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.
21 * \brief Audiohooks Architecture
23 * \author Joshua Colp <jcolp@digium.com>
27 <support_level>core</support_level>
32 ASTERISK_REGISTER_FILE()
36 #include "asterisk/channel.h"
37 #include "asterisk/utils.h"
38 #include "asterisk/lock.h"
39 #include "asterisk/linkedlists.h"
40 #include "asterisk/audiohook.h"
41 #include "asterisk/slinfactory.h"
42 #include "asterisk/frame.h"
43 #include "asterisk/translate.h"
44 #include "asterisk/format_cache.h"
46 #define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */
47 #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. */
49 struct ast_audiohook_translate {
50 struct ast_trans_pvt *trans_pvt;
51 struct ast_format *format;
54 struct ast_audiohook_list {
55 /* If all the audiohooks in this list are capable
56 * of processing slinear at any sample rate, this
57 * variable will be set and the sample rate will
58 * be preserved during ast_audiohook_write_list()*/
59 int native_slin_compatible;
60 int list_internal_samp_rate;/*!< Internal sample rate used when writing to the audiohook list */
62 struct ast_audiohook_translate in_translate[2];
63 struct ast_audiohook_translate out_translate[2];
64 AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
65 AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
66 AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
69 static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
71 struct ast_format *slin;
73 if (audiohook->hook_internal_samp_rate == rate) {
77 audiohook->hook_internal_samp_rate = rate;
79 slin = ast_format_cache_get_slin_by_rate(rate);
81 /* Setup the factories that are needed for this audiohook type */
82 switch (audiohook->type) {
83 case AST_AUDIOHOOK_TYPE_SPY:
84 case AST_AUDIOHOOK_TYPE_WHISPER:
86 ast_slinfactory_destroy(&audiohook->read_factory);
87 ast_slinfactory_destroy(&audiohook->write_factory);
89 ast_slinfactory_init_with_format(&audiohook->read_factory, slin);
90 ast_slinfactory_init_with_format(&audiohook->write_factory, slin);
99 /*! \brief Initialize an audiohook structure
101 * \param audiohook Audiohook structure
103 * \param source, init_flags
105 * \return Returns 0 on success, -1 on failure
107 int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags init_flags)
109 /* Need to keep the type and source */
110 audiohook->type = type;
111 audiohook->source = source;
113 /* Initialize lock that protects our audiohook */
114 ast_mutex_init(&audiohook->lock);
115 ast_cond_init(&audiohook->trigger, NULL);
117 audiohook->init_flags = init_flags;
119 /* initialize internal rate at 8khz, this will adjust if necessary */
120 audiohook_set_internal_rate(audiohook, 8000, 0);
122 /* Since we are just starting out... this audiohook is new */
123 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
128 /*! \brief Destroys an audiohook structure
129 * \param audiohook Audiohook structure
130 * \return Returns 0 on success, -1 on failure
132 int ast_audiohook_destroy(struct ast_audiohook *audiohook)
134 /* Drop the factories used by this audiohook type */
135 switch (audiohook->type) {
136 case AST_AUDIOHOOK_TYPE_SPY:
137 case AST_AUDIOHOOK_TYPE_WHISPER:
138 ast_slinfactory_destroy(&audiohook->read_factory);
139 ast_slinfactory_destroy(&audiohook->write_factory);
145 /* Destroy translation path if present */
146 if (audiohook->trans_pvt)
147 ast_translator_free_path(audiohook->trans_pvt);
149 ao2_cleanup(audiohook->format);
151 /* Lock and trigger be gone! */
152 ast_cond_destroy(&audiohook->trigger);
153 ast_mutex_destroy(&audiohook->lock);
158 /*! \brief Writes a frame into the audiohook structure
159 * \param audiohook Audiohook structure
160 * \param direction Direction the audio frame came from
161 * \param frame Frame to write in
162 * \return Returns 0 on success, -1 on failure
164 int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
166 struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
167 struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
168 struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime;
169 int our_factory_samples;
171 int other_factory_samples;
172 int other_factory_ms;
175 /* Update last feeding time to be current */
176 *rwtime = ast_tvnow();
178 our_factory_samples = ast_slinfactory_available(factory);
179 our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / (audiohook->hook_internal_samp_rate / 1000));
180 other_factory_samples = ast_slinfactory_available(other_factory);
181 other_factory_ms = other_factory_samples / (audiohook->hook_internal_samp_rate / 1000);
183 if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && other_factory_samples && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
184 ast_debug(1, "Flushing audiohook %p so it remains in sync\n", audiohook);
185 ast_slinfactory_flush(factory);
186 ast_slinfactory_flush(other_factory);
189 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))) {
190 ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
191 ast_slinfactory_flush(factory);
192 ast_slinfactory_flush(other_factory);
195 /* swap frame data for zeros if mute is required */
196 if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) ||
197 (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) ||
198 (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE))) {
202 if (muteme && frame->datalen > 0) {
203 ast_frame_clear(frame);
206 /* Write frame out to respective factory */
207 ast_slinfactory_feed(factory, frame);
209 /* If we need to notify the respective handler of this audiohook, do so */
210 if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
211 ast_cond_signal(&audiohook->trigger);
212 } else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
213 ast_cond_signal(&audiohook->trigger);
214 } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
215 ast_cond_signal(&audiohook->trigger);
221 static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
223 struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
224 int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
226 struct ast_frame frame = {
227 .frametype = AST_FRAME_VOICE,
228 .subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate),
230 .datalen = sizeof(buf),
234 /* Ensure the factory is able to give us the samples we want */
235 if (samples > ast_slinfactory_available(factory)) {
239 /* Read data in from factory */
240 if (!ast_slinfactory_read(factory, buf, samples)) {
244 /* If a volume adjustment needs to be applied apply it */
246 ast_frame_adjust_volume(&frame, vol);
249 return ast_frdup(&frame);
252 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 int i = 0, usable_read, usable_write;
255 short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
256 struct ast_frame frame = {
257 .frametype = AST_FRAME_VOICE,
259 .datalen = sizeof(buf1),
263 /* Make sure both factories have the required samples */
264 usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
265 usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
267 if (!usable_read && !usable_write) {
268 /* If both factories are unusable bail out */
269 ast_debug(1, "Read factory %p and write factory %p both fail to provide %zu samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
273 /* If we want to provide only a read factory make sure we aren't waiting for other audio */
274 if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
275 ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
279 /* If we want to provide only a write factory make sure we aren't waiting for other audio */
280 if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
281 ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
285 /* Start with the read factory... if there are enough samples, read them in */
287 if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
289 /* Adjust read volume if need be */
290 if (audiohook->options.read_volume) {
292 short adjust_value = abs(audiohook->options.read_volume);
293 for (count = 0; count < samples; count++) {
294 if (audiohook->options.read_volume > 0) {
295 ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
296 } else if (audiohook->options.read_volume < 0) {
297 ast_slinear_saturated_divide(&buf1[count], &adjust_value);
303 ast_debug(1, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
306 /* Move on to the write factory... if there are enough samples, read them in */
308 if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
310 /* Adjust write volume if need be */
311 if (audiohook->options.write_volume) {
313 short adjust_value = abs(audiohook->options.write_volume);
314 for (count = 0; count < samples; count++) {
315 if (audiohook->options.write_volume > 0) {
316 ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
317 } else if (audiohook->options.write_volume < 0) {
318 ast_slinear_saturated_divide(&buf2[count], &adjust_value);
324 ast_debug(1, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
327 /* Basically we figure out which buffer to use... and if mixing can be done here */
328 if (read_buf && read_reference) {
329 frame.data.ptr = buf1;
330 *read_reference = ast_frdup(&frame);
332 if (write_buf && write_reference) {
333 frame.data.ptr = buf2;
334 *write_reference = ast_frdup(&frame);
337 if (read_buf && write_buf) {
338 for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++) {
339 ast_slinear_saturated_add(data1, data2);
342 } else if (read_buf) {
344 } else if (write_buf) {
350 /* Make the final buffer part of the frame, so it gets duplicated fine */
351 frame.data.ptr = final_buf;
353 frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
355 /* Yahoo, a combined copy of the audio! */
356 return ast_frdup(&frame);
359 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)
361 struct ast_frame *read_frame = NULL, *final_frame = NULL;
362 struct ast_format *slin;
364 audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1);
366 if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
367 audiohook_read_frame_both(audiohook, samples, read_reference, write_reference) :
368 audiohook_read_frame_single(audiohook, samples, direction)))) {
372 slin = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
374 /* If they don't want signed linear back out, we'll have to send it through the translation path */
375 if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) {
376 /* Rebuild translation path if different format then previously */
377 if (ast_format_cmp(format, audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) {
378 if (audiohook->trans_pvt) {
379 ast_translator_free_path(audiohook->trans_pvt);
380 audiohook->trans_pvt = NULL;
383 /* 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 */
384 if (!(audiohook->trans_pvt = ast_translator_build_path(format, slin))) {
385 ast_frfree(read_frame);
388 ao2_replace(audiohook->format, format);
390 /* Convert to requested format, and allow the read in frame to be freed */
391 final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
393 final_frame = read_frame;
399 /*! \brief Reads a frame in from the audiohook structure
400 * \param audiohook Audiohook structure
401 * \param samples Number of samples wanted in requested output format
402 * \param direction Direction the audio frame came from
403 * \param format Format of frame remote side wants back
404 * \return Returns frame on success, NULL on failure
406 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format)
408 return audiohook_read_frame_helper(audiohook, samples, direction, format, NULL, NULL);
411 /*! \brief Reads a frame in from the audiohook structure
412 * \param audiohook Audiohook structure
413 * \param samples Number of samples wanted
414 * \param direction Direction the audio frame came from
415 * \param format Format of frame remote side wants back
416 * \param read_frame frame pointer for copying read frame data
417 * \param write_frame frame pointer for copying write frame data
418 * \return Returns frame on success, NULL on failure
420 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)
422 return audiohook_read_frame_helper(audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, format, read_frame, write_frame);
425 static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list)
427 struct ast_audiohook *ah = NULL;
428 audiohook_list->native_slin_compatible = 1;
429 AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, ah, list) {
430 if (!(ah->init_flags & AST_AUDIOHOOK_MANIPULATE_ALL_RATES)) {
431 audiohook_list->native_slin_compatible = 0;
437 /*! \brief Attach audiohook to channel
438 * \param chan Channel
439 * \param audiohook Audiohook structure
440 * \return Returns 0 on success, -1 on failure
442 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
444 ast_channel_lock(chan);
446 if (!ast_channel_audiohooks(chan)) {
447 struct ast_audiohook_list *ahlist;
448 /* Whoops... allocate a new structure */
449 if (!(ahlist = ast_calloc(1, sizeof(*ahlist)))) {
450 ast_channel_unlock(chan);
453 ast_channel_audiohooks_set(chan, ahlist);
454 AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->spy_list);
455 AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->whisper_list);
456 AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->manipulate_list);
457 /* This sample rate will adjust as necessary when writing to the list. */
458 ast_channel_audiohooks(chan)->list_internal_samp_rate = 8000;
461 /* Drop into respective list */
462 if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
463 AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
464 } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
465 AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
466 } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
467 AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
471 audiohook_set_internal_rate(audiohook, ast_channel_audiohooks(chan)->list_internal_samp_rate, 1);
472 audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
474 /* Change status over to running since it is now attached */
475 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
477 if (ast_channel_is_bridged(chan)) {
478 ast_channel_set_unbridged_nolock(chan, 1);
481 ast_channel_unlock(chan);
486 /*! \brief Update audiohook's status
487 * \param audiohook Audiohook structure
488 * \param status Audiohook status enum
490 * \note once status is updated to DONE, this function can not be used to set the
491 * status back to any other setting. Setting DONE effectively locks the status as such.
494 void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
496 ast_audiohook_lock(audiohook);
497 if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
498 audiohook->status = status;
499 ast_cond_signal(&audiohook->trigger);
501 ast_audiohook_unlock(audiohook);
504 /*! \brief Detach audiohook from channel
505 * \param audiohook Audiohook structure
506 * \return Returns 0 on success, -1 on failure
508 int ast_audiohook_detach(struct ast_audiohook *audiohook)
510 if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
514 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
516 while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
517 ast_audiohook_trigger_wait(audiohook);
523 void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
526 struct ast_audiohook *audiohook;
528 if (!audiohook_list) {
533 while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
534 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
537 /* Drop any whispering sources */
538 while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
539 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
542 /* Drop any manipulaters */
543 while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
544 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
545 audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
548 /* Drop translation paths if present */
549 for (i = 0; i < 2; i++) {
550 if (audiohook_list->in_translate[i].trans_pvt) {
551 ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
552 ao2_cleanup(audiohook_list->in_translate[i].format);
554 if (audiohook_list->out_translate[i].trans_pvt) {
555 ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
556 ao2_cleanup(audiohook_list->in_translate[i].format);
561 ast_free(audiohook_list);
564 /*! \brief find an audiohook based on its source
565 * \param audiohook_list The list of audiohooks to search in
566 * \param source The source of the audiohook we wish to find
567 * \return Return the corresponding audiohook or NULL if it cannot be found.
569 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
571 struct ast_audiohook *audiohook = NULL;
573 AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
574 if (!strcasecmp(audiohook->source, source)) {
579 AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
580 if (!strcasecmp(audiohook->source, source)) {
585 AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
586 if (!strcasecmp(audiohook->source, source)) {
594 static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook)
596 enum ast_audiohook_status oldstatus;
598 /* By locking both channels and the audiohook, we can assure that
599 * another thread will not have a chance to read the audiohook's status
600 * as done, even though ast_audiohook_remove signals the trigger
603 ast_audiohook_lock(audiohook);
604 oldstatus = audiohook->status;
606 ast_audiohook_remove(old_chan, audiohook);
607 ast_audiohook_attach(new_chan, audiohook);
609 audiohook->status = oldstatus;
610 ast_audiohook_unlock(audiohook);
613 void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
615 struct ast_audiohook *audiohook;
617 if (!ast_channel_audiohooks(old_chan)) {
621 audiohook = find_audiohook_by_source(ast_channel_audiohooks(old_chan), source);
626 audiohook_move(old_chan, new_chan, audiohook);
629 void ast_audiohook_move_all(struct ast_channel *old_chan, struct ast_channel *new_chan)
631 struct ast_audiohook *audiohook;
632 struct ast_audiohook_list *audiohook_list;
634 audiohook_list = ast_channel_audiohooks(old_chan);
635 if (!audiohook_list) {
639 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
640 audiohook_move(old_chan, new_chan, audiohook);
642 AST_LIST_TRAVERSE_SAFE_END;
644 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
645 audiohook_move(old_chan, new_chan, audiohook);
647 AST_LIST_TRAVERSE_SAFE_END;
649 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
650 audiohook_move(old_chan, new_chan, audiohook);
652 AST_LIST_TRAVERSE_SAFE_END;
655 /*! \brief Detach specified source audiohook from channel
656 * \param chan Channel to detach from
657 * \param source Name of source to detach
658 * \return Returns 0 on success, -1 on failure
660 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
662 struct ast_audiohook *audiohook = NULL;
664 ast_channel_lock(chan);
666 /* Ensure the channel has audiohooks on it */
667 if (!ast_channel_audiohooks(chan)) {
668 ast_channel_unlock(chan);
672 audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
674 ast_channel_unlock(chan);
676 if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
677 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
680 return (audiohook ? 0 : -1);
684 * \brief Remove an audiohook from a specified channel
686 * \param chan Channel to remove from
687 * \param audiohook Audiohook to remove
689 * \return Returns 0 on success, -1 on failure
691 * \note The channel does not need to be locked before calling this function
693 int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
695 ast_channel_lock(chan);
697 if (!ast_channel_audiohooks(chan)) {
698 ast_channel_unlock(chan);
702 if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
703 AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
704 } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
705 AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
706 } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
707 AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
710 audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
711 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
713 if (ast_channel_is_bridged(chan)) {
714 ast_channel_set_unbridged_nolock(chan, 1);
717 ast_channel_unlock(chan);
722 /*! \brief Pass a DTMF frame off to be handled by the audiohook core
723 * \param chan Channel that the list is coming off of
724 * \param audiohook_list List of audiohooks
725 * \param direction Direction frame is coming in from
726 * \param frame The frame itself
727 * \return Return frame on success, NULL on failure
729 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)
731 struct ast_audiohook *audiohook = NULL;
734 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
735 ast_audiohook_lock(audiohook);
736 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
737 AST_LIST_REMOVE_CURRENT(list);
739 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
740 ast_audiohook_unlock(audiohook);
741 audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
742 if (ast_channel_is_bridged(chan)) {
743 ast_channel_set_unbridged_nolock(chan, 1);
747 if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF)) {
748 audiohook->manipulate_callback(audiohook, chan, frame, direction);
750 ast_audiohook_unlock(audiohook);
752 AST_LIST_TRAVERSE_SAFE_END;
754 /* if an audiohook got removed, reset samplerate compatibility */
756 audiohook_list_set_samplerate_compatibility(audiohook_list);
761 static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_list *audiohook_list,
762 enum ast_audiohook_direction direction, struct ast_frame *frame)
764 struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ?
765 &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
766 struct ast_frame *new_frame = frame;
767 struct ast_format *slin;
769 /* If we are capable of maintaining doing samplerates other that 8khz, update
770 * the internal audiohook_list's rate and higher samplerate audio arrives. By
771 * updating the list's rate, all the audiohooks in the list will be updated as well
772 * as the are written and read from. */
773 if (audiohook_list->native_slin_compatible) {
774 audiohook_list->list_internal_samp_rate =
775 MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate);
778 slin = ast_format_cache_get_slin_by_rate(audiohook_list->list_internal_samp_rate);
779 if (ast_format_cmp(frame->subclass.format, slin) == AST_FORMAT_CMP_EQUAL) {
783 if (ast_format_cmp(frame->subclass.format, in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) {
784 if (in_translate->trans_pvt) {
785 ast_translator_free_path(in_translate->trans_pvt);
787 if (!(in_translate->trans_pvt = ast_translator_build_path(slin, frame->subclass.format))) {
790 ao2_replace(in_translate->format, frame->subclass.format);
793 if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) {
800 static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook_list *audiohook_list,
801 enum ast_audiohook_direction direction, struct ast_frame *slin_frame, struct ast_format *outformat)
803 struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
804 struct ast_frame *outframe = NULL;
805 if (ast_format_cmp(slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
806 /* rebuild translators if necessary */
807 if (ast_format_cmp(out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
808 if (out_translate->trans_pvt) {
809 ast_translator_free_path(out_translate->trans_pvt);
811 if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) {
814 ao2_replace(out_translate->format, outformat);
816 /* translate back to the format the frame came in as. */
817 if (!(outframe = ast_translate(out_translate->trans_pvt, slin_frame, 0))) {
825 * \brief Pass an AUDIO frame off to be handled by the audiohook core
828 * This function has 3 ast_frames and 3 parts to handle each. At the beginning of this
829 * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
832 * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
833 * format. The result of this part is middle_frame is guaranteed to be in
834 * SLINEAR format for Part_2.
835 * Part_2: Send middle_frame off to spies and manipulators. At this point middle_frame is
836 * either a new frame as result of the translation, or points directly to the start_frame
837 * because no translation to SLINEAR audio was required.
838 * Part_3: Translate end_frame's audio back into the format of start frame if necessary. This
839 * is only necessary if manipulation of middle_frame occurred.
841 * \param chan Channel that the list is coming off of
842 * \param audiohook_list List of audiohooks
843 * \param direction Direction frame is coming in from
844 * \param frame The frame itself
845 * \return Return frame on success, NULL on failure
847 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)
849 struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
850 struct ast_audiohook *audiohook = NULL;
852 int middle_frame_manipulated = 0;
855 /* ---Part_1. translate start_frame to SLINEAR if necessary. */
856 if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
859 samples = middle_frame->samples;
861 /* ---Part_2: Send middle_frame to spy and manipulator lists. middle_frame is guaranteed to be SLINEAR here.*/
862 /* Queue up signed linear frame to each spy */
863 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
864 ast_audiohook_lock(audiohook);
865 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
866 AST_LIST_REMOVE_CURRENT(list);
868 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
869 ast_audiohook_unlock(audiohook);
870 if (ast_channel_is_bridged(chan)) {
871 ast_channel_set_unbridged_nolock(chan, 1);
875 audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
876 ast_audiohook_write_frame(audiohook, direction, middle_frame);
877 ast_audiohook_unlock(audiohook);
879 AST_LIST_TRAVERSE_SAFE_END;
881 /* If this frame is being written out to the channel then we need to use whisper sources */
882 if (!AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
884 short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
885 memset(&combine_buf, 0, sizeof(combine_buf));
886 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
887 struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
888 ast_audiohook_lock(audiohook);
889 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
890 AST_LIST_REMOVE_CURRENT(list);
892 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
893 ast_audiohook_unlock(audiohook);
894 if (ast_channel_is_bridged(chan)) {
895 ast_channel_set_unbridged_nolock(chan, 1);
899 audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
900 if (ast_slinfactory_available(factory) >= samples && ast_slinfactory_read(factory, read_buf, samples)) {
901 /* Take audio from this whisper source and combine it into our main buffer */
902 for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++) {
903 ast_slinear_saturated_add(data1, data2);
906 ast_audiohook_unlock(audiohook);
908 AST_LIST_TRAVERSE_SAFE_END;
909 /* We take all of the combined whisper sources and combine them into the audio being written out */
910 for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++) {
911 ast_slinear_saturated_add(data1, data2);
913 middle_frame_manipulated = 1;
916 /* Pass off frame to manipulate audiohooks */
917 if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
918 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
919 ast_audiohook_lock(audiohook);
920 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
921 AST_LIST_REMOVE_CURRENT(list);
923 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
924 ast_audiohook_unlock(audiohook);
925 /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
926 audiohook->manipulate_callback(audiohook, chan, NULL, direction);
927 if (ast_channel_is_bridged(chan)) {
928 ast_channel_set_unbridged_nolock(chan, 1);
932 audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
933 /* Feed in frame to manipulation. */
934 if (!audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
935 /* If the manipulation fails then the frame will be returned in its original state.
936 * Since there are potentially more manipulator callbacks in the list, no action should
937 * be taken here to exit early. */
938 middle_frame_manipulated = 1;
940 ast_audiohook_unlock(audiohook);
942 AST_LIST_TRAVERSE_SAFE_END;
945 /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
946 if (middle_frame_manipulated) {
947 if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, start_frame->subclass.format))) {
948 /* translation failed, so just pass back the input frame */
949 end_frame = start_frame;
952 end_frame = start_frame;
954 /* clean up our middle_frame if required */
955 if (middle_frame != end_frame) {
956 ast_frfree(middle_frame);
960 /* Before returning, if an audiohook got removed, reset samplerate compatibility */
962 audiohook_list_set_samplerate_compatibility(audiohook_list);
968 int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
970 return !audiohook_list
971 || (AST_LIST_EMPTY(&audiohook_list->spy_list)
972 && AST_LIST_EMPTY(&audiohook_list->whisper_list)
973 && AST_LIST_EMPTY(&audiohook_list->manipulate_list));
976 /*! \brief Pass a frame off to be handled by the audiohook core
977 * \param chan Channel that the list is coming off of
978 * \param audiohook_list List of audiohooks
979 * \param direction Direction frame is coming in from
980 * \param frame The frame itself
981 * \return Return frame on success, NULL on failure
983 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)
985 /* Pass off frame to it's respective list write function */
986 if (frame->frametype == AST_FRAME_VOICE) {
987 return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
988 } else if (frame->frametype == AST_FRAME_DTMF) {
989 return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
995 /*! \brief Wait for audiohook trigger to be triggered
996 * \param audiohook Audiohook to wait on
998 void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
1000 struct timeval wait;
1003 wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
1004 ts.tv_sec = wait.tv_sec;
1005 ts.tv_nsec = wait.tv_usec * 1000;
1007 ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
1012 /* Count number of channel audiohooks by type, regardless of type */
1013 int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
1016 struct ast_audiohook *ah = NULL;
1018 if (!ast_channel_audiohooks(chan)) {
1023 case AST_AUDIOHOOK_TYPE_SPY:
1024 AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
1025 if (!strcmp(ah->source, source)) {
1030 case AST_AUDIOHOOK_TYPE_WHISPER:
1031 AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
1032 if (!strcmp(ah->source, source)) {
1037 case AST_AUDIOHOOK_TYPE_MANIPULATE:
1038 AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
1039 if (!strcmp(ah->source, source)) {
1045 ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
1052 /* Count number of channel audiohooks by type that are running */
1053 int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
1056 struct ast_audiohook *ah = NULL;
1057 if (!ast_channel_audiohooks(chan))
1061 case AST_AUDIOHOOK_TYPE_SPY:
1062 AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
1063 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1067 case AST_AUDIOHOOK_TYPE_WHISPER:
1068 AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
1069 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1073 case AST_AUDIOHOOK_TYPE_MANIPULATE:
1074 AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
1075 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1080 ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
1086 /*! \brief Audiohook volume adjustment structure */
1087 struct audiohook_volume {
1088 struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
1089 int read_adjustment; /*!< Value to adjust frames read from the channel by */
1090 int write_adjustment; /*!< Value to adjust frames written to the channel by */
1093 /*! \brief Callback used to destroy the audiohook volume datastore
1094 * \param data Volume information structure
1095 * \return Returns nothing
1097 static void audiohook_volume_destroy(void *data)
1099 struct audiohook_volume *audiohook_volume = data;
1101 /* Destroy the audiohook as it is no longer in use */
1102 ast_audiohook_destroy(&audiohook_volume->audiohook);
1104 /* Finally free ourselves, we are of no more use */
1105 ast_free(audiohook_volume);
1110 /*! \brief Datastore used to store audiohook volume information */
1111 static const struct ast_datastore_info audiohook_volume_datastore = {
1113 .destroy = audiohook_volume_destroy,
1116 /*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
1117 * \param audiohook Audiohook attached to the channel
1118 * \param chan Channel we are attached to
1119 * \param frame Frame of audio we want to manipulate
1120 * \param direction Direction the audio came in from
1121 * \return Returns 0 on success, -1 on failure
1123 static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
1125 struct ast_datastore *datastore = NULL;
1126 struct audiohook_volume *audiohook_volume = NULL;
1129 /* If the audiohook is shutting down don't even bother */
1130 if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
1134 /* Try to find the datastore containg adjustment information, if we can't just bail out */
1135 if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
1139 audiohook_volume = datastore->data;
1141 /* Based on direction grab the appropriate adjustment value */
1142 if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
1143 gain = &audiohook_volume->read_adjustment;
1144 } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
1145 gain = &audiohook_volume->write_adjustment;
1148 /* If an adjustment value is present modify the frame */
1149 if (gain && *gain) {
1150 ast_frame_adjust_volume(frame, *gain);
1156 /*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
1157 * \param chan Channel to look on
1158 * \param create Whether to create the datastore if not found
1159 * \return Returns audiohook_volume structure on success, NULL on failure
1161 static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
1163 struct ast_datastore *datastore = NULL;
1164 struct audiohook_volume *audiohook_volume = NULL;
1166 /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
1167 if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
1168 return datastore->data;
1171 /* 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 */
1172 if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
1176 /* Create a new audiohook_volume structure to contain our adjustments and audiohook */
1177 if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
1178 ast_datastore_free(datastore);
1182 /* Setup our audiohook structure so we can manipulate the audio */
1183 ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
1184 audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback;
1186 /* Attach the audiohook_volume blob to the datastore and attach to the channel */
1187 datastore->data = audiohook_volume;
1188 ast_channel_datastore_add(chan, datastore);
1190 /* All is well... put the audiohook into motion */
1191 ast_audiohook_attach(chan, &audiohook_volume->audiohook);
1193 return audiohook_volume;
1196 /*! \brief Adjust the volume on frames read from or written to a channel
1197 * \param chan Channel to muck with
1198 * \param direction Direction to set on
1199 * \param volume Value to adjust the volume by
1200 * \return Returns 0 on success, -1 on failure
1202 int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
1204 struct audiohook_volume *audiohook_volume = NULL;
1206 /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
1207 if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
1211 /* Now based on the direction set the proper value */
1212 if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1213 audiohook_volume->read_adjustment = volume;
1215 if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1216 audiohook_volume->write_adjustment = volume;
1222 /*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
1223 * \param chan Channel to retrieve volume adjustment from
1224 * \param direction Direction to retrieve
1225 * \return Returns adjustment value
1227 int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
1229 struct audiohook_volume *audiohook_volume = NULL;
1232 /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
1233 if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
1237 /* Grab the adjustment value based on direction given */
1238 if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
1239 adjustment = audiohook_volume->read_adjustment;
1240 } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
1241 adjustment = audiohook_volume->write_adjustment;
1247 /*! \brief Adjust the volume on frames read from or written to a channel
1248 * \param chan Channel to muck with
1249 * \param direction Direction to increase
1250 * \param volume Value to adjust the adjustment by
1251 * \return Returns 0 on success, -1 on failure
1253 int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
1255 struct audiohook_volume *audiohook_volume = NULL;
1257 /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
1258 if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
1262 /* Based on the direction change the specific adjustment value */
1263 if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1264 audiohook_volume->read_adjustment += volume;
1266 if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1267 audiohook_volume->write_adjustment += volume;
1273 /*! \brief Mute frames read from or written to a channel
1274 * \param chan Channel to muck with
1275 * \param source Type of audiohook
1276 * \param flag which flag to set / clear
1277 * \param clear set or clear
1278 * \return Returns 0 on success, -1 on failure
1280 int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
1282 struct ast_audiohook *audiohook = NULL;
1284 ast_channel_lock(chan);
1286 /* Ensure the channel has audiohooks on it */
1287 if (!ast_channel_audiohooks(chan)) {
1288 ast_channel_unlock(chan);
1292 audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
1296 ast_clear_flag(audiohook, flag);
1298 ast_set_flag(audiohook, flag);
1302 ast_channel_unlock(chan);
1304 return (audiohook ? 0 : -1);