1883c0091c42a5c7511fbc727dc4dc69f5c0df55
[asterisk/asterisk.git] / main / audiohook.c
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 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
33
34 #include <signal.h>
35
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"
45
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. */
48
49 struct ast_audiohook_translate {
50         struct ast_trans_pvt *trans_pvt;
51         struct ast_format *format;
52 };
53
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 */
61
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;
67 };
68
69 static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
70 {
71         struct ast_format *slin;
72
73         if (audiohook->hook_internal_samp_rate == rate) {
74                 return 0;
75         }
76
77         audiohook->hook_internal_samp_rate = rate;
78
79         slin = ast_format_cache_get_slin_by_rate(rate);
80
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:
85                 if (reset) {
86                         ast_slinfactory_destroy(&audiohook->read_factory);
87                         ast_slinfactory_destroy(&audiohook->write_factory);
88                 }
89                 ast_slinfactory_init_with_format(&audiohook->read_factory, slin);
90                 ast_slinfactory_init_with_format(&audiohook->write_factory, slin);
91                 break;
92         default:
93                 break;
94         }
95
96         return 0;
97 }
98
99 /*! \brief Initialize an audiohook structure
100  *
101  * \param audiohook Audiohook structure
102  * \param type
103  * \param source, init_flags
104  *
105  * \return Returns 0 on success, -1 on failure
106  */
107 int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags init_flags)
108 {
109         /* Need to keep the type and source */
110         audiohook->type = type;
111         audiohook->source = source;
112
113         /* Initialize lock that protects our audiohook */
114         ast_mutex_init(&audiohook->lock);
115         ast_cond_init(&audiohook->trigger, NULL);
116
117         audiohook->init_flags = init_flags;
118
119         /* initialize internal rate at 8khz, this will adjust if necessary */
120         audiohook_set_internal_rate(audiohook, 8000, 0);
121
122         /* Since we are just starting out... this audiohook is new */
123         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
124
125         return 0;
126 }
127
128 /*! \brief Destroys an audiohook structure
129  * \param audiohook Audiohook structure
130  * \return Returns 0 on success, -1 on failure
131  */
132 int ast_audiohook_destroy(struct ast_audiohook *audiohook)
133 {
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);
140                 break;
141         default:
142                 break;
143         }
144
145         /* Destroy translation path if present */
146         if (audiohook->trans_pvt)
147                 ast_translator_free_path(audiohook->trans_pvt);
148
149         ao2_cleanup(audiohook->format);
150
151         /* Lock and trigger be gone! */
152         ast_cond_destroy(&audiohook->trigger);
153         ast_mutex_destroy(&audiohook->lock);
154
155         return 0;
156 }
157
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
163  */
164 int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
165 {
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;
170         int our_factory_ms;
171         int other_factory_samples;
172         int other_factory_ms;
173         int muteme = 0;
174
175         /* Update last feeding time to be current */
176         *rwtime = ast_tvnow();
177
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);
182
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);
187         }
188
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);
193         }
194
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))) {
199                         muteme = 1;
200         }
201
202         if (muteme && frame->datalen > 0) {
203                 ast_frame_clear(frame);
204         }
205
206         /* Write frame out to respective factory */
207         ast_slinfactory_feed(factory, frame);
208
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);
216         }
217
218         return 0;
219 }
220
221 static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
222 {
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);
225         short buf[samples];
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),
229                 .data.ptr = buf,
230                 .datalen = sizeof(buf),
231                 .samples = samples,
232         };
233
234         /* Ensure the factory is able to give us the samples we want */
235         if (samples > ast_slinfactory_available(factory)) {
236                 return NULL;
237         }
238
239         /* Read data in from factory */
240         if (!ast_slinfactory_read(factory, buf, samples)) {
241                 return NULL;
242         }
243
244         /* If a volume adjustment needs to be applied apply it */
245         if (vol) {
246                 ast_frame_adjust_volume(&frame, vol);
247         }
248
249         return ast_frdup(&frame);
250 }
251
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)
253 {
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,
258                 .data.ptr = NULL,
259                 .datalen = sizeof(buf1),
260                 .samples = samples,
261         };
262
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);
266
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);
270                 return NULL;
271         }
272
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);
276                 return NULL;
277         }
278
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);
282                 return NULL;
283         }
284
285         /* Start with the read factory... if there are enough samples, read them in */
286         if (usable_read) {
287                 if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
288                         read_buf = buf1;
289                         /* Adjust read volume if need be */
290                         if (audiohook->options.read_volume) {
291                                 int count = 0;
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);
298                                         }
299                                 }
300                         }
301                 }
302         } else {
303                 ast_debug(1, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
304         }
305
306         /* Move on to the write factory... if there are enough samples, read them in */
307         if (usable_write) {
308                 if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
309                         write_buf = buf2;
310                         /* Adjust write volume if need be */
311                         if (audiohook->options.write_volume) {
312                                 int count = 0;
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);
319                                         }
320                                 }
321                         }
322                 }
323         } else {
324                 ast_debug(1, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
325         }
326
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);
331         }
332         if (write_buf && write_reference) {
333                 frame.data.ptr = buf2;
334                 *write_reference = ast_frdup(&frame);
335         }
336
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);
340                 }
341                 final_buf = buf1;
342         } else if (read_buf) {
343                 final_buf = buf1;
344         } else if (write_buf) {
345                 final_buf = buf2;
346         } else {
347                 return NULL;
348         }
349
350         /* Make the final buffer part of the frame, so it gets duplicated fine */
351         frame.data.ptr = final_buf;
352
353         frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
354
355         /* Yahoo, a combined copy of the audio! */
356         return ast_frdup(&frame);
357 }
358
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)
360 {
361         struct ast_frame *read_frame = NULL, *final_frame = NULL;
362         struct ast_format *slin;
363
364         audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1);
365
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)))) {
369                 return NULL;
370         }
371
372         slin = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
373
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;
381                         }
382
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);
386                                 return NULL;
387                         }
388                         ao2_replace(audiohook->format, format);
389                 }
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);
392         } else {
393                 final_frame = read_frame;
394         }
395
396         return final_frame;
397 }
398
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
405  */
406 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format)
407 {
408         return audiohook_read_frame_helper(audiohook, samples, direction, format, NULL, NULL);
409 }
410
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
419  */
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)
421 {
422         return audiohook_read_frame_helper(audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, format, read_frame, write_frame);
423 }
424
425 static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list)
426 {
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;
432                         return;
433                 }
434         }
435 }
436
437 /*! \brief Attach audiohook to channel
438  * \param chan Channel
439  * \param audiohook Audiohook structure
440  * \return Returns 0 on success, -1 on failure
441  */
442 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
443 {
444         ast_channel_lock(chan);
445
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);
451                         return -1;
452                 }
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;
459         }
460
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);
468         }
469
470
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));
473
474         /* Change status over to running since it is now attached */
475         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
476
477         if (ast_channel_is_bridged(chan)) {
478                 ast_channel_set_unbridged_nolock(chan, 1);
479         }
480
481         ast_channel_unlock(chan);
482
483         return 0;
484 }
485
486 /*! \brief Update audiohook's status
487  * \param audiohook Audiohook structure
488  * \param status Audiohook status enum
489  *
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.
492  */
493
494 void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
495 {
496         ast_audiohook_lock(audiohook);
497         if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
498                 audiohook->status = status;
499                 ast_cond_signal(&audiohook->trigger);
500         }
501         ast_audiohook_unlock(audiohook);
502 }
503
504 /*! \brief Detach audiohook from channel
505  * \param audiohook Audiohook structure
506  * \return Returns 0 on success, -1 on failure
507  */
508 int ast_audiohook_detach(struct ast_audiohook *audiohook)
509 {
510         if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
511                 return 0;
512         }
513
514         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
515
516         while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
517                 ast_audiohook_trigger_wait(audiohook);
518         }
519
520         return 0;
521 }
522
523 void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
524 {
525         int i;
526         struct ast_audiohook *audiohook;
527
528         if (!audiohook_list) {
529                 return;
530         }
531
532         /* Drop any spies */
533         while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
534                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
535         }
536
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);
540         }
541
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);
546         }
547
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);
553                 }
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);
557                 }
558         }
559
560         /* Free ourselves */
561         ast_free(audiohook_list);
562 }
563
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.
568  */
569 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
570 {
571         struct ast_audiohook *audiohook = NULL;
572
573         AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
574                 if (!strcasecmp(audiohook->source, source)) {
575                         return audiohook;
576                 }
577         }
578
579         AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
580                 if (!strcasecmp(audiohook->source, source)) {
581                         return audiohook;
582                 }
583         }
584
585         AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
586                 if (!strcasecmp(audiohook->source, source)) {
587                         return audiohook;
588                 }
589         }
590
591         return NULL;
592 }
593
594 static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook)
595 {
596         enum ast_audiohook_status oldstatus;
597
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
601          * condition.
602          */
603         ast_audiohook_lock(audiohook);
604         oldstatus = audiohook->status;
605
606         ast_audiohook_remove(old_chan, audiohook);
607         ast_audiohook_attach(new_chan, audiohook);
608
609         audiohook->status = oldstatus;
610         ast_audiohook_unlock(audiohook);
611 }
612
613 void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
614 {
615         struct ast_audiohook *audiohook;
616
617         if (!ast_channel_audiohooks(old_chan)) {
618                 return;
619         }
620
621         audiohook = find_audiohook_by_source(ast_channel_audiohooks(old_chan), source);
622         if (!audiohook) {
623                 return;
624         }
625
626         audiohook_move(old_chan, new_chan, audiohook);
627 }
628
629 void ast_audiohook_move_all(struct ast_channel *old_chan, struct ast_channel *new_chan)
630 {
631         struct ast_audiohook *audiohook;
632         struct ast_audiohook_list *audiohook_list;
633
634         audiohook_list = ast_channel_audiohooks(old_chan);
635         if (!audiohook_list) {
636                 return;
637         }
638
639         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
640                 audiohook_move(old_chan, new_chan, audiohook);
641         }
642         AST_LIST_TRAVERSE_SAFE_END;
643
644         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
645                 audiohook_move(old_chan, new_chan, audiohook);
646         }
647         AST_LIST_TRAVERSE_SAFE_END;
648
649         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
650                 audiohook_move(old_chan, new_chan, audiohook);
651         }
652         AST_LIST_TRAVERSE_SAFE_END;
653 }
654
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
659  */
660 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
661 {
662         struct ast_audiohook *audiohook = NULL;
663
664         ast_channel_lock(chan);
665
666         /* Ensure the channel has audiohooks on it */
667         if (!ast_channel_audiohooks(chan)) {
668                 ast_channel_unlock(chan);
669                 return -1;
670         }
671
672         audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
673
674         ast_channel_unlock(chan);
675
676         if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
677                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
678         }
679
680         return (audiohook ? 0 : -1);
681 }
682
683 /*!
684  * \brief Remove an audiohook from a specified channel
685  *
686  * \param chan Channel to remove from
687  * \param audiohook Audiohook to remove
688  *
689  * \return Returns 0 on success, -1 on failure
690  *
691  * \note The channel does not need to be locked before calling this function
692  */
693 int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
694 {
695         ast_channel_lock(chan);
696
697         if (!ast_channel_audiohooks(chan)) {
698                 ast_channel_unlock(chan);
699                 return -1;
700         }
701
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);
708         }
709
710         audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
711         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
712
713         if (ast_channel_is_bridged(chan)) {
714                 ast_channel_set_unbridged_nolock(chan, 1);
715         }
716
717         ast_channel_unlock(chan);
718
719         return 0;
720 }
721
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
728  */
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)
730 {
731         struct ast_audiohook *audiohook = NULL;
732         int removed = 0;
733
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);
738                         removed = 1;
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);
744                         }
745                         continue;
746                 }
747                 if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF)) {
748                         audiohook->manipulate_callback(audiohook, chan, frame, direction);
749                 }
750                 ast_audiohook_unlock(audiohook);
751         }
752         AST_LIST_TRAVERSE_SAFE_END;
753
754         /* if an audiohook got removed, reset samplerate compatibility */
755         if (removed) {
756                 audiohook_list_set_samplerate_compatibility(audiohook_list);
757         }
758         return frame;
759 }
760
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)
763 {
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;
768
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);
776         }
777
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) {
780                 return new_frame;
781         }
782
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);
786                 }
787                 if (!(in_translate->trans_pvt = ast_translator_build_path(slin, frame->subclass.format))) {
788                         return NULL;
789                 }
790                 ao2_replace(in_translate->format, frame->subclass.format);
791         }
792
793         if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) {
794                 return NULL;
795         }
796
797         return new_frame;
798 }
799
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)
802 {
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);
810                         }
811                         if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) {
812                                 return NULL;
813                         }
814                         ao2_replace(out_translate->format, outformat);
815                 }
816                 /* translate back to the format the frame came in as. */
817                 if (!(outframe = ast_translate(out_translate->trans_pvt, slin_frame, 0))) {
818                         return NULL;
819                 }
820         }
821         return outframe;
822 }
823
824 /*!
825  * \brief Pass an AUDIO frame off to be handled by the audiohook core
826  *
827  * \details
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
830  * input frame.
831  *
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.
840  *
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
846  */
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)
848 {
849         struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
850         struct ast_audiohook *audiohook = NULL;
851         int samples;
852         int middle_frame_manipulated = 0;
853         int removed = 0;
854
855         /* ---Part_1. translate start_frame to SLINEAR if necessary. */
856         if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
857                 return frame;
858         }
859         samples = middle_frame->samples;
860
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);
867                         removed = 1;
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);
872                         }
873                         continue;
874                 }
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);
878         }
879         AST_LIST_TRAVERSE_SAFE_END;
880
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)) {
883                 int i = 0;
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);
891                                 removed = 1;
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);
896                                 }
897                                 continue;
898                         }
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);
904                                 }
905                         }
906                         ast_audiohook_unlock(audiohook);
907                 }
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);
912                 }
913                 middle_frame_manipulated = 1;
914         }
915
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);
922                                 removed = 1;
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);
929                                 }
930                                 continue;
931                         }
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;
939                         }
940                         ast_audiohook_unlock(audiohook);
941                 }
942                 AST_LIST_TRAVERSE_SAFE_END;
943         }
944
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;
950                 }
951         } else {
952                 end_frame = start_frame;
953         }
954         /* clean up our middle_frame if required */
955         if (middle_frame != end_frame) {
956                 ast_frfree(middle_frame);
957                 middle_frame = NULL;
958         }
959
960         /* Before returning, if an audiohook got removed, reset samplerate compatibility */
961         if (removed) {
962                 audiohook_list_set_samplerate_compatibility(audiohook_list);
963         }
964
965         return end_frame;
966 }
967
968 int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
969 {
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));
974 }
975
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
982  */
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)
984 {
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);
990         } else {
991                 return frame;
992         }
993 }
994
995 /*! \brief Wait for audiohook trigger to be triggered
996  * \param audiohook Audiohook to wait on
997  */
998 void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
999 {
1000         struct timeval wait;
1001         struct timespec ts;
1002
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;
1006
1007         ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
1008
1009         return;
1010 }
1011
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)
1014 {
1015         int count = 0;
1016         struct ast_audiohook *ah = NULL;
1017
1018         if (!ast_channel_audiohooks(chan)) {
1019                 return -1;
1020         }
1021
1022         switch (type) {
1023                 case AST_AUDIOHOOK_TYPE_SPY:
1024                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
1025                                 if (!strcmp(ah->source, source)) {
1026                                         count++;
1027                                 }
1028                         }
1029                         break;
1030                 case AST_AUDIOHOOK_TYPE_WHISPER:
1031                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
1032                                 if (!strcmp(ah->source, source)) {
1033                                         count++;
1034                                 }
1035                         }
1036                         break;
1037                 case AST_AUDIOHOOK_TYPE_MANIPULATE:
1038                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
1039                                 if (!strcmp(ah->source, source)) {
1040                                         count++;
1041                                 }
1042                         }
1043                         break;
1044                 default:
1045                         ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
1046                         return -1;
1047         }
1048
1049         return count;
1050 }
1051
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)
1054 {
1055         int count = 0;
1056         struct ast_audiohook *ah = NULL;
1057         if (!ast_channel_audiohooks(chan))
1058                 return -1;
1059
1060         switch (type) {
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))
1064                                         count++;
1065                         }
1066                         break;
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))
1070                                         count++;
1071                         }
1072                         break;
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))
1076                                         count++;
1077                         }
1078                         break;
1079                 default:
1080                         ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
1081                         return -1;
1082         }
1083         return count;
1084 }
1085
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 */
1091 };
1092
1093 /*! \brief Callback used to destroy the audiohook volume datastore
1094  * \param data Volume information structure
1095  * \return Returns nothing
1096  */
1097 static void audiohook_volume_destroy(void *data)
1098 {
1099         struct audiohook_volume *audiohook_volume = data;
1100
1101         /* Destroy the audiohook as it is no longer in use */
1102         ast_audiohook_destroy(&audiohook_volume->audiohook);
1103
1104         /* Finally free ourselves, we are of no more use */
1105         ast_free(audiohook_volume);
1106
1107         return;
1108 }
1109
1110 /*! \brief Datastore used to store audiohook volume information */
1111 static const struct ast_datastore_info audiohook_volume_datastore = {
1112         .type = "Volume",
1113         .destroy = audiohook_volume_destroy,
1114 };
1115
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
1122  */
1123 static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
1124 {
1125         struct ast_datastore *datastore = NULL;
1126         struct audiohook_volume *audiohook_volume = NULL;
1127         int *gain = NULL;
1128
1129         /* If the audiohook is shutting down don't even bother */
1130         if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
1131                 return 0;
1132         }
1133
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))) {
1136                 return 0;
1137         }
1138
1139         audiohook_volume = datastore->data;
1140
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;
1146         }
1147
1148         /* If an adjustment value is present modify the frame */
1149         if (gain && *gain) {
1150                 ast_frame_adjust_volume(frame, *gain);
1151         }
1152
1153         return 0;
1154 }
1155
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
1160  */
1161 static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
1162 {
1163         struct ast_datastore *datastore = NULL;
1164         struct audiohook_volume *audiohook_volume = NULL;
1165
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;
1169         }
1170
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))) {
1173                 return NULL;
1174         }
1175
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);
1179                 return NULL;
1180         }
1181
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;
1185
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);
1189
1190         /* All is well... put the audiohook into motion */
1191         ast_audiohook_attach(chan, &audiohook_volume->audiohook);
1192
1193         return audiohook_volume;
1194 }
1195
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
1201  */
1202 int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
1203 {
1204         struct audiohook_volume *audiohook_volume = NULL;
1205
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)))) {
1208                 return -1;
1209         }
1210
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;
1214         }
1215         if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1216                 audiohook_volume->write_adjustment = volume;
1217         }
1218
1219         return 0;
1220 }
1221
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
1226  */
1227 int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
1228 {
1229         struct audiohook_volume *audiohook_volume = NULL;
1230         int adjustment = 0;
1231
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))) {
1234                 return 0;
1235         }
1236
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;
1242         }
1243
1244         return adjustment;
1245 }
1246
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
1252  */
1253 int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
1254 {
1255         struct audiohook_volume *audiohook_volume = NULL;
1256
1257         /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
1258         if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
1259                 return -1;
1260         }
1261
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;
1265         }
1266         if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1267                 audiohook_volume->write_adjustment += volume;
1268         }
1269
1270         return 0;
1271 }
1272
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
1279  */
1280 int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
1281 {
1282         struct ast_audiohook *audiohook = NULL;
1283
1284         ast_channel_lock(chan);
1285
1286         /* Ensure the channel has audiohooks on it */
1287         if (!ast_channel_audiohooks(chan)) {
1288                 ast_channel_unlock(chan);
1289                 return -1;
1290         }
1291
1292         audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
1293
1294         if (audiohook) {
1295                 if (clear) {
1296                         ast_clear_flag(audiohook, flag);
1297                 } else {
1298                         ast_set_flag(audiohook, flag);
1299                 }
1300         }
1301
1302         ast_channel_unlock(chan);
1303
1304         return (audiohook ? 0 : -1);
1305 }