
92  double cfreq;
93  int ret;
94 
95  if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) {
96  /* variable bitrate
97  * NOTE: we use the oggenc range of -1 to 10 for global_quality for
98  * user convenience, but libvorbis uses -0.1 to 1.0.
99  */
100  float q = avctx->global_quality / (float)FF_QP2LAMBDA;
101  /* default to 3 if the user did not set quality or bitrate */
102  if (!(avctx->flags & CODEC_FLAG_QSCALE))
103  q = 3.0;
104  if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
105  avctx->sample_rate,
106  q / 10.0)))
107  goto error;
108  } else {
109  int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
110  int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
111 
112  /* average bitrate */
113  if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
114  avctx->sample_rate, maxrate,
115  avctx->bit_rate, minrate)))
116  goto error;
117 
118  /* variable bitrate by estimate, disable slow rate management */
119  if (minrate == -1 && maxrate == -1)
120  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
121  goto error;
122  }
123 
124  /* cutoff frequency */
125  if (avctx->cutoff > 0) {
126  cfreq = avctx->cutoff / 1000.0;
127  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
128  goto error;
129  }
130 
131  /* impulse block bias */
132  if (s->iblock) {
133  if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
134  goto error;
135  }
136 
137  if ((ret = vorbis_encode_setup_init(vi)))
138  goto error;
139 
140  return 0;
141 error:
142  return vorbis_error_to_averror(ret);
143 }
144 
145 /* How many bytes are needed for a buffer of length 'l' */
146 static int xiph_len(int l)
147 {
148  return 1 + l / 255 + l;
149 }
150 
152 {
153  OggVorbisContext *s = avctx->priv_data;
154 
155  /* notify vorbisenc this is EOF */
156  if (s->dsp_initialized)
157  vorbis_analysis_wrote(&s->vd, 0);
158 
159  vorbis_block_clear(&s->vb);
160  vorbis_dsp_clear(&s->vd);
161  vorbis_info_clear(&s->vi);
162 
164  ff_af_queue_close(&s->afq);
165 #if FF_API_OLD_ENCODE_AUDIO
166  av_freep(&avctx->coded_frame);
167 #endif
168  av_freep(&avctx->extradata);
169 
170  return 0;
171 }
172 
174 {
175  OggVorbisContext *s = avctx->priv_data;
176  ogg_packet header, header_comm, header_code;
177  uint8_t *p;
178  unsigned int offset;
179  int ret;
180 
181  vorbis_info_init(&s->vi);
182  if ((ret = oggvorbis_init_encoder(&s->vi, avctx))) {
183  av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
184  goto error;
185  }
186  if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
187  av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
188  ret = vorbis_error_to_averror(ret);
189  goto error;
190  }
191  s->dsp_initialized = 1;
192  if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
193  av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
194  ret = vorbis_error_to_averror(ret);
195  goto error;
196  }
197 
198  vorbis_comment_init(&s->vc);
199  vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
200 
201  if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
202  &header_code))) {
203  ret = vorbis_error_to_averror(ret);
204  goto error;
205  }
206 
207  avctx->extradata_size = 1 + xiph_len(header.bytes) +
208  xiph_len(header_comm.bytes) +
209  header_code.bytes;
210  p = avctx->extradata = av_malloc(avctx->extradata_size +
212  if (!p) {
213  ret = AVERROR(ENOMEM);
214  goto error;
215  }
216  p[0] = 2;
217  offset = 1;
218  offset += av_xiphlacing(&p[offset], header.bytes);
219  offset += av_xiphlacing(&p[offset], header_comm.bytes);
220  memcpy(&p[offset], header.packet, header.bytes);
221  offset += header.bytes;
222  memcpy(&p[offset], header_comm.packet, header_comm.bytes);
223  offset += header_comm.bytes;
224  memcpy(&p[offset], header_code.packet, header_code.bytes);
225  offset += header_code.bytes;
226  assert(offset == avctx->extradata_size);
227 
228  if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
229  av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
230  return ret;
231  }
232 
233  vorbis_comment_clear(&s->vc);
234 
236  ff_af_queue_init(avctx, &s->afq);
237 
239  if (!s->pkt_fifo) {
240  ret = AVERROR(ENOMEM);
241  goto error;
242  }
243 
244 #if FF_API_OLD_ENCODE_AUDIO
245  avctx->coded_frame = avcodec_alloc_frame();
246  if (!avctx->coded_frame) {
247  ret = AVERROR(ENOMEM);
248  goto error;
249  }
250 #endif
251 
252  return 0;
253 error:
254  oggvorbis_encode_close(avctx);
255  return ret;
256 }
257 
259  const AVFrame *frame, int *got_packet_ptr)
260 {
261  OggVorbisContext *s = avctx->priv_data;
262  ogg_packet op;
263  int ret, duration;
264 
265  /* send samples to libvorbis */
266  if (frame) {
267  const int samples = frame->nb_samples;
268  float **buffer;
269  int c, channels = s->vi.channels;
270 
271  buffer = vorbis_analysis_buffer(&s->vd, samples);
272  for (c = 0; c < channels; c++) {
273  int co = (channels > 8) ? c :
275  memcpy(buffer[c], frame->extended_data[co],
276  samples * sizeof(*buffer[c]));
277  }
278  if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
279  av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
280  return vorbis_error_to_averror(ret);
281  }
282  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
283  return ret;
284  } else {
285  if (!s->eof)
286  if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
287  av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
288  return vorbis_error_to_averror(ret);
289  }
290  s->eof = 1;
291  }
292 
293  /* retrieve available packets from libvorbis */
294  while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
295  if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
296  break;
297  if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
298  break;
299 
300  /* add any available packets to the output packet buffer */
301  while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
302  if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
303  av_log(avctx, AV_LOG_ERROR, "packet buffer is too small");
304  return AVERROR_BUG;
305  }
306  av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
307  av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
308  }
309  if (ret < 0) {
310  av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
311  break;
312  }
313  }
314  if (ret < 0) {
315  av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
316  return vorbis_error_to_averror(ret);
317  }
318 
319  /* check for available packets */
320  if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
321  return 0;
322 
323  av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
324 
325  if ((ret = ff_alloc_packet(avpkt, op.bytes))) {
326  av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
327  return ret;
328  }
329  av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
330 
331  avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
332 
333  duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
334  if (duration > 0) {
335  /* we do not know encoder delay until we get the first packet from
336  * libvorbis, so we have to update the AudioFrameQueue counts */
337  if (!avctx->delay) {
338  avctx->delay = duration;
341  }
342  ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
343  }
344 
345  *got_packet_ptr = 1;
346  return 0;
347 }
348 
350  .name = "libvorbis",
351  .type = AVMEDIA_TYPE_AUDIO,
352  .id = AV_CODEC_ID_VORBIS,
353  .priv_data_size = sizeof(OggVorbisContext),
355  .encode2 = oggvorbis_encode_frame,
357  .capabilities = CODEC_CAP_DELAY,
358  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
360  .long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
361  .priv_class = &class,
362  .defaults = defaults,
363 };
static const AVCodecDefault defaults[]
Definition: libvorbis.c:70
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:61
#define OGGVORBIS_FRAME_SIZE
Definition: libvorbis.c:46
static int16_t * samples
This structure describes decoded (raw) audio or video data.
Definition: avcodec.h:989
AVOption.
Definition: opt.h:233
static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libvorbis.c:258
int avpriv_vorbis_parse_frame(VorbisParseContext *s, const uint8_t *buf, int buf_size)
Get the duration for a Vorbis packet.
AVFrame * coded_frame
the picture in the bitstream
Definition: avcodec.h:2725
struct OggVorbisContext OggVorbisContext
int size
Definition: avcodec.h:916
int eof
end-of-file flag
Definition: libvorbis.c:56
AVCodec.
Definition: avcodec.h:2960
static int64_t duration
Definition: avplay.c:249
int av_fifo_generic_write(AVFifoBuffer *f, void *src, int size, int(*func)(void *, void *, int))
Feed data from a user-supplied callback to an AVFifoBuffer.
Definition: fifo.c:82
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:151
AVFifoBuffer * pkt_fifo
output packet buffer
Definition: libvorbis.c:55
uint8_t
AVOptions.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1454
static const AVOption options[]
Definition: libvorbis.c:65
uint8_t * data
Definition: avcodec.h:915
Vorbis audio parser.
void av_fifo_free(AVFifoBuffer *f)
Free an AVFifoBuffer.
Definition: fifo.c:38
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:937
static int init(AVCodecParserContext *s)
Definition: h264_parser.c:335
float, planar
Definition: samplefmt.h:60
AVCodec ff_libvorbis_encoder
Definition: libvorbis.c:349
int dsp_initialized
vd has been initialized
Definition: libvorbis.c:57
sample_fmts
Definition: avconv_filter.c:63
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:88
static int vorbis_error_to_averror(int ov_err)
Definition: libvorbis.c:78
int av_fifo_generic_read(AVFifoBuffer *f, void *dest, int buf_size, void(*func)(void *, void *, int))
Feed data from an AVFifoBuffer to a user-supplied callback.
Definition: fifo.c:105
int flags
CODEC_FLAG_*.
Definition: avcodec.h:1434
int rc_max_rate
maximum bitrate
Definition: avcodec.h:2339
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:146
const char * name
Name of the codec implementation.
Definition: avcodec.h:2967
double iblock
impulse block bias option
Definition: libvorbis.c:60
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
vorbis_dsp_state vd
DSP state used for analysis.
Definition: libvorbis.c:53
AVFrame * avcodec_alloc_frame(void)
Allocate an AVFrame and set its fields to default values.
Definition: utils.c:616
int bit_rate
the average bitrate
Definition: avcodec.h:1404
static char buffer[20]
Definition: seek-test.c:31
int ff_alloc_packet(AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
Definition: utils.c:878
LIBAVUTIL_VERSION_INT
Definition: eval.c:52
static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avctx)
Definition: libvorbis.c:88
AVClass * av_class
class for AVOptions
Definition: libvorbis.c:51
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2124
const uint8_t ff_vorbis_encoding_channel_layout_offsets[8][8]
Definition: vorbis_data.c:37
NULL
Definition: eval.c:52
int av_fifo_space(AVFifoBuffer *f)
Return the amount of space in bytes in the AVFifoBuffer, that is the amount of data you can write int...
Definition: fifo.c:57
av_default_item_name
Definition: libvorbis.c:75
VorbisParseContext vp
parse context to get durations
Definition: libvorbis.c:61
external API header
AV_SAMPLE_FMT_NONE
Definition: avconv_filter.c:63
int sample_rate
samples per second
Definition: avcodec.h:2104
main external API structure.
Definition: avcodec.h:1339
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:326
a very simple circular buffer FIFO implementation
AudioFrameQueue afq
frame queue for timestamps
Definition: libvorbis.c:62
int extradata_size
Definition: avcodec.h:1455
unsigned int av_xiphlacing(unsigned char *s, unsigned int v)
Encode extradata length to a buffer.
Definition: utils.c:1984
Describe the class of an AVClass context structure.
Definition: log.h:33
vorbis_block vb
vorbis_block used for analysis
Definition: libvorbis.c:54
vorbis_info vi
vorbis_info used during init
Definition: libvorbis.c:52
#define BUFFER_SIZE
Definition: libvorbis.c:48
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1420
static int op(uint8_t **dst, const uint8_t *dst_end, GetByteContext *gb, int pixel, int count, int *x, int width, int linesize)
Perform decode operation.
Definition: anm.c:72
int av_fifo_size(AVFifoBuffer *f)
Return the amount of data in bytes in the AVFifoBuffer, that is the amount of data you can read from ...
Definition: fifo.c:52
common internal api header.
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
void * priv_data
Definition: avcodec.h:1382
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:2148
static int ogg_packet(AVFormatContext *s, int *str, int *dstart, int *dsize, int64_t *fpos)
Definition: oggdec.c:336
AVFifoBuffer * av_fifo_alloc(unsigned int size)
Initialize an AVFifoBuffer.
Definition: fifo.c:25
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
Definition: libvorbis.c:151
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int *duration)
Remove frame(s) from the queue.
int channels
number of audio channels
Definition: avcodec.h:2105
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
int avpriv_vorbis_parse_extradata(AVCodecContext *avctx, VorbisParseContext *s)
Initialize the Vorbis parser using headers in the extradata.
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
Definition: internal.h:140
int rc_min_rate
minimum bitrate
Definition: avcodec.h:2346
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: avcodec.h:1028
static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
Definition: libvorbis.c:173
This structure stores compressed data.
Definition: avcodec.h:898
int delay
Codec delay.
Definition: avcodec.h:1497
int nb_samples
number of audio samples (per channel) described by this frame
Definition: avcodec.h:1042
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:908
ogg_packet op
ogg packet
Definition: libvorbis.c:59
static int xiph_len(int l)
Definition: libvorbis.c:146
vorbis_comment vc
VorbisComment info.
Definition: libvorbis.c:58