ensure that the private structure for pseudo channels is created without 'leaking...
[asterisk-bristuff.git] / formats / format_ogg_vorbis.c
blobff796e79ff512f245a3a194e3bdcffb3ccd5dc83
1 /*
2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 2005, Jeff Ollie
6 * See http://www.asterisk.org for more information about
7 * the Asterisk project. Please do not directly contact
8 * any of the maintainers of this project for assistance;
9 * the project provides a web site, mailing lists and IRC
10 * channels for your use.
12 * This program is free software, distributed under the terms of
13 * the GNU General Public License Version 2. See the LICENSE file
14 * at the top of the source tree.
17 /*! \file
19 * \brief OGG/Vorbis streams.
20 * \arg File name extension: ogg
21 * \ingroup formats
24 /* the order of these dependencies is important... it also specifies
25 the link order of the libraries during linking
28 /*** MODULEINFO
29 <depend>vorbis</depend>
30 <depend>ogg</depend>
31 ***/
33 #include "asterisk.h"
35 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
37 #include <sys/types.h>
38 #include <netinet/in.h>
39 #include <arpa/inet.h>
40 #include <stdlib.h>
41 #include <sys/time.h>
42 #include <stdio.h>
43 #include <unistd.h>
44 #include <errno.h>
45 #include <string.h>
47 #include <vorbis/codec.h>
48 #include <vorbis/vorbisenc.h>
50 #ifdef _WIN32
51 #include <io.h>
52 #include <fcntl.h>
53 #endif
55 #include "asterisk/lock.h"
56 #include "asterisk/channel.h"
57 #include "asterisk/file.h"
58 #include "asterisk/logger.h"
59 #include "asterisk/module.h"
62 * this is the number of samples we deal with. Samples are converted
63 * to SLINEAR so each one uses 2 bytes in the buffer.
65 #define SAMPLES_MAX 160
66 #define BUF_SIZE (2*SAMPLES_MAX)
68 #define BLOCK_SIZE 4096 /* used internally in the vorbis routines */
70 struct vorbis_desc { /* format specific parameters */
71 /* structures for handling the Ogg container */
72 ogg_sync_state oy;
73 ogg_stream_state os;
74 ogg_page og;
75 ogg_packet op;
77 /* structures for handling Vorbis audio data */
78 vorbis_info vi;
79 vorbis_comment vc;
80 vorbis_dsp_state vd;
81 vorbis_block vb;
83 /*! \brief Indicates whether this filestream is set up for reading or writing. */
84 int writing;
86 /*! \brief Indicates whether an End of Stream condition has been detected. */
87 int eos;
90 /*!
91 * \brief Create a new OGG/Vorbis filestream and set it up for reading.
92 * \param s File that points to on disk storage of the OGG/Vorbis data.
93 * \return The new filestream.
95 static int ogg_vorbis_open(struct ast_filestream *s)
97 int i;
98 int bytes;
99 int result;
100 char **ptr;
101 char *buffer;
102 struct vorbis_desc *tmp = (struct vorbis_desc *)s->_private;
104 tmp->writing = 0;
106 ogg_sync_init(&tmp->oy);
108 buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
109 bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
110 ogg_sync_wrote(&tmp->oy, bytes);
112 result = ogg_sync_pageout(&tmp->oy, &tmp->og);
113 if (result != 1) {
114 if(bytes < BLOCK_SIZE) {
115 ast_log(LOG_ERROR, "Run out of data...\n");
116 } else {
117 ast_log(LOG_ERROR, "Input does not appear to be an Ogg bitstream.\n");
119 ogg_sync_clear(&tmp->oy);
120 return -1;
123 ogg_stream_init(&tmp->os, ogg_page_serialno(&tmp->og));
124 vorbis_info_init(&tmp->vi);
125 vorbis_comment_init(&tmp->vc);
127 if (ogg_stream_pagein(&tmp->os, &tmp->og) < 0) {
128 ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n");
129 error:
130 ogg_stream_clear(&tmp->os);
131 vorbis_comment_clear(&tmp->vc);
132 vorbis_info_clear(&tmp->vi);
133 ogg_sync_clear(&tmp->oy);
134 return -1;
137 if (ogg_stream_packetout(&tmp->os, &tmp->op) != 1) {
138 ast_log(LOG_ERROR, "Error reading initial header packet.\n");
139 goto error;
142 if (vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op) < 0) {
143 ast_log(LOG_ERROR, "This Ogg bitstream does not contain Vorbis audio data.\n");
144 goto error;
147 for (i = 0; i < 2 ; ) {
148 while (i < 2) {
149 result = ogg_sync_pageout(&tmp->oy, &tmp->og);
150 if (result == 0)
151 break;
152 if (result == 1) {
153 ogg_stream_pagein(&tmp->os, &tmp->og);
154 while(i < 2) {
155 result = ogg_stream_packetout(&tmp->os,&tmp->op);
156 if(result == 0)
157 break;
158 if(result < 0) {
159 ast_log(LOG_ERROR, "Corrupt secondary header. Exiting.\n");
160 goto error;
162 vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op);
163 i++;
168 buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
169 bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
170 if (bytes == 0 && i < 2) {
171 ast_log(LOG_ERROR, "End of file before finding all Vorbis headers!\n");
172 goto error;
174 ogg_sync_wrote(&tmp->oy, bytes);
177 for (ptr = tmp->vc.user_comments; *ptr; ptr++)
178 ast_log(LOG_DEBUG, "OGG/Vorbis comment: %s\n", *ptr);
179 ast_log(LOG_DEBUG, "OGG/Vorbis bitstream is %d channel, %ldHz\n", tmp->vi.channels, tmp->vi.rate);
180 ast_log(LOG_DEBUG, "OGG/Vorbis file encoded by: %s\n", tmp->vc.vendor);
182 if (tmp->vi.channels != 1) {
183 ast_log(LOG_ERROR, "Only monophonic OGG/Vorbis files are currently supported!\n");
184 goto error;
187 if (tmp->vi.rate != DEFAULT_SAMPLE_RATE) {
188 ast_log(LOG_ERROR, "Only 8000Hz OGG/Vorbis files are currently supported!\n");
189 vorbis_block_clear(&tmp->vb);
190 vorbis_dsp_clear(&tmp->vd);
191 goto error;
194 vorbis_synthesis_init(&tmp->vd, &tmp->vi);
195 vorbis_block_init(&tmp->vd, &tmp->vb);
197 return 0;
201 * \brief Create a new OGG/Vorbis filestream and set it up for writing.
202 * \param s File pointer that points to on-disk storage.
203 * \param comment Comment that should be embedded in the OGG/Vorbis file.
204 * \return A new filestream.
206 static int ogg_vorbis_rewrite(struct ast_filestream *s,
207 const char *comment)
209 ogg_packet header;
210 ogg_packet header_comm;
211 ogg_packet header_code;
212 struct vorbis_desc *tmp = (struct vorbis_desc *)s->_private;
214 tmp->writing = 1;
216 vorbis_info_init(&tmp->vi);
218 if (vorbis_encode_init_vbr(&tmp->vi, 1, DEFAULT_SAMPLE_RATE, 0.4)) {
219 ast_log(LOG_ERROR, "Unable to initialize Vorbis encoder!\n");
220 return -1;
223 vorbis_comment_init(&tmp->vc);
224 vorbis_comment_add_tag(&tmp->vc, "ENCODER", "Asterisk PBX");
225 if (comment)
226 vorbis_comment_add_tag(&tmp->vc, "COMMENT", (char *) comment);
228 vorbis_analysis_init(&tmp->vd, &tmp->vi);
229 vorbis_block_init(&tmp->vd, &tmp->vb);
231 ogg_stream_init(&tmp->os, ast_random());
233 vorbis_analysis_headerout(&tmp->vd, &tmp->vc, &header, &header_comm,
234 &header_code);
235 ogg_stream_packetin(&tmp->os, &header);
236 ogg_stream_packetin(&tmp->os, &header_comm);
237 ogg_stream_packetin(&tmp->os, &header_code);
239 while (!tmp->eos) {
240 if (ogg_stream_flush(&tmp->os, &tmp->og) == 0)
241 break;
242 fwrite(tmp->og.header, 1, tmp->og.header_len, s->f);
243 fwrite(tmp->og.body, 1, tmp->og.body_len, s->f);
244 if (ogg_page_eos(&tmp->og))
245 tmp->eos = 1;
248 return 0;
252 * \brief Write out any pending encoded data.
253 * \param s An OGG/Vorbis filestream.
254 * \param f The file to write to.
256 static void write_stream(struct vorbis_desc *s, FILE *f)
258 while (vorbis_analysis_blockout(&s->vd, &s->vb) == 1) {
259 vorbis_analysis(&s->vb, NULL);
260 vorbis_bitrate_addblock(&s->vb);
262 while (vorbis_bitrate_flushpacket(&s->vd, &s->op)) {
263 ogg_stream_packetin(&s->os, &s->op);
264 while (!s->eos) {
265 if (ogg_stream_pageout(&s->os, &s->og) == 0) {
266 break;
268 fwrite(s->og.header, 1, s->og.header_len, f);
269 fwrite(s->og.body, 1, s->og.body_len, f);
270 if (ogg_page_eos(&s->og)) {
271 s->eos = 1;
279 * \brief Write audio data from a frame to an OGG/Vorbis filestream.
280 * \param fs An OGG/Vorbis filestream.
281 * \param f A frame containing audio to be written to the filestream.
282 * \return -1 if there was an error, 0 on success.
284 static int ogg_vorbis_write(struct ast_filestream *fs, struct ast_frame *f)
286 int i;
287 float **buffer;
288 short *data;
289 struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
291 if (!s->writing) {
292 ast_log(LOG_ERROR, "This stream is not set up for writing!\n");
293 return -1;
296 if (f->frametype != AST_FRAME_VOICE) {
297 ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
298 return -1;
300 if (f->subclass != AST_FORMAT_SLINEAR) {
301 ast_log(LOG_WARNING, "Asked to write non-SLINEAR frame (%d)!\n",
302 f->subclass);
303 return -1;
305 if (!f->datalen)
306 return -1;
308 data = (short *) f->data;
310 buffer = vorbis_analysis_buffer(&s->vd, f->samples);
312 for (i = 0; i < f->samples; i++)
313 buffer[0][i] = (double)data[i] / 32768.0;
315 vorbis_analysis_wrote(&s->vd, f->samples);
317 write_stream(s, fs->f);
319 return 0;
323 * \brief Close a OGG/Vorbis filestream.
324 * \param fs A OGG/Vorbis filestream.
326 static void ogg_vorbis_close(struct ast_filestream *fs)
328 struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
330 if (s->writing) {
331 /* Tell the Vorbis encoder that the stream is finished
332 * and write out the rest of the data */
333 vorbis_analysis_wrote(&s->vd, 0);
334 write_stream(s, fs->f);
337 ogg_stream_clear(&s->os);
338 vorbis_block_clear(&s->vb);
339 vorbis_dsp_clear(&s->vd);
340 vorbis_comment_clear(&s->vc);
341 vorbis_info_clear(&s->vi);
343 if (s->writing) {
344 ogg_sync_clear(&s->oy);
349 * \brief Get audio data.
350 * \param fs An OGG/Vorbis filestream.
351 * \param pcm Pointer to a buffere to store audio data in.
354 static int read_samples(struct ast_filestream *fs, float ***pcm)
356 int samples_in;
357 int result;
358 char *buffer;
359 int bytes;
360 struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
362 while (1) {
363 samples_in = vorbis_synthesis_pcmout(&s->vd, pcm);
364 if (samples_in > 0) {
365 return samples_in;
368 /* The Vorbis decoder needs more data... */
369 /* See ifOGG has any packets in the current page for the Vorbis decoder. */
370 result = ogg_stream_packetout(&s->os, &s->op);
371 if (result > 0) {
372 /* Yes OGG had some more packets for the Vorbis decoder. */
373 if (vorbis_synthesis(&s->vb, &s->op) == 0) {
374 vorbis_synthesis_blockin(&s->vd, &s->vb);
377 continue;
380 if (result < 0)
381 ast_log(LOG_WARNING,
382 "Corrupt or missing data at this page position; continuing...\n");
384 /* No more packets left in the current page... */
386 if (s->eos) {
387 /* No more pages left in the stream */
388 return -1;
391 while (!s->eos) {
392 /* See ifOGG has any pages in it's internal buffers */
393 result = ogg_sync_pageout(&s->oy, &s->og);
394 if (result > 0) {
395 /* Yes, OGG has more pages in it's internal buffers,
396 add the page to the stream state */
397 result = ogg_stream_pagein(&s->os, &s->og);
398 if (result == 0) {
399 /* Yes, got a new,valid page */
400 if (ogg_page_eos(&s->og)) {
401 s->eos = 1;
403 break;
405 ast_log(LOG_WARNING,
406 "Invalid page in the bitstream; continuing...\n");
409 if (result < 0)
410 ast_log(LOG_WARNING,
411 "Corrupt or missing data in bitstream; continuing...\n");
413 /* No, we need to read more data from the file descrptor */
414 /* get a buffer from OGG to read the data into */
415 buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
416 /* read more data from the file descriptor */
417 bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
418 /* Tell OGG how many bytes we actually read into the buffer */
419 ogg_sync_wrote(&s->oy, bytes);
420 if (bytes == 0) {
421 s->eos = 1;
428 * \brief Read a frame full of audio data from the filestream.
429 * \param fs The filestream.
430 * \param whennext Number of sample times to schedule the next call.
431 * \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
433 static struct ast_frame *ogg_vorbis_read(struct ast_filestream *fs,
434 int *whennext)
436 int clipflag = 0;
437 int i;
438 int j;
439 double accumulator[SAMPLES_MAX];
440 int val;
441 int samples_in;
442 int samples_out = 0;
443 struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
444 short *buf; /* SLIN data buffer */
446 fs->fr.frametype = AST_FRAME_VOICE;
447 fs->fr.subclass = AST_FORMAT_SLINEAR;
448 fs->fr.mallocd = 0;
449 AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
450 buf = (short *)(fs->fr.data); /* SLIN data buffer */
452 while (samples_out != SAMPLES_MAX) {
453 float **pcm;
454 int len = SAMPLES_MAX - samples_out;
456 /* See ifVorbis decoder has some audio data for us ... */
457 samples_in = read_samples(fs, &pcm);
458 if (samples_in <= 0)
459 break;
461 /* Got some audio data from Vorbis... */
462 /* Convert the float audio data to 16-bit signed linear */
464 clipflag = 0;
465 if (samples_in > len)
466 samples_in = len;
467 for (j = 0; j < samples_in; j++)
468 accumulator[j] = 0.0;
470 for (i = 0; i < s->vi.channels; i++) {
471 float *mono = pcm[i];
472 for (j = 0; j < samples_in; j++)
473 accumulator[j] += mono[j];
476 for (j = 0; j < samples_in; j++) {
477 val = accumulator[j] * 32767.0 / s->vi.channels;
478 if (val > 32767) {
479 val = 32767;
480 clipflag = 1;
481 } else if (val < -32768) {
482 val = -32768;
483 clipflag = 1;
485 buf[samples_out + j] = val;
488 if (clipflag)
489 ast_log(LOG_WARNING, "Clipping in frame %ld\n", (long) (s->vd.sequence));
490 /* Tell the Vorbis decoder how many samples we actually used. */
491 vorbis_synthesis_read(&s->vd, samples_in);
492 samples_out += samples_in;
495 if (samples_out > 0) {
496 fs->fr.datalen = samples_out * 2;
497 fs->fr.samples = samples_out;
498 *whennext = samples_out;
500 return &fs->fr;
501 } else {
502 return NULL;
507 * \brief Trucate an OGG/Vorbis filestream.
508 * \param s The filestream to truncate.
509 * \return 0 on success, -1 on failure.
512 static int ogg_vorbis_trunc(struct ast_filestream *s)
514 ast_log(LOG_WARNING, "Truncation is not supported on OGG/Vorbis streams!\n");
515 return -1;
519 * \brief Seek to a specific position in an OGG/Vorbis filestream.
520 * \param s The filestream to truncate.
521 * \param sample_offset New position for the filestream, measured in 8KHz samples.
522 * \param whence Location to measure
523 * \return 0 on success, -1 on failure.
525 static int ogg_vorbis_seek(struct ast_filestream *s, off_t sample_offset, int whence)
527 ast_log(LOG_WARNING, "Seeking is not supported on OGG/Vorbis streams!\n");
528 return -1;
531 static off_t ogg_vorbis_tell(struct ast_filestream *s)
533 ast_log(LOG_WARNING, "Telling is not supported on OGG/Vorbis streams!\n");
534 return -1;
537 static const struct ast_format vorbis_f = {
538 .name = "ogg_vorbis",
539 .exts = "ogg",
540 .format = AST_FORMAT_SLINEAR,
541 .open = ogg_vorbis_open,
542 .rewrite = ogg_vorbis_rewrite,
543 .write = ogg_vorbis_write,
544 .seek = ogg_vorbis_seek,
545 .trunc = ogg_vorbis_trunc,
546 .tell = ogg_vorbis_tell,
547 .read = ogg_vorbis_read,
548 .close = ogg_vorbis_close,
549 .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
550 .desc_size = sizeof(struct vorbis_desc),
553 static int load_module(void)
555 return ast_format_register(&vorbis_f);
558 static int unload_module(void)
560 return ast_format_unregister(vorbis_f.name);
563 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OGG/Vorbis audio");