Fix a memory leak in func_curl. Every thread that used this function leaked
[asterisk-bristuff.git] / main / audiohook.c
blob809c176295e80b80cc0d522d1b7b727e8f86fde4
1 /*
2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2007, Digium, Inc.
6 * Joshua Colp <jcolp@digium.com>
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
19 /*! \file
21 * \brief Audiohooks Architecture
23 * \author Joshua Colp <jcolp@digium.com>
26 #include "asterisk.h"
28 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
30 #include <stdio.h>
31 #include <stdlib.h>
32 #include <string.h>
33 #include <signal.h>
34 #include <errno.h>
35 #include <unistd.h>
37 #include "asterisk/logger.h"
38 #include "asterisk/channel.h"
39 #include "asterisk/options.h"
40 #include "asterisk/utils.h"
41 #include "asterisk/lock.h"
42 #include "asterisk/linkedlists.h"
43 #include "asterisk/audiohook.h"
44 #include "asterisk/slinfactory.h"
45 #include "asterisk/frame.h"
46 #include "asterisk/translate.h"
48 struct ast_audiohook_translate {
49 struct ast_trans_pvt *trans_pvt;
50 int format;
53 struct ast_audiohook_list {
54 struct ast_audiohook_translate in_translate[2];
55 struct ast_audiohook_translate out_translate[2];
56 AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
57 AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
58 AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
61 /*! \brief Initialize an audiohook structure
62 * \param audiohook Audiohook structure
63 * \param type
64 * \param source
65 * \return Returns 0 on success, -1 on failure
67 int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source)
69 /* Need to keep the type and source */
70 audiohook->type = type;
71 audiohook->source = source;
73 /* Initialize lock that protects our audiohook */
74 ast_mutex_init(&audiohook->lock);
75 ast_cond_init(&audiohook->trigger, NULL);
77 /* Setup the factories that are needed for this audiohook type */
78 switch (type) {
79 case AST_AUDIOHOOK_TYPE_SPY:
80 ast_slinfactory_init(&audiohook->read_factory);
81 case AST_AUDIOHOOK_TYPE_WHISPER:
82 ast_slinfactory_init(&audiohook->write_factory);
83 break;
84 default:
85 break;
88 /* Since we are just starting out... this audiohook is new */
89 audiohook->status = AST_AUDIOHOOK_STATUS_NEW;
91 return 0;
94 /*! \brief Destroys an audiohook structure
95 * \param audiohook Audiohook structure
96 * \return Returns 0 on success, -1 on failure
98 int ast_audiohook_destroy(struct ast_audiohook *audiohook)
100 /* Drop the factories used by this audiohook type */
101 switch (audiohook->type) {
102 case AST_AUDIOHOOK_TYPE_SPY:
103 ast_slinfactory_destroy(&audiohook->read_factory);
104 case AST_AUDIOHOOK_TYPE_WHISPER:
105 ast_slinfactory_destroy(&audiohook->write_factory);
106 break;
107 default:
108 break;
111 /* Destroy translation path if present */
112 if (audiohook->trans_pvt)
113 ast_translator_free_path(audiohook->trans_pvt);
115 /* Lock and trigger be gone! */
116 ast_cond_destroy(&audiohook->trigger);
117 ast_mutex_destroy(&audiohook->lock);
119 return 0;
122 /*! \brief Writes a frame into the audiohook structure
123 * \param audiohook Audiohook structure
124 * \param direction Direction the audio frame came from
125 * \param frame Frame to write in
126 * \return Returns 0 on success, -1 on failure
128 int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
130 struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
131 struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
132 struct timeval *time = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *time;
134 /* Update last feeding time to be current */
135 *time = ast_tvnow();
137 /* If we are using a sync trigger and this factory suddenly got audio fed in after a lapse, then flush both factories to ensure they remain in sync */
138 if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && ast_slinfactory_available(other_factory) && (ast_tvdiff_ms(*time, previous_time) > (ast_slinfactory_available(other_factory) / 8))) {
139 if (option_debug)
140 ast_log(LOG_DEBUG, "Flushing audiohook %p so it remains in sync\n", audiohook);
141 ast_slinfactory_flush(factory);
142 ast_slinfactory_flush(other_factory);
145 /* Write frame out to respective factory */
146 ast_slinfactory_feed(factory, frame);
148 /* If we need to notify the respective handler of this audiohook, do so */
149 if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
150 ast_cond_signal(&audiohook->trigger);
151 } else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
152 ast_cond_signal(&audiohook->trigger);
153 } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
154 ast_cond_signal(&audiohook->trigger);
157 return 0;
160 static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
162 struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
163 int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
164 short buf[samples];
165 struct ast_frame frame = {
166 .frametype = AST_FRAME_VOICE,
167 .subclass = AST_FORMAT_SLINEAR,
168 .data = buf,
169 .datalen = sizeof(buf),
170 .samples = samples,
173 /* Ensure the factory is able to give us the samples we want */
174 if (samples > ast_slinfactory_available(factory))
175 return NULL;
177 /* Read data in from factory */
178 if (!ast_slinfactory_read(factory, buf, samples))
179 return NULL;
181 /* If a volume adjustment needs to be applied apply it */
182 if (vol)
183 ast_frame_adjust_volume(&frame, vol);
185 return ast_frdup(&frame);
188 static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
190 int i = 0, usable_read, usable_write;
191 short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
192 struct ast_frame frame = {
193 .frametype = AST_FRAME_VOICE,
194 .subclass = AST_FORMAT_SLINEAR,
195 .data = NULL,
196 .datalen = sizeof(buf1),
197 .samples = samples,
200 /* Make sure both factories have the required samples */
201 usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
202 usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
204 if (!usable_read && !usable_write) {
205 /* If both factories are unusable bail out */
206 if (option_debug)
207 ast_log(LOG_DEBUG, "Read factory %p and write factory %p both fail to provide %zd samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
208 return NULL;
211 /* If we want to provide only a read factory make sure we aren't waiting for other audio */
212 if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
213 if (option_debug > 2)
214 ast_log(LOG_DEBUG, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
215 return NULL;
218 /* If we want to provide only a write factory make sure we aren't waiting for other audio */
219 if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
220 if (option_debug > 2)
221 ast_log(LOG_DEBUG, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
222 return NULL;
225 /* Start with the read factory... if there are enough samples, read them in */
226 if (usable_read) {
227 if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
228 read_buf = buf1;
229 /* Adjust read volume if need be */
230 if (audiohook->options.read_volume) {
231 int count = 0;
232 short adjust_value = abs(audiohook->options.read_volume);
233 for (count = 0; count < samples; count++) {
234 if (audiohook->options.read_volume > 0)
235 ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
236 else if (audiohook->options.read_volume < 0)
237 ast_slinear_saturated_divide(&buf1[count], &adjust_value);
241 } else if (option_debug)
242 ast_log(LOG_DEBUG, "Failed to get %zd samples from read factory %p\n", samples, &audiohook->read_factory);
244 /* Move on to the write factory... if there are enough samples, read them in */
245 if (usable_write) {
246 if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
247 write_buf = buf2;
248 /* Adjust write volume if need be */
249 if (audiohook->options.write_volume) {
250 int count = 0;
251 short adjust_value = abs(audiohook->options.write_volume);
252 for (count = 0; count < samples; count++) {
253 if (audiohook->options.write_volume > 0)
254 ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
255 else if (audiohook->options.write_volume < 0)
256 ast_slinear_saturated_divide(&buf2[count], &adjust_value);
260 } else if (option_debug)
261 ast_log(LOG_DEBUG, "Failed to get %zd samples from write factory %p\n", samples, &audiohook->write_factory);
263 /* Basically we figure out which buffer to use... and if mixing can be done here */
264 if (!read_buf && !write_buf)
265 return NULL;
266 else if (read_buf && write_buf) {
267 for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
268 ast_slinear_saturated_add(data1, data2);
269 final_buf = buf1;
270 } else if (read_buf)
271 final_buf = buf1;
272 else if (write_buf)
273 final_buf = buf2;
275 /* Make the final buffer part of the frame, so it gets duplicated fine */
276 frame.data = final_buf;
278 /* Yahoo, a combined copy of the audio! */
279 return ast_frdup(&frame);
282 /*! \brief Reads a frame in from the audiohook structure
283 * \param audiohook Audiohook structure
284 * \param samples Number of samples wanted
285 * \param direction Direction the audio frame came from
286 * \param format Format of frame remote side wants back
287 * \return Returns frame on success, NULL on failure
289 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format)
291 struct ast_frame *read_frame = NULL, *final_frame = NULL;
293 if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
294 return NULL;
296 /* If they don't want signed linear back out, we'll have to send it through the translation path */
297 if (format != AST_FORMAT_SLINEAR) {
298 /* Rebuild translation path if different format then previously */
299 if (audiohook->format != format) {
300 if (audiohook->trans_pvt) {
301 ast_translator_free_path(audiohook->trans_pvt);
302 audiohook->trans_pvt = NULL;
304 /* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */
305 if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
306 ast_frfree(read_frame);
307 return NULL;
310 /* Convert to requested format, and allow the read in frame to be freed */
311 final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
312 } else {
313 final_frame = read_frame;
316 return final_frame;
319 /*! \brief Attach audiohook to channel
320 * \param chan Channel
321 * \param audiohook Audiohook structure
322 * \return Returns 0 on success, -1 on failure
324 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
326 ast_channel_lock(chan);
328 if (!chan->audiohooks) {
329 /* Whoops... allocate a new structure */
330 if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
331 ast_channel_unlock(chan);
332 return -1;
334 AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
335 AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
336 AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
339 /* Drop into respective list */
340 if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
341 AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
342 else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
343 AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
344 else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
345 AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
347 /* Change status over to running since it is now attached */
348 audiohook->status = AST_AUDIOHOOK_STATUS_RUNNING;
350 ast_channel_unlock(chan);
352 return 0;
355 /*! \brief Detach audiohook from channel
356 * \param audiohook Audiohook structure
357 * \return Returns 0 on success, -1 on failure
359 int ast_audiohook_detach(struct ast_audiohook *audiohook)
361 if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
362 return 0;
364 audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
366 while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
367 ast_audiohook_trigger_wait(audiohook);
369 return 0;
372 /*! \brief Detach audiohooks from list and destroy said list
373 * \param audiohook_list List of audiohooks
374 * \return Returns 0 on success, -1 on failure
376 int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
378 int i = 0;
379 struct ast_audiohook *audiohook = NULL;
381 /* Drop any spies */
382 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
383 ast_audiohook_lock(audiohook);
384 AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list);
385 audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
386 ast_cond_signal(&audiohook->trigger);
387 ast_audiohook_unlock(audiohook);
389 AST_LIST_TRAVERSE_SAFE_END
391 /* Drop any whispering sources */
392 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
393 ast_audiohook_lock(audiohook);
394 AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list);
395 audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
396 ast_cond_signal(&audiohook->trigger);
397 ast_audiohook_unlock(audiohook);
399 AST_LIST_TRAVERSE_SAFE_END
401 /* Drop any manipulaters */
402 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
403 ast_audiohook_lock(audiohook);
404 AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
405 audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
406 ast_audiohook_unlock(audiohook);
407 audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
409 AST_LIST_TRAVERSE_SAFE_END
411 /* Drop translation paths if present */
412 for (i = 0; i < 2; i++) {
413 if (audiohook_list->in_translate[i].trans_pvt)
414 ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
415 if (audiohook_list->out_translate[i].trans_pvt)
416 ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
419 /* Free ourselves */
420 ast_free(audiohook_list);
422 return 0;
425 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
427 struct ast_audiohook *audiohook = NULL;
429 AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
430 if (!strcasecmp(audiohook->source, source))
431 return audiohook;
434 AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
435 if (!strcasecmp(audiohook->source, source))
436 return audiohook;
439 AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
440 if (!strcasecmp(audiohook->source, source))
441 return audiohook;
444 return NULL;
447 /*! \brief Detach specified source audiohook from channel
448 * \param chan Channel to detach from
449 * \param source Name of source to detach
450 * \return Returns 0 on success, -1 on failure
452 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
454 struct ast_audiohook *audiohook = NULL;
456 ast_channel_lock(chan);
458 /* Ensure the channel has audiohooks on it */
459 if (!chan->audiohooks) {
460 ast_channel_unlock(chan);
461 return -1;
464 audiohook = find_audiohook_by_source(chan->audiohooks, source);
466 ast_channel_unlock(chan);
468 if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
469 audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
471 return (audiohook ? 0 : -1);
474 /*! \brief Pass a DTMF frame off to be handled by the audiohook core
475 * \param chan Channel that the list is coming off of
476 * \param audiohook_list List of audiohooks
477 * \param direction Direction frame is coming in from
478 * \param frame The frame itself
479 * \return Return frame on success, NULL on failure
481 static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
483 struct ast_audiohook *audiohook = NULL;
485 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
486 ast_audiohook_lock(audiohook);
487 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
488 AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
489 audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
490 ast_audiohook_unlock(audiohook);
491 audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
492 continue;
494 if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
495 audiohook->manipulate_callback(audiohook, chan, frame, direction);
496 ast_audiohook_unlock(audiohook);
498 AST_LIST_TRAVERSE_SAFE_END
500 return frame;
503 /*! \brief Pass an AUDIO frame off to be handled by the audiohook core
504 * \param chan Channel that the list is coming off of
505 * \param audiohook_list List of audiohooks
506 * \param direction Direction frame is coming in from
507 * \param frame The frame itself
508 * \return Return frame on success, NULL on failure
510 static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
512 struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
513 struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
514 struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
515 struct ast_audiohook *audiohook = NULL;
516 int samples = frame->samples;
518 /* If the frame coming in is not signed linear we have to send it through the in_translate path */
519 if (frame->subclass != AST_FORMAT_SLINEAR) {
520 if (in_translate->format != frame->subclass) {
521 if (in_translate->trans_pvt)
522 ast_translator_free_path(in_translate->trans_pvt);
523 if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass)))
524 return frame;
525 in_translate->format = frame->subclass;
527 if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
528 return frame;
531 /* Queue up signed linear frame to each spy */
532 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
533 ast_audiohook_lock(audiohook);
534 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
535 AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list);
536 audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
537 ast_cond_signal(&audiohook->trigger);
538 ast_audiohook_unlock(audiohook);
539 continue;
541 ast_audiohook_write_frame(audiohook, direction, middle_frame);
542 ast_audiohook_unlock(audiohook);
544 AST_LIST_TRAVERSE_SAFE_END
546 /* If this frame is being written out to the channel then we need to use whisper sources */
547 if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
548 int i = 0;
549 short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
550 memset(&combine_buf, 0, sizeof(combine_buf));
551 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
552 ast_audiohook_lock(audiohook);
553 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
554 AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list);
555 audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
556 ast_cond_signal(&audiohook->trigger);
557 ast_audiohook_unlock(audiohook);
558 continue;
560 if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
561 /* Take audio from this whisper source and combine it into our main buffer */
562 for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
563 ast_slinear_saturated_add(data1, data2);
565 ast_audiohook_unlock(audiohook);
567 AST_LIST_TRAVERSE_SAFE_END
568 /* We take all of the combined whisper sources and combine them into the audio being written out */
569 for (i = 0, data1 = middle_frame->data, data2 = combine_buf; i < samples; i++, data1++, data2++)
570 ast_slinear_saturated_add(data1, data2);
571 end_frame = middle_frame;
574 /* Pass off frame to manipulate audiohooks */
575 if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
576 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
577 ast_audiohook_lock(audiohook);
578 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
579 AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
580 audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
581 ast_audiohook_unlock(audiohook);
582 /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
583 audiohook->manipulate_callback(audiohook, chan, NULL, direction);
584 continue;
586 /* Feed in frame to manipulation */
587 audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
588 ast_audiohook_unlock(audiohook);
590 AST_LIST_TRAVERSE_SAFE_END
591 end_frame = middle_frame;
594 /* Now we figure out what to do with our end frame (whether to transcode or not) */
595 if (middle_frame == end_frame) {
596 /* Middle frame was modified and became the end frame... let's see if we need to transcode */
597 if (end_frame->subclass != start_frame->subclass) {
598 if (out_translate->format != start_frame->subclass) {
599 if (out_translate->trans_pvt)
600 ast_translator_free_path(out_translate->trans_pvt);
601 if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass, AST_FORMAT_SLINEAR))) {
602 /* We can't transcode this... drop our middle frame and return the original */
603 ast_frfree(middle_frame);
604 return start_frame;
606 out_translate->format = start_frame->subclass;
608 /* Transcode from our middle (signed linear) frame to new format of the frame that came in */
609 if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
610 /* Failed to transcode the frame... drop it and return the original */
611 ast_frfree(middle_frame);
612 return start_frame;
614 /* Here's the scoop... middle frame is no longer of use to us */
615 ast_frfree(middle_frame);
617 } else {
618 /* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
619 ast_frfree(middle_frame);
622 return end_frame;
625 /*! \brief Pass a frame off to be handled by the audiohook core
626 * \param chan Channel that the list is coming off of
627 * \param audiohook_list List of audiohooks
628 * \param direction Direction frame is coming in from
629 * \param frame The frame itself
630 * \return Return frame on success, NULL on failure
632 struct ast_frame *ast_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
634 /* Pass off frame to it's respective list write function */
635 if (frame->frametype == AST_FRAME_VOICE)
636 return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
637 else if (frame->frametype == AST_FRAME_DTMF)
638 return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
639 else
640 return frame;
644 /*! \brief Wait for audiohook trigger to be triggered
645 * \param audiohook Audiohook to wait on
647 void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
649 struct timeval tv;
650 struct timespec ts;
652 tv = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
653 ts.tv_sec = tv.tv_sec;
654 ts.tv_nsec = tv.tv_usec * 1000;
656 ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
658 return;