Asterisk - The Open Source Telephony Project  GIT-master-0190e70
audiohook.c
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1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 1999 - 2007, Digium, Inc.
5  *
6  * Joshua Colp <jcolp@digium.com>
7  *
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.
13  *
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.
17  */
18 
19 /*! \file
20  *
21  * \brief Audiohooks Architecture
22  *
23  * \author Joshua Colp <jcolp@digium.com>
24  */
25 
26 /*** MODULEINFO
27  <support_level>core</support_level>
28  ***/
29 
30 #include "asterisk.h"
31 
32 #include <signal.h>
33 
34 #include "asterisk/channel.h"
35 #include "asterisk/utils.h"
36 #include "asterisk/lock.h"
37 #include "asterisk/linkedlists.h"
38 #include "asterisk/audiohook.h"
39 #include "asterisk/slinfactory.h"
40 #include "asterisk/frame.h"
41 #include "asterisk/translate.h"
42 #include "asterisk/format_cache.h"
43 
44 #define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */
45 #define AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE 100 /*!< When small queue is enabled, this is the maximum amount of audio that can remain queued at a time. */
46 #define AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE 500 /*!< Otheriwise we still don't want the queue to grow indefinitely */
47 
48 #define DEFAULT_INTERNAL_SAMPLE_RATE 8000
49 
52  struct ast_format *format;
53 };
54 
56  /* If all the audiohooks in this list are capable
57  * of processing slinear at any sample rate, this
58  * variable will be set and the sample rate will
59  * be preserved during ast_audiohook_write_list()*/
61  int list_internal_samp_rate;/*!< Internal sample rate used when writing to the audiohook list */
62 
63  struct ast_audiohook_translate in_translate[2];
64  struct ast_audiohook_translate out_translate[2];
66  AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
67  AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
68 };
69 
70 static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
71 {
72  struct ast_format *slin;
73 
74  if (audiohook->hook_internal_samp_rate == rate) {
75  return 0;
76  }
77 
78  audiohook->hook_internal_samp_rate = rate;
79 
81 
82  /* Setup the factories that are needed for this audiohook type */
83  switch (audiohook->type) {
86  if (reset) {
89  }
92  break;
93  default:
94  break;
95  }
96 
97  return 0;
98 }
99 
100 /*! \brief Initialize an audiohook structure
101  *
102  * \param audiohook Audiohook structure
103  * \param type
104  * \param source, init_flags
105  *
106  * \return Returns 0 on success, -1 on failure
107  */
108 int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags init_flags)
109 {
110  /* Need to keep the type and source */
111  audiohook->type = type;
112  audiohook->source = source;
113 
114  /* Initialize lock that protects our audiohook */
115  ast_mutex_init(&audiohook->lock);
116  ast_cond_init(&audiohook->trigger, NULL);
117 
118  audiohook->init_flags = init_flags;
119 
120  /* initialize internal rate at 8khz, this will adjust if necessary */
122 
123  /* Since we are just starting out... this audiohook is new */
125 
126  return 0;
127 }
128 
129 /*! \brief Destroys an audiohook structure
130  * \param audiohook Audiohook structure
131  * \return Returns 0 on success, -1 on failure
132  */
133 int ast_audiohook_destroy(struct ast_audiohook *audiohook)
134 {
135  /* Drop the factories used by this audiohook type */
136  switch (audiohook->type) {
141  break;
142  default:
143  break;
144  }
145 
146  /* Destroy translation path if present */
147  if (audiohook->trans_pvt)
149 
150  ao2_cleanup(audiohook->format);
151 
152  /* Lock and trigger be gone! */
153  ast_cond_destroy(&audiohook->trigger);
154  ast_mutex_destroy(&audiohook->lock);
155 
156  return 0;
157 }
158 
159 #define SHOULD_MUTE(hook, dir) \
160  ((ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ) && (dir == AST_AUDIOHOOK_DIRECTION_READ)) || \
161  (ast_test_flag(hook, AST_AUDIOHOOK_MUTE_WRITE) && (dir == AST_AUDIOHOOK_DIRECTION_WRITE)) || \
162  (ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE)))
163 
164 /*! \brief Writes a frame into the audiohook structure
165  * \param audiohook Audiohook structure
166  * \param direction Direction the audio frame came from
167  * \param frame Frame to write in
168  * \return Returns 0 on success, -1 on failure
169  */
170 int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
171 {
172  struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
173  struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
174  struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime;
175  int our_factory_samples;
176  int our_factory_ms;
177  int other_factory_samples;
178  int other_factory_ms;
179 
180  /* Update last feeding time to be current */
181  *rwtime = ast_tvnow();
182 
183  our_factory_samples = ast_slinfactory_available(factory);
184  our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / (audiohook->hook_internal_samp_rate / 1000));
185  other_factory_samples = ast_slinfactory_available(other_factory);
186  other_factory_ms = other_factory_samples / (audiohook->hook_internal_samp_rate / 1000);
187 
188  if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
189  ast_debug(1, "Flushing audiohook %p so it remains in sync\n", audiohook);
190  ast_slinfactory_flush(factory);
191  ast_slinfactory_flush(other_factory);
192  }
193 
194  if (ast_test_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE) && ((our_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE))) {
195  ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
196  ast_slinfactory_flush(factory);
197  ast_slinfactory_flush(other_factory);
198  } else if ((our_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE)) {
199  ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
200  ast_slinfactory_flush(factory);
201  ast_slinfactory_flush(other_factory);
202  }
203 
204  /* Write frame out to respective factory */
205  ast_slinfactory_feed(factory, frame);
206 
207  /* If we need to notify the respective handler of this audiohook, do so */
209  ast_cond_signal(&audiohook->trigger);
211  ast_cond_signal(&audiohook->trigger);
212  } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
213  ast_cond_signal(&audiohook->trigger);
214  }
215 
216  return 0;
217 }
218 
219 static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
220 {
221  struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
222  int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
223  short buf[samples];
224  struct ast_frame frame = {
226  .subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate),
227  .data.ptr = buf,
228  .datalen = sizeof(buf),
229  .samples = samples,
230  };
231 
232  /* Ensure the factory is able to give us the samples we want */
233  if (samples > ast_slinfactory_available(factory)) {
234  return NULL;
235  }
236 
237  /* Read data in from factory */
238  if (!ast_slinfactory_read(factory, buf, samples)) {
239  return NULL;
240  }
241 
242  if (SHOULD_MUTE(audiohook, direction)) {
243  /* Swap frame data for zeros if mute is required */
244  ast_frame_clear(&frame);
245  } else if (vol) {
246  /* If a volume adjustment needs to be applied apply it */
247  ast_frame_adjust_volume(&frame, vol);
248  }
249 
250  return ast_frdup(&frame);
251 }
252 
253 static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples, struct ast_frame **read_reference, struct ast_frame **write_reference)
254 {
255  int count;
256  int usable_read;
257  int usable_write;
258  short adjust_value;
259  short buf1[samples];
260  short buf2[samples];
261  short *read_buf = NULL;
262  short *write_buf = NULL;
263  struct ast_frame frame = {
265  .datalen = sizeof(buf1),
266  .samples = samples,
267  };
268 
269  /* Make sure both factories have the required samples */
270  usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
271  usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
272 
273  if (!usable_read && !usable_write) {
274  /* If both factories are unusable bail out */
275  ast_debug(1, "Read factory %p and write factory %p both fail to provide %zu samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
276  return NULL;
277  }
278 
279  /* If we want to provide only a read factory make sure we aren't waiting for other audio */
280  if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
281  ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
282  return NULL;
283  }
284 
285  /* If we want to provide only a write factory make sure we aren't waiting for other audio */
286  if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
287  ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
288  return NULL;
289  }
290 
291  /* Start with the read factory... if there are enough samples, read them in */
292  if (usable_read) {
293  if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
294  read_buf = buf1;
295 
296  if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ))) {
297  /* Clear the frame data if we are muting */
298  memset(buf1, 0, sizeof(buf1));
299  } else if (audiohook->options.read_volume) {
300  /* Adjust read volume if need be */
301  adjust_value = abs(audiohook->options.read_volume);
302  for (count = 0; count < samples; count++) {
303  if (audiohook->options.read_volume > 0) {
304  ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
305  } else if (audiohook->options.read_volume < 0) {
306  ast_slinear_saturated_divide(&buf1[count], &adjust_value);
307  }
308  }
309  }
310  }
311  } else {
312  ast_debug(1, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
313  }
314 
315  /* Move on to the write factory... if there are enough samples, read them in */
316  if (usable_write) {
317  if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
318  write_buf = buf2;
319 
320  if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE))) {
321  /* Clear the frame data if we are muting */
322  memset(buf2, 0, sizeof(buf2));
323  } else if (audiohook->options.write_volume) {
324  /* Adjust write volume if need be */
325  adjust_value = abs(audiohook->options.write_volume);
326  for (count = 0; count < samples; count++) {
327  if (audiohook->options.write_volume > 0) {
328  ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
329  } else if (audiohook->options.write_volume < 0) {
330  ast_slinear_saturated_divide(&buf2[count], &adjust_value);
331  }
332  }
333  }
334  }
335  } else {
336  ast_debug(1, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
337  }
338 
340 
341  /* Should we substitute silence if one side lacks audio? */
343  if (read_reference && !read_buf && write_buf) {
344  read_buf = buf1;
345  memset(buf1, 0, sizeof(buf1));
346  } else if (write_reference && read_buf && !write_buf) {
347  write_buf = buf2;
348  memset(buf2, 0, sizeof(buf2));
349  }
350  }
351 
352  /* Basically we figure out which buffer to use... and if mixing can be done here */
353  if (read_buf && read_reference) {
354  frame.data.ptr = read_buf;
355  *read_reference = ast_frdup(&frame);
356  }
357  if (write_buf && write_reference) {
358  frame.data.ptr = write_buf;
359  *write_reference = ast_frdup(&frame);
360  }
361 
362  /* Make the correct buffer part of the built frame, so it gets duplicated. */
363  if (read_buf) {
364  frame.data.ptr = read_buf;
365  if (write_buf) {
366  for (count = 0; count < samples; count++) {
367  ast_slinear_saturated_add(read_buf++, write_buf++);
368  }
369  }
370  } else if (write_buf) {
371  frame.data.ptr = write_buf;
372  } else {
373  return NULL;
374  }
375 
376  /* Yahoo, a combined copy of the audio! */
377  return ast_frdup(&frame);
378 }
379 
380 static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format, struct ast_frame **read_reference, struct ast_frame **write_reference)
381 {
382  struct ast_frame *read_frame = NULL, *final_frame = NULL;
383  struct ast_format *slin;
384 
385  /*
386  * Update the rate if compatibility mode is turned off or if it is
387  * turned on and the format rate is higher than the current rate.
388  *
389  * This makes it so any unnecessary rate switching/resetting does
390  * not take place and also any associated audiohook_list's internal
391  * sample rate maintains the highest sample rate between hooks.
392  */
393  if (!ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) ||
395  ast_format_get_sample_rate(format) > audiohook->hook_internal_samp_rate)) {
397  }
398 
399  /* If the sample rate of the requested format differs from that of the underlying audiohook
400  * sample rate determine how many samples we actually need to get from the audiohook. This
401  * needs to occur as the signed linear factory stores them at the rate of the audiohook.
402  * We do this by determining the duration of audio they've requested and then determining
403  * how many samples that would be in the audiohook format.
404  */
405  if (ast_format_get_sample_rate(format) != audiohook->hook_internal_samp_rate) {
406  samples = (audiohook->hook_internal_samp_rate / 1000) * (samples / (ast_format_get_sample_rate(format) / 1000));
407  }
408 
409  if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
410  audiohook_read_frame_both(audiohook, samples, read_reference, write_reference) :
411  audiohook_read_frame_single(audiohook, samples, direction)))) {
412  return NULL;
413  }
414 
416 
417  /* If they don't want signed linear back out, we'll have to send it through the translation path */
418  if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) {
419  /* Rebuild translation path if different format then previously */
420  if (ast_format_cmp(format, audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) {
421  if (audiohook->trans_pvt) {
423  audiohook->trans_pvt = NULL;
424  }
425 
426  /* 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 */
427  if (!(audiohook->trans_pvt = ast_translator_build_path(format, slin))) {
428  ast_frfree(read_frame);
429  return NULL;
430  }
431  ao2_replace(audiohook->format, format);
432  }
433  /* Convert to requested format, and allow the read in frame to be freed */
434  final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
435  } else {
436  final_frame = read_frame;
437  }
438 
439  return final_frame;
440 }
441 
442 /*! \brief Reads a frame in from the audiohook structure
443  * \param audiohook Audiohook structure
444  * \param samples Number of samples wanted in requested output format
445  * \param direction Direction the audio frame came from
446  * \param format Format of frame remote side wants back
447  * \return Returns frame on success, NULL on failure
448  */
449 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format)
450 {
451  return audiohook_read_frame_helper(audiohook, samples, direction, format, NULL, NULL);
452 }
453 
454 /*! \brief Reads a frame in from the audiohook structure
455  * \param audiohook Audiohook structure
456  * \param samples Number of samples wanted
457  * \param direction Direction the audio frame came from
458  * \param format Format of frame remote side wants back
459  * \param read_frame frame pointer for copying read frame data
460  * \param write_frame frame pointer for copying write frame data
461  * \return Returns frame on success, NULL on failure
462  */
463 struct ast_frame *ast_audiohook_read_frame_all(struct ast_audiohook *audiohook, size_t samples, struct ast_format *format, struct ast_frame **read_frame, struct ast_frame **write_frame)
464 {
465  return audiohook_read_frame_helper(audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, format, read_frame, write_frame);
466 }
467 
469 {
470  struct ast_audiohook *ah = NULL;
471 
472  /*
473  * Anytime the samplerate compatibility is set (attach/remove an audiohook) the
474  * list's internal sample rate needs to be reset so that the next time processing
475  * through write_list, if needed, it will get updated to the correct rate.
476  *
477  * A list's internal rate always chooses the higher between its own rate and a
478  * given rate. If the current rate is being driven by an audiohook that wanted a
479  * higher rate then when this audiohook is removed the list's rate would remain
480  * at that level when it should be lower, and with no way to lower it since any
481  * rate compared against it would be lower.
482  *
483  * By setting it back to the lowest rate it can recalulate the new highest rate.
484  */
486 
487  audiohook_list->native_slin_compatible = 1;
488  AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, ah, list) {
490  audiohook_list->native_slin_compatible = 0;
491  return;
492  }
493  }
494 }
495 
496 /*! \brief Attach audiohook to channel
497  * \param chan Channel
498  * \param audiohook Audiohook structure
499  * \return Returns 0 on success, -1 on failure
500  */
501 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
502 {
503  ast_channel_lock(chan);
504 
505  /* Don't allow an audiohook to be attached to a channel that is already hung up.
506  * The hang up process is what actually notifies the audiohook that it should
507  * stop.
508  */
510  ast_channel_unlock(chan);
511  return -1;
512  }
513 
514  if (!ast_channel_audiohooks(chan)) {
515  struct ast_audiohook_list *ahlist;
516  /* Whoops... allocate a new structure */
517  if (!(ahlist = ast_calloc(1, sizeof(*ahlist)))) {
518  ast_channel_unlock(chan);
519  return -1;
520  }
521  ast_channel_audiohooks_set(chan, ahlist);
525  /* This sample rate will adjust as necessary when writing to the list. */
527  }
528 
529  /* Drop into respective list */
530  if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
531  AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
532  } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
533  AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
534  } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
536  }
537 
538  /*
539  * Initialize the audiohook's rate to the default. If it needs to be,
540  * it will get updated later.
541  */
544 
545  /* Change status over to running since it is now attached */
547 
548  if (ast_channel_is_bridged(chan)) {
550  }
551 
552  ast_channel_unlock(chan);
553 
554  return 0;
555 }
556 
557 /*! \brief Update audiohook's status
558  * \param audiohook Audiohook structure
559  * \param status Audiohook status enum
560  *
561  * \note once status is updated to DONE, this function can not be used to set the
562  * status back to any other setting. Setting DONE effectively locks the status as such.
563  */
564 
566 {
567  ast_audiohook_lock(audiohook);
568  if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
569  audiohook->status = status;
570  ast_cond_signal(&audiohook->trigger);
571  }
572  ast_audiohook_unlock(audiohook);
573 }
574 
575 /*! \brief Detach audiohook from channel
576  * \param audiohook Audiohook structure
577  * \return Returns 0 on success, -1 on failure
578  */
579 int ast_audiohook_detach(struct ast_audiohook *audiohook)
580 {
581  if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
582  return 0;
583  }
584 
586 
587  while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
588  ast_audiohook_trigger_wait(audiohook);
589  }
590 
591  return 0;
592 }
593 
594 void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
595 {
596  int i;
597  struct ast_audiohook *audiohook;
598 
599  if (!audiohook_list) {
600  return;
601  }
602 
603  /* Drop any spies */
604  while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
606  }
607 
608  /* Drop any whispering sources */
609  while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
611  }
612 
613  /* Drop any manipulaters */
614  while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
616  audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
617  }
618 
619  /* Drop translation paths if present */
620  for (i = 0; i < 2; i++) {
621  if (audiohook_list->in_translate[i].trans_pvt) {
622  ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
623  ao2_cleanup(audiohook_list->in_translate[i].format);
624  }
625  if (audiohook_list->out_translate[i].trans_pvt) {
627  ao2_cleanup(audiohook_list->in_translate[i].format);
628  }
629  }
630 
631  /* Free ourselves */
632  ast_free(audiohook_list);
633 }
634 
635 /*! \brief find an audiohook based on its source
636  * \param audiohook_list The list of audiohooks to search in
637  * \param source The source of the audiohook we wish to find
638  * \return Return the corresponding audiohook or NULL if it cannot be found.
639  */
640 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
641 {
642  struct ast_audiohook *audiohook = NULL;
643 
644  AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
645  if (!strcasecmp(audiohook->source, source)) {
646  return audiohook;
647  }
648  }
649 
650  AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
651  if (!strcasecmp(audiohook->source, source)) {
652  return audiohook;
653  }
654  }
655 
656  AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
657  if (!strcasecmp(audiohook->source, source)) {
658  return audiohook;
659  }
660  }
661 
662  return NULL;
663 }
664 
665 static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook)
666 {
667  enum ast_audiohook_status oldstatus;
668 
669  /* By locking both channels and the audiohook, we can assure that
670  * another thread will not have a chance to read the audiohook's status
671  * as done, even though ast_audiohook_remove signals the trigger
672  * condition.
673  */
674  ast_audiohook_lock(audiohook);
675  oldstatus = audiohook->status;
676 
677  ast_audiohook_remove(old_chan, audiohook);
678  ast_audiohook_attach(new_chan, audiohook);
679 
680  audiohook->status = oldstatus;
681  ast_audiohook_unlock(audiohook);
682 }
683 
684 void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
685 {
686  struct ast_audiohook *audiohook;
687 
688  if (!ast_channel_audiohooks(old_chan)) {
689  return;
690  }
691 
692  audiohook = find_audiohook_by_source(ast_channel_audiohooks(old_chan), source);
693  if (!audiohook) {
694  return;
695  }
696 
697  audiohook_move(old_chan, new_chan, audiohook);
698 }
699 
700 void ast_audiohook_move_all(struct ast_channel *old_chan, struct ast_channel *new_chan)
701 {
702  struct ast_audiohook *audiohook;
703  struct ast_audiohook_list *audiohook_list;
704 
705  audiohook_list = ast_channel_audiohooks(old_chan);
706  if (!audiohook_list) {
707  return;
708  }
709 
710  AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
711  audiohook_move(old_chan, new_chan, audiohook);
712  }
714 
715  AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
716  audiohook_move(old_chan, new_chan, audiohook);
717  }
719 
720  AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
721  audiohook_move(old_chan, new_chan, audiohook);
722  }
724 }
725 
726 /*! \brief Detach specified source audiohook from channel
727  * \param chan Channel to detach from
728  * \param source Name of source to detach
729  * \return Returns 0 on success, -1 on failure
730  */
731 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
732 {
733  struct ast_audiohook *audiohook = NULL;
734 
735  ast_channel_lock(chan);
736 
737  /* Ensure the channel has audiohooks on it */
738  if (!ast_channel_audiohooks(chan)) {
739  ast_channel_unlock(chan);
740  return -1;
741  }
742 
743  audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
744 
745  ast_channel_unlock(chan);
746 
747  if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
749  }
750 
751  return (audiohook ? 0 : -1);
752 }
753 
754 /*!
755  * \brief Remove an audiohook from a specified channel
756  *
757  * \param chan Channel to remove from
758  * \param audiohook Audiohook to remove
759  *
760  * \return Returns 0 on success, -1 on failure
761  *
762  * \note The channel does not need to be locked before calling this function
763  */
764 int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
765 {
766  ast_channel_lock(chan);
767 
768  if (!ast_channel_audiohooks(chan)) {
769  ast_channel_unlock(chan);
770  return -1;
771  }
772 
773  if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
774  AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
775  } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
776  AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
777  } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
778  AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
779  }
780 
783 
784  if (ast_channel_is_bridged(chan)) {
786  }
787 
788  ast_channel_unlock(chan);
789 
790  return 0;
791 }
792 
793 /*! \brief Pass a DTMF frame off to be handled by the audiohook core
794  * \param chan Channel that the list is coming off of
795  * \param audiohook_list List of audiohooks
796  * \param direction Direction frame is coming in from
797  * \param frame The frame itself
798  * \return Return frame on success, NULL on failure
799  */
800 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)
801 {
802  struct ast_audiohook *audiohook = NULL;
803  int removed = 0;
804 
805  AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
806  ast_audiohook_lock(audiohook);
807  if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
809  removed = 1;
811  ast_audiohook_unlock(audiohook);
812  audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
813  if (ast_channel_is_bridged(chan)) {
815  }
816  continue;
817  }
818  if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF)) {
819  audiohook->manipulate_callback(audiohook, chan, frame, direction);
820  }
821  ast_audiohook_unlock(audiohook);
822  }
824 
825  /* if an audiohook got removed, reset samplerate compatibility */
826  if (removed) {
828  }
829  return frame;
830 }
831 
832 static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_list *audiohook_list,
833  enum ast_audiohook_direction direction, struct ast_frame *frame)
834 {
835  struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ?
836  &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
837  struct ast_frame *new_frame = frame;
838  struct ast_format *slin;
839 
840  /*
841  * If we are capable of sample rates other that 8khz, update the internal
842  * audiohook_list's rate and higher sample rate audio arrives. If native
843  * slin compatibility is turned on all audiohooks in the list will be
844  * updated as well during read/write processing.
845  */
846  audiohook_list->list_internal_samp_rate =
848 
850  if (ast_format_cmp(frame->subclass.format, slin) == AST_FORMAT_CMP_EQUAL) {
851  return new_frame;
852  }
853 
854  if (!in_translate->format ||
855  ast_format_cmp(frame->subclass.format, in_translate->format) != AST_FORMAT_CMP_EQUAL) {
856  struct ast_trans_pvt *new_trans;
857 
858  new_trans = ast_translator_build_path(slin, frame->subclass.format);
859  if (!new_trans) {
860  return NULL;
861  }
862 
863  if (in_translate->trans_pvt) {
864  ast_translator_free_path(in_translate->trans_pvt);
865  }
866  in_translate->trans_pvt = new_trans;
867 
868  ao2_replace(in_translate->format, frame->subclass.format);
869  }
870 
871  if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) {
872  return NULL;
873  }
874 
875  return new_frame;
876 }
877 
879  enum ast_audiohook_direction direction, struct ast_frame *slin_frame, struct ast_format *outformat)
880 {
881  struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
882  struct ast_frame *outframe = NULL;
883  if (ast_format_cmp(slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
884  /* rebuild translators if necessary */
885  if (ast_format_cmp(out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
886  if (out_translate->trans_pvt) {
887  ast_translator_free_path(out_translate->trans_pvt);
888  }
889  if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) {
890  return NULL;
891  }
892  ao2_replace(out_translate->format, outformat);
893  }
894  /* translate back to the format the frame came in as. */
895  if (!(outframe = ast_translate(out_translate->trans_pvt, slin_frame, 0))) {
896  return NULL;
897  }
898  }
899  return outframe;
900 }
901 
902 /*!
903  *\brief Set the audiohook's internal sample rate to the audiohook_list's rate,
904  * but only when native slin compatibility is turned on.
905  *
906  * \param audiohook_list audiohook_list data object
907  * \param audiohook the audiohook to update
908  * \param rate the current max internal sample rate
909  */
910 static void audiohook_list_set_hook_rate(struct ast_audiohook_list *audiohook_list,
911  struct ast_audiohook *audiohook, int *rate)
912 {
913  /* The rate should always be the max between itself and the hook */
914  if (audiohook->hook_internal_samp_rate > *rate) {
915  *rate = audiohook->hook_internal_samp_rate;
916  }
917 
918  /*
919  * If native slin compatibility is turned on then update the audiohook
920  * with the audiohook_list's current rate. Note, the audiohook's rate is
921  * set to the audiohook_list's rate and not the given rate. If there is
922  * a change in rate the hook's rate is changed on its next check.
923  */
924  if (audiohook_list->native_slin_compatible) {
926  audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
927  } else {
929  }
930 }
931 
932 /*!
933  * \brief Pass an AUDIO frame off to be handled by the audiohook core
934  *
935  * \details
936  * This function has 3 ast_frames and 3 parts to handle each. At the beginning of this
937  * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
938  * input frame.
939  *
940  * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
941  * format. The result of this part is middle_frame is guaranteed to be in
942  * SLINEAR format for Part_2.
943  * Part_2: Send middle_frame off to spies and manipulators. At this point middle_frame is
944  * either a new frame as result of the translation, or points directly to the start_frame
945  * because no translation to SLINEAR audio was required.
946  * Part_3: Translate end_frame's audio back into the format of start frame if necessary. This
947  * is only necessary if manipulation of middle_frame occurred.
948  *
949  * \param chan Channel that the list is coming off of
950  * \param audiohook_list List of audiohooks
951  * \param direction Direction frame is coming in from
952  * \param frame The frame itself
953  * \return Return frame on success, NULL on failure
954  */
955 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)
956 {
957  struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
958  struct ast_audiohook *audiohook = NULL;
959  int samples;
960  int middle_frame_manipulated = 0;
961  int removed = 0;
962  int internal_sample_rate;
963 
964  /* ---Part_1. translate start_frame to SLINEAR if necessary. */
965  if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
966  return frame;
967  }
968 
969  /* If the translation resulted in an interpolated frame then immediately return as audiohooks
970  * rely on actual media being present to do things.
971  */
972  if (!middle_frame->data.ptr) {
973  if (middle_frame != start_frame) {
974  ast_frfree(middle_frame);
975  }
976  return start_frame;
977  }
978 
979  samples = middle_frame->samples;
980 
981  /*
982  * While processing each audiohook check to see if the internal sample rate needs
983  * to be adjusted (it should be the highest rate specified between formats and
984  * hooks). The given audiohook_list's internal sample rate is then set to the
985  * updated value before returning.
986  *
987  * If slin compatibility mode is turned on then an audiohook's internal sample
988  * rate is set to its audiohook_list's rate. If an audiohook_list's rate is
989  * adjusted during this pass then the change is picked up by the audiohooks
990  * on the next pass.
991  */
992  internal_sample_rate = audiohook_list->list_internal_samp_rate;
993 
994  /* ---Part_2: Send middle_frame to spy and manipulator lists. middle_frame is guaranteed to be SLINEAR here.*/
995  /* Queue up signed linear frame to each spy */
996  AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
997  ast_audiohook_lock(audiohook);
998  if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
1000  removed = 1;
1002  ast_audiohook_unlock(audiohook);
1003  if (ast_channel_is_bridged(chan)) {
1005  }
1006  continue;
1007  }
1008  audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
1009  ast_audiohook_write_frame(audiohook, direction, middle_frame);
1010  ast_audiohook_unlock(audiohook);
1011  }
1013 
1014  /* If this frame is being written out to the channel then we need to use whisper sources */
1015  if (!AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
1016  int i = 0;
1017  short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
1018  memset(&combine_buf, 0, sizeof(combine_buf));
1019  AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
1020  struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
1021  ast_audiohook_lock(audiohook);
1022  if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
1024  removed = 1;
1026  ast_audiohook_unlock(audiohook);
1027  if (ast_channel_is_bridged(chan)) {
1029  }
1030  continue;
1031  }
1032  audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
1033  if (ast_slinfactory_available(factory) >= samples && ast_slinfactory_read(factory, read_buf, samples)) {
1034  /* Take audio from this whisper source and combine it into our main buffer */
1035  for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++) {
1036  ast_slinear_saturated_add(data1, data2);
1037  }
1038  }
1039  ast_audiohook_unlock(audiohook);
1040  }
1042  /* We take all of the combined whisper sources and combine them into the audio being written out */
1043  for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++) {
1044  ast_slinear_saturated_add(data1, data2);
1045  }
1046  middle_frame_manipulated = 1;
1047  }
1048 
1049  /* Pass off frame to manipulate audiohooks */
1050  if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
1051  AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
1052  ast_audiohook_lock(audiohook);
1053  if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
1055  removed = 1;
1057  ast_audiohook_unlock(audiohook);
1058  /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
1059  audiohook->manipulate_callback(audiohook, chan, NULL, direction);
1060  if (ast_channel_is_bridged(chan)) {
1062  }
1063  continue;
1064  }
1065  audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
1066  /*
1067  * Feed in frame to manipulation.
1068  */
1069  if (!audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
1070  /*
1071  * XXX FAILURES ARE IGNORED XXX
1072  * If the manipulation fails then the frame will be returned in its original state.
1073  * Since there are potentially more manipulator callbacks in the list, no action should
1074  * be taken here to exit early.
1075  */
1076  middle_frame_manipulated = 1;
1077  }
1078  ast_audiohook_unlock(audiohook);
1079  }
1081  }
1082 
1083  /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
1084  if (middle_frame_manipulated) {
1085  if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, start_frame->subclass.format))) {
1086  /* translation failed, so just pass back the input frame */
1087  end_frame = start_frame;
1088  }
1089  } else {
1090  end_frame = start_frame;
1091  }
1092  /* clean up our middle_frame if required */
1093  if (middle_frame != end_frame) {
1094  ast_frfree(middle_frame);
1095  middle_frame = NULL;
1096  }
1097 
1098  /* Before returning, if an audiohook got removed, reset samplerate compatibility */
1099  if (removed) {
1101  } else {
1102  /*
1103  * Set the audiohook_list's rate to the updated rate. Note that if a hook
1104  * was removed then the list's internal rate is reset to the default.
1105  */
1106  audiohook_list->list_internal_samp_rate = internal_sample_rate;
1107  }
1108 
1109  return end_frame;
1110 }
1111 
1113 {
1114  return !audiohook_list
1115  || (AST_LIST_EMPTY(&audiohook_list->spy_list)
1116  && AST_LIST_EMPTY(&audiohook_list->whisper_list)
1117  && AST_LIST_EMPTY(&audiohook_list->manipulate_list));
1118 }
1119 
1120 /*! \brief Pass a frame off to be handled by the audiohook core
1121  * \param chan Channel that the list is coming off of
1122  * \param audiohook_list List of audiohooks
1123  * \param direction Direction frame is coming in from
1124  * \param frame The frame itself
1125  * \return Return frame on success, NULL on failure
1126  */
1127 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)
1128 {
1129  /* Pass off frame to it's respective list write function */
1130  if (frame->frametype == AST_FRAME_VOICE) {
1131  return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
1132  } else if (frame->frametype == AST_FRAME_DTMF) {
1133  return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
1134  } else {
1135  return frame;
1136  }
1137 }
1138 
1139 /*! \brief Wait for audiohook trigger to be triggered
1140  * \param audiohook Audiohook to wait on
1141  */
1143 {
1144  struct timeval wait;
1145  struct timespec ts;
1146 
1147  wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
1148  ts.tv_sec = wait.tv_sec;
1149  ts.tv_nsec = wait.tv_usec * 1000;
1150 
1151  ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
1152 
1153  return;
1154 }
1155 
1156 /* Count number of channel audiohooks by type, regardless of type */
1158 {
1159  int count = 0;
1160  struct ast_audiohook *ah = NULL;
1161 
1162  if (!ast_channel_audiohooks(chan)) {
1163  return -1;
1164  }
1165 
1166  switch (type) {
1168  AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
1169  if (!strcmp(ah->source, source)) {
1170  count++;
1171  }
1172  }
1173  break;
1175  AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
1176  if (!strcmp(ah->source, source)) {
1177  count++;
1178  }
1179  }
1180  break;
1182  AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
1183  if (!strcmp(ah->source, source)) {
1184  count++;
1185  }
1186  }
1187  break;
1188  default:
1189  ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
1190  return -1;
1191  }
1192 
1193  return count;
1194 }
1195 
1196 /* Count number of channel audiohooks by type that are running */
1198 {
1199  int count = 0;
1200  struct ast_audiohook *ah = NULL;
1201  if (!ast_channel_audiohooks(chan))
1202  return -1;
1203 
1204  switch (type) {
1206  AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
1207  if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1208  count++;
1209  }
1210  break;
1212  AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
1213  if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1214  count++;
1215  }
1216  break;
1218  AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
1219  if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1220  count++;
1221  }
1222  break;
1223  default:
1224  ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
1225  return -1;
1226  }
1227  return count;
1228 }
1229 
1230 /*! \brief Audiohook volume adjustment structure */
1232  struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
1233  int read_adjustment; /*!< Value to adjust frames read from the channel by */
1234  int write_adjustment; /*!< Value to adjust frames written to the channel by */
1235 };
1236 
1237 /*! \brief Callback used to destroy the audiohook volume datastore
1238  * \param data Volume information structure
1239  * \return Returns nothing
1240  */
1241 static void audiohook_volume_destroy(void *data)
1242 {
1243  struct audiohook_volume *audiohook_volume = data;
1244 
1245  /* Destroy the audiohook as it is no longer in use */
1246  ast_audiohook_destroy(&audiohook_volume->audiohook);
1247 
1248  /* Finally free ourselves, we are of no more use */
1249  ast_free(audiohook_volume);
1250 
1251  return;
1252 }
1253 
1254 /*! \brief Datastore used to store audiohook volume information */
1256  .type = "Volume",
1257  .destroy = audiohook_volume_destroy,
1258 };
1259 
1260 /*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
1261  * \param audiohook Audiohook attached to the channel
1262  * \param chan Channel we are attached to
1263  * \param frame Frame of audio we want to manipulate
1264  * \param direction Direction the audio came in from
1265  * \return Returns 0 on success, -1 on failure
1266  */
1267 static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
1268 {
1269  struct ast_datastore *datastore = NULL;
1271  int *gain = NULL;
1272 
1273  /* If the audiohook is shutting down don't even bother */
1274  if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
1275  return 0;
1276  }
1277 
1278  /* Try to find the datastore containg adjustment information, if we can't just bail out */
1279  if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
1280  return 0;
1281  }
1282 
1283  audiohook_volume = datastore->data;
1284 
1285  /* Based on direction grab the appropriate adjustment value */
1286  if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
1287  gain = &audiohook_volume->read_adjustment;
1288  } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
1289  gain = &audiohook_volume->write_adjustment;
1290  }
1291 
1292  /* If an adjustment value is present modify the frame */
1293  if (gain && *gain) {
1294  ast_frame_adjust_volume(frame, *gain);
1295  }
1296 
1297  return 0;
1298 }
1299 
1300 /*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
1301  * \param chan Channel to look on
1302  * \param create Whether to create the datastore if not found
1303  * \return Returns audiohook_volume structure on success, NULL on failure
1304  */
1305 static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
1306 {
1307  struct ast_datastore *datastore = NULL;
1309 
1310  /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
1311  if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
1312  return datastore->data;
1313  }
1314 
1315  /* If we are not allowed to create a datastore or if we fail to create a datastore, bail out now as we have nothing for them */
1316  if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
1317  return NULL;
1318  }
1319 
1320  /* Create a new audiohook_volume structure to contain our adjustments and audiohook */
1321  if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
1322  ast_datastore_free(datastore);
1323  return NULL;
1324  }
1325 
1326  /* Setup our audiohook structure so we can manipulate the audio */
1329 
1330  /* Attach the audiohook_volume blob to the datastore and attach to the channel */
1331  datastore->data = audiohook_volume;
1332  ast_channel_datastore_add(chan, datastore);
1333 
1334  /* All is well... put the audiohook into motion */
1335  ast_audiohook_attach(chan, &audiohook_volume->audiohook);
1336 
1337  return audiohook_volume;
1338 }
1339 
1340 /*! \brief Adjust the volume on frames read from or written to a channel
1341  * \param chan Channel to muck with
1342  * \param direction Direction to set on
1343  * \param volume Value to adjust the volume by
1344  * \return Returns 0 on success, -1 on failure
1345  */
1347 {
1349 
1350  /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
1351  if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
1352  return -1;
1353  }
1354 
1355  /* Now based on the direction set the proper value */
1356  if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1357  audiohook_volume->read_adjustment = volume;
1358  }
1359  if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1360  audiohook_volume->write_adjustment = volume;
1361  }
1362 
1363  return 0;
1364 }
1365 
1366 /*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
1367  * \param chan Channel to retrieve volume adjustment from
1368  * \param direction Direction to retrieve
1369  * \return Returns adjustment value
1370  */
1372 {
1374  int adjustment = 0;
1375 
1376  /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
1377  if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
1378  return 0;
1379  }
1380 
1381  /* Grab the adjustment value based on direction given */
1382  if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
1383  adjustment = audiohook_volume->read_adjustment;
1384  } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
1385  adjustment = audiohook_volume->write_adjustment;
1386  }
1387 
1388  return adjustment;
1389 }
1390 
1391 /*! \brief Adjust the volume on frames read from or written to a channel
1392  * \param chan Channel to muck with
1393  * \param direction Direction to increase
1394  * \param volume Value to adjust the adjustment by
1395  * \return Returns 0 on success, -1 on failure
1396  */
1398 {
1400 
1401  /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
1402  if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
1403  return -1;
1404  }
1405 
1406  /* Based on the direction change the specific adjustment value */
1407  if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1408  audiohook_volume->read_adjustment += volume;
1409  }
1410  if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1411  audiohook_volume->write_adjustment += volume;
1412  }
1413 
1414  return 0;
1415 }
1416 
1417 /*! \brief Mute frames read from or written to a channel
1418  * \param chan Channel to muck with
1419  * \param source Type of audiohook
1420  * \param flag which flag to set / clear
1421  * \param clear set or clear
1422  * \return Returns 0 on success, -1 on failure
1423  */
1424 int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
1425 {
1426  struct ast_audiohook *audiohook = NULL;
1427 
1428  ast_channel_lock(chan);
1429 
1430  /* Ensure the channel has audiohooks on it */
1431  if (!ast_channel_audiohooks(chan)) {
1432  ast_channel_unlock(chan);
1433  return -1;
1434  }
1435 
1436  audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
1437 
1438  if (audiohook) {
1439  if (clear) {
1440  ast_clear_flag(audiohook, flag);
1441  } else {
1442  ast_set_flag(audiohook, flag);
1443  }
1444  }
1445 
1446  ast_channel_unlock(chan);
1447 
1448  return (audiohook ? 0 : -1);
1449 }
struct ast_audiohook audiohook
Definition: audiohook.c:1232
const char * type
Definition: datastore.h:32
static const char type[]
Definition: chan_ooh323.c:109
Audiohook volume adjustment structure.
Definition: audiohook.c:1231
#define ast_channel_lock(chan)
Definition: channel.h:2890
Main Channel structure associated with a channel.
#define ast_frdup(fr)
Copies a frame.
struct ast_slinfactory write_factory
Definition: audiohook.h:112
ast_audiohook_flags
Definition: audiohook.h:54
Asterisk locking-related definitions:
Asterisk main include file. File version handling, generic pbx functions.
int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
Adjust the volume on frames read from or written to a channel.
Definition: audiohook.c:1346
static void audiohook_volume_destroy(void *data)
Callback used to destroy the audiohook volume datastore.
Definition: audiohook.c:1241
enum ast_audiohook_type type
Definition: audiohook.h:106
struct ast_format * format
Definition: audiohook.c:52
void ast_slinfactory_flush(struct ast_slinfactory *sf)
Flush the contents of a slinfactory.
Definition: slinfactory.c:204
int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags init_flags)
Initialize an audiohook structure.
Definition: audiohook.c:108
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)
Pass an AUDIO frame off to be handled by the audiohook core.
Definition: audiohook.c:955
int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
Attach audiohook to channel.
Definition: audiohook.c:501
int list_internal_samp_rate
Definition: audiohook.c:61
#define ast_test_flag(p, flag)
Definition: utils.h:63
ast_audiohook_init_flags
Definition: audiohook.h:71
Support for translation of data formats. translate.c.
int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
Retrieve the volume adjustment value on frames read from or written to a channel. ...
Definition: audiohook.c:1371
void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
Update audiohook&#39;s status.
Definition: audiohook.c:565
char buf[BUFSIZE]
Definition: eagi_proxy.c:66
#define ast_set_flag(p, flag)
Definition: utils.h:70
Audiohooks Architecture.
union ast_frame::@257 data
struct ast_audiohook_translate out_translate[2]
Definition: audiohook.c:64
int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
Writes a frame into the audiohook structure.
Definition: audiohook.c:170
int ast_slinfactory_feed(struct ast_slinfactory *sf, struct ast_frame *f)
Feed audio into a slinfactory.
Definition: slinfactory.c:77
Structure for a data store type.
Definition: datastore.h:31
static struct audiohook_volume * audiohook_volume_get(struct ast_channel *chan, int create)
Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a chann...
Definition: audiohook.c:1305
void ast_channel_audiohooks_set(struct ast_channel *chan, struct ast_audiohook_list *value)
Definition of a media format.
Definition: format.c:43
struct ast_audiohook_list::@338 manipulate_list
#define ast_cond_init(cond, attr)
Definition: lock.h:199
static struct ast_threadstorage buf2
struct timeval ast_tvnow(void)
Returns current timeval. Meant to replace calls to gettimeofday().
Definition: time.h:150
ast_mutex_t lock
Definition: audiohook.h:104
#define AST_LIST_EMPTY(head)
Checks whether the specified list contains any entries.
Definition: linkedlists.h:449
struct ast_trans_pvt * trans_pvt
Definition: audiohook.c:51
struct ast_frame * ast_translate(struct ast_trans_pvt *tr, struct ast_frame *f, int consume)
translates one or more frames Apply an input frame into the translator and receive zero or one output...
Definition: translate.c:559
static const struct ast_datastore_info audiohook_volume_datastore
Datastore used to store audiohook volume information.
Definition: audiohook.c:1255
int64_t ast_tvdiff_ms(struct timeval end, struct timeval start)
Computes the difference (in milliseconds) between two struct timeval instances.
Definition: time.h:98
Structure for a data store object.
Definition: datastore.h:68
struct ast_datastore * ast_channel_datastore_find(struct ast_channel *chan, const struct ast_datastore_info *info, const char *uid)
Find a datastore on a channel.
Definition: channel.c:2390
int native_slin_compatible
Definition: audiohook.c:60
static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
Definition: audiohook.c:70
#define NULL
Definition: resample.c:96
#define AST_LIST_REMOVE(head, elm, field)
Removes a specific entry from a list.
Definition: linkedlists.h:855
#define AST_FRAME_DTMF
#define AST_LIST_TRAVERSE_SAFE_END
Closes a safe loop traversal block.
Definition: linkedlists.h:614
#define ast_cond_signal(cond)
Definition: lock.h:201
struct ast_format * format
Definition: audiohook.h:115
unsigned int ast_slinfactory_available(const struct ast_slinfactory *sf)
Retrieve number of samples currently in a slinfactory.
Definition: slinfactory.c:199
int ast_datastore_free(struct ast_datastore *datastore)
Free a data store object.
Definition: datastore.c:68
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)
Pass a DTMF frame off to be handled by the audiohook core.
Definition: audiohook.c:800
struct ast_frame_subclass subclass
void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
Detach audiohooks from list and destroy said list.
Definition: audiohook.c:594
Utility functions.
int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
Find out how many audiohooks from a certain source exist on a given channel, regardless of status...
Definition: audiohook.c:1157
#define AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE
Definition: audiohook.c:46
int ast_audiohook_destroy(struct ast_audiohook *audiohook)
Destroys an audiohook structure.
Definition: audiohook.c:133
#define AST_AUDIOHOOK_SYNC_TOLERANCE
Definition: audiohook.c:44
ast_audiohook_manipulate_callback manipulate_callback
Definition: audiohook.h:117
#define SHOULD_MUTE(hook, dir)
Definition: audiohook.c:159
#define ast_audiohook_unlock(ah)
Unlock an audiohook.
Definition: audiohook.h:300
static struct ast_frame * audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples, struct ast_frame **read_reference, struct ast_frame **write_reference)
Definition: audiohook.c:253
#define ast_debug(level,...)
Log a DEBUG message.
Definition: logger.h:444
void write_buf(int file, char *buffer, int num)
Definition: eagi_proxy.c:312
#define MAX(a, b)
Definition: utils.h:228
static struct ast_audiohook * find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
find an audiohook based on its source
Definition: audiohook.c:640
enum ast_format_cmp_res ast_format_cmp(const struct ast_format *format1, const struct ast_format *format2)
Compare two formats.
Definition: format.c:201
struct ast_trans_pvt * ast_translator_build_path(struct ast_format *dest, struct ast_format *source)
Builds a translator path Build a path (possibly NULL) from source to dest.
Definition: translate.c:479
General Asterisk PBX channel definitions.
int ast_audiohook_detach(struct ast_audiohook *audiohook)
Detach audiohook from channel.
Definition: audiohook.c:579
void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
Move an audiohook from one channel to a new one.
Definition: audiohook.c:684
int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
Mute frames read from or written to a channel.
Definition: audiohook.c:1424
enum ast_audiohook_init_flags init_flags
Definition: audiohook.h:108
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)
Pass a frame off to be handled by the audiohook core.
Definition: audiohook.c:1127
ast_audiohook_type
Definition: audiohook.h:35
#define AST_LIST_REMOVE_CURRENT(field)
Removes the current entry from a list during a traversal.
Definition: linkedlists.h:556
Asterisk internal frame definitions.
static force_inline void ast_slinear_saturated_add(short *input, short *value)
Definition: utils.h:365
struct timeval ast_samp2tv(unsigned int _nsamp, unsigned int _rate)
Returns a timeval corresponding to the duration of n samples at rate r. Useful to convert samples to ...
Definition: time.h:238
static force_inline void ast_slinear_saturated_multiply(short *input, short *value)
Definition: utils.h:391
A set of macros to manage forward-linked lists.
struct ast_trans_pvt * trans_pvt
Definition: audiohook.h:116
void ast_channel_set_unbridged_nolock(struct ast_channel *chan, int value)
Variant of ast_channel_set_unbridged. Use this if the channel is already locked prior to calling...
#define AST_LIST_REMOVE_HEAD(head, field)
Removes and returns the head entry from a list.
Definition: linkedlists.h:832
AST_LIST_HEAD_NOLOCK(contactliststruct, contact)
void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
Wait for audiohook trigger to be triggered.
Definition: audiohook.c:1142
void ast_slinfactory_destroy(struct ast_slinfactory *sf)
Destroy the contents of a slinfactory.
Definition: slinfactory.c:58
unsigned int hook_internal_samp_rate
Definition: audiohook.h:119
int ast_slinfactory_init_with_format(struct ast_slinfactory *sf, struct ast_format *slin_out)
Initialize a slinfactory.
Definition: slinfactory.c:46
int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
Determine if a audiohook_list is empty or not.
Definition: audiohook.c:1112
#define AST_LIST_INSERT_TAIL(head, elm, field)
Appends a list entry to the tail of a list.
Definition: linkedlists.h:730
Default structure for translators, with the basic fields and buffers, all allocated as part of the sa...
Definition: translate.h:213
static void audiohook_list_set_hook_rate(struct ast_audiohook_list *audiohook_list, struct ast_audiohook *audiohook, int *rate)
Set the audiohook&#39;s internal sample rate to the audiohook_list&#39;s rate, but only when native slin comp...
Definition: audiohook.c:910
struct timeval ast_tvadd(struct timeval a, struct timeval b)
Returns the sum of two timevals a + b.
Definition: extconf.c:2283
#define ast_cond_destroy(cond)
Definition: lock.h:200
struct ast_frame * ast_audiohook_read_frame_all(struct ast_audiohook *audiohook, size_t samples, struct ast_format *format, struct ast_frame **read_frame, struct ast_frame **write_frame)
Reads a frame in from the audiohook structure.
Definition: audiohook.c:463
#define AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE
Definition: audiohook.c:45
struct ast_audiohook_list::@337 whisper_list
int ast_channel_is_bridged(const struct ast_channel *chan)
Determine if a channel is in a bridge.
Definition: channel.c:10603
#define AST_LIST_TRAVERSE(head, var, field)
Loops over (traverses) the entries in a list.
Definition: linkedlists.h:490
static struct ast_threadstorage buf1
long int flag
Definition: f2c.h:83
#define ast_channel_unlock(chan)
Definition: channel.h:2891
const char * source
Definition: audiohook.h:109
A machine to gather up arbitrary frames and convert them to raw slinear on demand.
#define ast_free(a)
Definition: astmm.h:182
#define ast_calloc(num, len)
A wrapper for calloc()
Definition: astmm.h:204
static struct ast_frame * read_frame(struct ast_filestream *s, int *whennext)
Definition: file.c:874
static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook)
Definition: audiohook.c:665
static force_inline void ast_slinear_saturated_divide(short *input, short *value)
Definition: utils.h:417
struct ast_audiohook_list::@336 spy_list
#define ast_clear_flag(p, flag)
Definition: utils.h:77
ast_cond_t trigger
Definition: audiohook.h:105
ast_audiohook_direction
Definition: audiohook.h:48
int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
Detach specified source audiohook from channel.
Definition: audiohook.c:731
struct ast_slinfactory read_factory
Definition: audiohook.h:111
struct ast_audiohook_translate in_translate[2]
Definition: audiohook.c:63
void * data
Definition: datastore.h:70
struct ast_audiohook_options options
Definition: audiohook.h:118
#define AST_LIST_HEAD_INIT_NOLOCK(head)
Initializes a list head structure.
Definition: linkedlists.h:680
int ast_frame_clear(struct ast_frame *frame)
Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR.
Definition: main/frame.c:850
#define ao2_replace(dst, src)
Definition: astobj2.h:517
#define abs(x)
Definition: f2c.h:195
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
struct timeval write_time
Definition: audiohook.h:114
int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
Find out how many spies of a certain type exist on a given channel, and are in state running...
Definition: audiohook.c:1197
#define ast_frfree(fr)
unsigned int ast_format_get_sample_rate(const struct ast_format *format)
Get the sample rate of a media format.
Definition: format.c:379
Data structure associated with a single frame of data.
int ast_slinfactory_read(struct ast_slinfactory *sf, short *buf, size_t samples)
Read samples from a slinfactory.
Definition: slinfactory.c:145
int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
Adjust the volume on frames read from or written to a channel.
Definition: audiohook.c:1397
static struct ast_frame * audiohook_read_frame_helper(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format, struct ast_frame **read_reference, struct ast_frame **write_reference)
Definition: audiohook.c:380
enum ast_audiohook_status status
Definition: audiohook.h:107
#define ast_datastore_alloc(info, uid)
Definition: datastore.h:89
static struct ast_frame * audiohook_list_translate_to_native(struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *slin_frame, struct ast_format *outformat)
Definition: audiohook.c:878
#define AST_LIST_TRAVERSE_SAFE_BEGIN(head, var, field)
Loops safely over (traverses) the entries in a list.
Definition: linkedlists.h:528
enum ast_frame_type frametype
int ast_frame_adjust_volume(struct ast_frame *f, int adjustment)
Adjusts the volume of the audio samples contained in a frame.
Definition: main/frame.c:778
#define ast_mutex_init(pmutex)
Definition: lock.h:184
struct ast_flags * ast_channel_flags(struct ast_channel *chan)
#define ast_mutex_destroy(a)
Definition: lock.h:186
void ast_audiohook_move_all(struct ast_channel *old_chan, struct ast_channel *new_chan)
Move all audiohooks from one channel to another.
Definition: audiohook.c:700
struct ast_format * format
struct ast_frame * ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format)
Reads a frame in from the audiohook structure.
Definition: audiohook.c:449
struct ast_audiohook_list * ast_channel_audiohooks(const struct ast_channel *chan)
struct ast_audiohook::@219 list
static struct ast_frame * audiohook_list_translate_to_slin(struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
Definition: audiohook.c:832
struct timeval read_time
Definition: audiohook.h:113
ast_audiohook_status
Definition: audiohook.h:41
#define ast_audiohook_lock(ah)
Lock an audiohook.
Definition: audiohook.h:295
int ast_channel_datastore_add(struct ast_channel *chan, struct ast_datastore *datastore)
Add a datastore to a channel.
Definition: channel.c:2376
static struct ast_frame * audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
Definition: audiohook.c:219
static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list)
Definition: audiohook.c:468
#define ast_cond_timedwait(cond, mutex, time)
Definition: lock.h:204
int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
Remove an audiohook from a specified channel.
Definition: audiohook.c:764
#define DEFAULT_INTERNAL_SAMPLE_RATE
Definition: audiohook.c:48
struct ast_format * ast_format_cache_get_slin_by_rate(unsigned int rate)
Retrieve the best signed linear format given a sample rate.
Definition: format_cache.c:520
void ast_translator_free_path(struct ast_trans_pvt *tr)
Frees a translator path Frees the given translator path structure.
Definition: translate.c:469
jack_status_t status
Definition: app_jack.c:146
static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
Helper function which actually gets called by audiohooks to perform the adjustment.
Definition: audiohook.c:1267
Media Format Cache API.