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 #include "asterisk.h"
  27
  28 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
  29
  30 #include <signal.h>
  31
  32 #include "asterisk/channel.h"
  33 #include "asterisk/utils.h"
  34 #include "asterisk/lock.h"
  35 #include "asterisk/linkedlists.h"
  36 #include "asterisk/audiohook.h"
  37 #include "asterisk/slinfactory.h"
  38 #include "asterisk/frame.h"
  39 #include "asterisk/translate.h"
  40
  41 struct ast_audiohook_translate {
  42         struct ast_trans_pvt *trans_pvt;
  43         format_t format;
  44 };
  45
  46 struct ast_audiohook_list {
  47         struct ast_audiohook_translate in_translate[2];
  48         struct ast_audiohook_translate out_translate[2];
  49         AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
  50         AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
  51         AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
  52 };
  53
  54 /*! \brief Initialize an audiohook structure
  55  * \param audiohook Audiohook structure
  56  * \param type
  57  * \param source
  58  * \return Returns 0 on success, -1 on failure
  59  */
  60 int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source)
  61 {
  62         /* Need to keep the type and source */
  63         audiohook->type = type;
  64         audiohook->source = source;
  65
  66         /* Initialize lock that protects our audiohook */
  67         ast_mutex_init(&audiohook->lock);
  68         ast_cond_init(&audiohook->trigger, NULL);
  69
  70         /* Setup the factories that are needed for this audiohook type */
  71         switch (type) {
  72         case AST_AUDIOHOOK_TYPE_SPY:
  73                 ast_slinfactory_init(&audiohook->read_factory);
  74         case AST_AUDIOHOOK_TYPE_WHISPER:
  75                 ast_slinfactory_init(&audiohook->write_factory);
  76                 break;
  77         default:
  78                 break;
  79         }
  80
  81         /* Since we are just starting out... this audiohook is new */
  82         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
  83
  84         return 0;
  85 }
  86
  87 /*! \brief Destroys an audiohook structure
  88  * \param audiohook Audiohook structure
  89  * \return Returns 0 on success, -1 on failure
  90  */
  91 int ast_audiohook_destroy(struct ast_audiohook *audiohook)
  92 {
  93         /* Drop the factories used by this audiohook type */
  94         switch (audiohook->type) {
  95         case AST_AUDIOHOOK_TYPE_SPY:
  96                 ast_slinfactory_destroy(&audiohook->read_factory);
  97         case AST_AUDIOHOOK_TYPE_WHISPER:
  98                 ast_slinfactory_destroy(&audiohook->write_factory);
  99                 break;
 100         default:
 101                 break;
 102         }
 103
 104         /* Destroy translation path if present */
 105         if (audiohook->trans_pvt)
 106                 ast_translator_free_path(audiohook->trans_pvt);
 107
 108         /* Lock and trigger be gone! */
 109         ast_cond_destroy(&audiohook->trigger);
 110         ast_mutex_destroy(&audiohook->lock);
 111
 112         return 0;
 113 }
 114
 115 /*! \brief Writes a frame into the audiohook structure
 116  * \param audiohook Audiohook structure
 117  * \param direction Direction the audio frame came from
 118  * \param frame Frame to write in
 119  * \return Returns 0 on success, -1 on failure
 120  */
 121 int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
 122 {
 123         struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
 124         struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
 125         struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime;
 126         int our_factory_samples;
 127         int our_factory_ms;
 128         int other_factory_samples;
 129         int other_factory_ms;
 130         int muteme = 0;
 131
 132         /* Update last feeding time to be current */
 133         *rwtime = ast_tvnow();
 134
 135         our_factory_samples = ast_slinfactory_available(factory);
 136         our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / 8);
 137         other_factory_samples = ast_slinfactory_available(other_factory);
 138         other_factory_ms = other_factory_samples / 8;
 139
 140         if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && other_factory_samples && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
 141                 if (option_debug)
 142                         ast_log(LOG_DEBUG, "Flushing audiohook %p so it remains in sync\n", audiohook);
 143                 ast_slinfactory_flush(factory);
 144                 ast_slinfactory_flush(other_factory);
 145         }
 146
 147         if (ast_test_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE) && (our_factory_samples > 640 || other_factory_samples > 640)) {
 148                 if (option_debug) {
 149                         ast_log(LOG_DEBUG, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
 150                 }
 151                 ast_slinfactory_flush(factory);
 152                 ast_slinfactory_flush(other_factory);
 153         }
 154
 155         /* swap frame data for zeros if mute is required */
 156         if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) ||
 157                 (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) ||
 158                 (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE))) {
 159                         muteme = 1;
 160         }
 161
 162         if (muteme && frame->datalen > 0) {
 163                 ast_frame_clear(frame);
 164         }
 165
 166         /* Write frame out to respective factory */
 167         ast_slinfactory_feed(factory, frame);
 168
 169         /* If we need to notify the respective handler of this audiohook, do so */
 170         if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
 171                 ast_cond_signal(&audiohook->trigger);
 172         } else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
 173                 ast_cond_signal(&audiohook->trigger);
 174         } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
 175                 ast_cond_signal(&audiohook->trigger);
 176         }
 177
 178         return 0;
 179 }
 180
 181 static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
 182 {
 183         struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
 184         int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
 185         short buf[samples];
 186         struct ast_frame frame = {
 187                 .frametype = AST_FRAME_VOICE,
 188                 .subclass.codec = AST_FORMAT_SLINEAR,
 189                 .data.ptr = buf,
 190                 .datalen = sizeof(buf),
 191                 .samples = samples,
 192         };
 193
 194         /* Ensure the factory is able to give us the samples we want */
 195         if (samples > ast_slinfactory_available(factory))
 196                 return NULL;
 197
 198         /* Read data in from factory */
 199         if (!ast_slinfactory_read(factory, buf, samples))
 200                 return NULL;
 201
 202         /* If a volume adjustment needs to be applied apply it */
 203         if (vol)
 204                 ast_frame_adjust_volume(&frame, vol);
 205
 206         return ast_frdup(&frame);
 207 }
 208
 209 static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
 210 {
 211         int i = 0, usable_read, usable_write;
 212         short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
 213         struct ast_frame frame = {
 214                 .frametype = AST_FRAME_VOICE,
 215                 .subclass.codec = AST_FORMAT_SLINEAR,
 216                 .data.ptr = NULL,
 217                 .datalen = sizeof(buf1),
 218                 .samples = samples,
 219         };
 220
 221         /* Make sure both factories have the required samples */
 222         usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
 223         usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
 224
 225         if (!usable_read && !usable_write) {
 226                 /* If both factories are unusable bail out */
 227                 ast_debug(1, "Read factory %p and write factory %p both fail to provide %zd samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
 228                 return NULL;
 229         }
 230
 231         /* If we want to provide only a read factory make sure we aren't waiting for other audio */
 232         if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
 233                 ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
 234                 return NULL;
 235         }
 236
 237         /* If we want to provide only a write factory make sure we aren't waiting for other audio */
 238         if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
 239                 ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
 240                 return NULL;
 241         }
 242
 243         /* Start with the read factory... if there are enough samples, read them in */
 244         if (usable_read) {
 245                 if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
 246                         read_buf = buf1;
 247                         /* Adjust read volume if need be */
 248                         if (audiohook->options.read_volume) {
 249                                 int count = 0;
 250                                 short adjust_value = abs(audiohook->options.read_volume);
 251                                 for (count = 0; count < samples; count++) {
 252                                         if (audiohook->options.read_volume > 0)
 253                                                 ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
 254                                         else if (audiohook->options.read_volume < 0)
 255                                                 ast_slinear_saturated_divide(&buf1[count], &adjust_value);
 256                                 }
 257                         }
 258                 }
 259         } else if (option_debug)
 260                 ast_log(LOG_DEBUG, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
 261
 262         /* Move on to the write factory... if there are enough samples, read them in */
 263         if (usable_write) {
 264                 if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
 265                         write_buf = buf2;
 266                         /* Adjust write volume if need be */
 267                         if (audiohook->options.write_volume) {
 268                                 int count = 0;
 269                                 short adjust_value = abs(audiohook->options.write_volume);
 270                                 for (count = 0; count < samples; count++) {
 271                                         if (audiohook->options.write_volume > 0)
 272                                                 ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
 273                                         else if (audiohook->options.write_volume < 0)
 274                                                 ast_slinear_saturated_divide(&buf2[count], &adjust_value);
 275                                 }
 276                         }
 277                 }
 278         } else if (option_debug)
 279                 ast_log(LOG_DEBUG, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
 280
 281         /* Basically we figure out which buffer to use... and if mixing can be done here */
 282         if (!read_buf && !write_buf)
 283                 return NULL;
 284         else if (read_buf && write_buf) {
 285                 for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
 286                         ast_slinear_saturated_add(data1, data2);
 287                 final_buf = buf1;
 288         } else if (read_buf)
 289                 final_buf = buf1;
 290         else if (write_buf)
 291                 final_buf = buf2;
 292
 293         /* Make the final buffer part of the frame, so it gets duplicated fine */
 294         frame.data.ptr = final_buf;
 295
 296         /* Yahoo, a combined copy of the audio! */
 297         return ast_frdup(&frame);
 298 }
 299
 300 /*! \brief Reads a frame in from the audiohook structure
 301  * \param audiohook Audiohook structure
 302  * \param samples Number of samples wanted
 303  * \param direction Direction the audio frame came from
 304  * \param format Format of frame remote side wants back
 305  * \return Returns frame on success, NULL on failure
 306  */
 307 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, format_t format)
 308 {
 309         struct ast_frame *read_frame = NULL, *final_frame = NULL;
 310
 311         if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
 312                 return NULL;
 313
 314         /* If they don't want signed linear back out, we'll have to send it through the translation path */
 315         if (format != AST_FORMAT_SLINEAR) {
 316                 /* Rebuild translation path if different format then previously */
 317                 if (audiohook->format != format) {
 318                         if (audiohook->trans_pvt) {
 319                                 ast_translator_free_path(audiohook->trans_pvt);
 320                                 audiohook->trans_pvt = NULL;
 321                         }
 322                         /* 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 */
 323                         if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
 324                                 ast_frfree(read_frame);
 325                                 return NULL;
 326                         }
 327                 }
 328                 /* Convert to requested format, and allow the read in frame to be freed */
 329                 final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
 330         } else {
 331                 final_frame = read_frame;
 332         }
 333
 334         return final_frame;
 335 }
 336
 337 /*! \brief Attach audiohook to channel
 338  * \param chan Channel
 339  * \param audiohook Audiohook structure
 340  * \return Returns 0 on success, -1 on failure
 341  */
 342 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
 343 {
 344         ast_channel_lock(chan);
 345
 346         if (!chan->audiohooks) {
 347                 /* Whoops... allocate a new structure */
 348                 if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
 349                         ast_channel_unlock(chan);
 350                         return -1;
 351                 }
 352                 AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
 353                 AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
 354                 AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
 355         }
 356
 357         /* Drop into respective list */
 358         if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
 359                 AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
 360         else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
 361                 AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
 362         else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
 363                 AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
 364
 365         /* Change status over to running since it is now attached */
 366         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
 367
 368         ast_channel_unlock(chan);
 369
 370         return 0;
 371 }
 372
 373 /*! \brief Update audiohook's status
 374  * \param audiohook Audiohook structure
 375  * \param status Audiohook status enum
 376  *
 377  * \note once status is updated to DONE, this function can not be used to set the
 378  * status back to any other setting.  Setting DONE effectively locks the status as such.
 379  */
 380
 381 void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
 382 {
 383         ast_audiohook_lock(audiohook);
 384         if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
 385                 audiohook->status = status;
 386                 ast_cond_signal(&audiohook->trigger);
 387         }
 388         ast_audiohook_unlock(audiohook);
 389 }
 390
 391 /*! \brief Detach audiohook from channel
 392  * \param audiohook Audiohook structure
 393  * \return Returns 0 on success, -1 on failure
 394  */
 395 int ast_audiohook_detach(struct ast_audiohook *audiohook)
 396 {
 397         if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
 398                 return 0;
 399
 400         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
 401
 402         while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
 403                 ast_audiohook_trigger_wait(audiohook);
 404
 405         return 0;
 406 }
 407
 408 /*! \brief Detach audiohooks from list and destroy said list
 409  * \param audiohook_list List of audiohooks
 410  * \return Returns 0 on success, -1 on failure
 411  */
 412 int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
 413 {
 414         int i = 0;
 415         struct ast_audiohook *audiohook = NULL;
 416
 417         /* Drop any spies */
 418         while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
 419                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 420         }
 421
 422         /* Drop any whispering sources */
 423         while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
 424                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 425         }
 426
 427         /* Drop any manipulaters */
 428         while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
 429                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 430                 audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
 431         }
 432
 433         /* Drop translation paths if present */
 434         for (i = 0; i < 2; i++) {
 435                 if (audiohook_list->in_translate[i].trans_pvt)
 436                         ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
 437                 if (audiohook_list->out_translate[i].trans_pvt)
 438                         ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
 439         }
 440
 441         /* Free ourselves */
 442         ast_free(audiohook_list);
 443
 444         return 0;
 445 }
 446
 447 /*! \brief find an audiohook based on its source
 448  * \param audiohook_list The list of audiohooks to search in
 449  * \param source The source of the audiohook we wish to find
 450  * \return Return the corresponding audiohook or NULL if it cannot be found.
 451  */
 452 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
 453 {
 454         struct ast_audiohook *audiohook = NULL;
 455
 456         AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
 457                 if (!strcasecmp(audiohook->source, source))
 458                         return audiohook;
 459         }
 460
 461         AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
 462                 if (!strcasecmp(audiohook->source, source))
 463                         return audiohook;
 464         }
 465
 466         AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
 467                 if (!strcasecmp(audiohook->source, source))
 468                         return audiohook;
 469         }
 470
 471         return NULL;
 472 }
 473
 474 void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
 475 {
 476         struct ast_audiohook *audiohook;
 477         enum ast_audiohook_status oldstatus;
 478
 479         if (!old_chan->audiohooks || !(audiohook = find_audiohook_by_source(old_chan->audiohooks, source))) {
 480                 return;
 481         }
 482
 483         /* By locking both channels and the audiohook, we can assure that
 484          * another thread will not have a chance to read the audiohook's status
 485          * as done, even though ast_audiohook_remove signals the trigger
 486          * condition.
 487          */
 488         ast_audiohook_lock(audiohook);
 489         oldstatus = audiohook->status;
 490
 491         ast_audiohook_remove(old_chan, audiohook);
 492         ast_audiohook_attach(new_chan, audiohook);
 493
 494         audiohook->status = oldstatus;
 495         ast_audiohook_unlock(audiohook);
 496 }
 497
 498 /*! \brief Detach specified source audiohook from channel
 499  * \param chan Channel to detach from
 500  * \param source Name of source to detach
 501  * \return Returns 0 on success, -1 on failure
 502  */
 503 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
 504 {
 505         struct ast_audiohook *audiohook = NULL;
 506
 507         ast_channel_lock(chan);
 508
 509         /* Ensure the channel has audiohooks on it */
 510         if (!chan->audiohooks) {
 511                 ast_channel_unlock(chan);
 512                 return -1;
 513         }
 514
 515         audiohook = find_audiohook_by_source(chan->audiohooks, source);
 516
 517         ast_channel_unlock(chan);
 518
 519         if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
 520                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
 521
 522         return (audiohook ? 0 : -1);
 523 }
 524
 525 /*!
 526  * \brief Remove an audiohook from a specified channel
 527  *
 528  * \param chan Channel to remove from
 529  * \param audiohook Audiohook to remove
 530  *
 531  * \return Returns 0 on success, -1 on failure
 532  *
 533  * \note The channel does not need to be locked before calling this function
 534  */
 535 int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
 536 {
 537         ast_channel_lock(chan);
 538
 539         if (!chan->audiohooks) {
 540                 ast_channel_unlock(chan);
 541                 return -1;
 542         }
 543
 544         if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
 545                 AST_LIST_REMOVE(&chan->audiohooks->spy_list, audiohook, list);
 546         else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
 547                 AST_LIST_REMOVE(&chan->audiohooks->whisper_list, audiohook, list);
 548         else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
 549                 AST_LIST_REMOVE(&chan->audiohooks->manipulate_list, audiohook, list);
 550
 551         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 552
 553         ast_channel_unlock(chan);
 554
 555         return 0;
 556 }
 557
 558 /*! \brief Pass a DTMF frame off to be handled by the audiohook core
 559  * \param chan Channel that the list is coming off of
 560  * \param audiohook_list List of audiohooks
 561  * \param direction Direction frame is coming in from
 562  * \param frame The frame itself
 563  * \return Return frame on success, NULL on failure
 564  */
 565 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)
 566 {
 567         struct ast_audiohook *audiohook = NULL;
 568
 569         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
 570                 ast_audiohook_lock(audiohook);
 571                 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 572                         AST_LIST_REMOVE_CURRENT(list);
 573                         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 574                         ast_audiohook_unlock(audiohook);
 575                         audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
 576                         continue;
 577                 }
 578                 if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
 579                         audiohook->manipulate_callback(audiohook, chan, frame, direction);
 580                 ast_audiohook_unlock(audiohook);
 581         }
 582         AST_LIST_TRAVERSE_SAFE_END;
 583
 584         return frame;
 585 }
 586
 587 /*!
 588  * \brief Pass an AUDIO frame off to be handled by the audiohook core
 589  *
 590  * \details
 591  * This function has 3 ast_frames and 3 parts to handle each.  At the beginning of this
 592  * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
 593  * input frame.
 594  *
 595  * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
 596  *         format.  The result of this part is middle_frame is guaranteed to be in
 597  *         SLINEAR format for Part_2.
 598  * Part_2: Send middle_frame off to spies and manipulators.  At this point middle_frame is
 599  *         either a new frame as result of the translation, or points directly to the start_frame
 600  *         because no translation to SLINEAR audio was required.  The result of this part
 601  *         is end_frame will be updated to point to middle_frame if any audiohook manipulation
 602  *         took place.
 603  * Part_3: Translate end_frame's audio back into the format of start frame if necessary.
 604  *         At this point if middle_frame != end_frame, we are guaranteed that no manipulation
 605  *         took place and middle_frame can be freed as it was translated... If middle_frame was
 606  *         not translated and still pointed to start_frame, it would be equal to end_frame as well
 607  *         regardless if manipulation took place which would not result in this free.  The result
 608  *         of this part is end_frame is guaranteed to be the format of start_frame for the return.
 609  *
 610  * \param chan Channel that the list is coming off of
 611  * \param audiohook_list List of audiohooks
 612  * \param direction Direction frame is coming in from
 613  * \param frame The frame itself
 614  * \return Return frame on success, NULL on failure
 615  */
 616 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)
 617 {
 618         struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
 619         struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
 620         struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
 621         struct ast_audiohook *audiohook = NULL;
 622         int samples = frame->samples;
 623
 624         /* ---Part_1. translate start_frame to SLINEAR if necessary. */
 625         /* If the frame coming in is not signed linear we have to send it through the in_translate path */
 626         if (frame->subclass.codec != AST_FORMAT_SLINEAR) {
 627                 if (in_translate->format != frame->subclass.codec) {
 628                         if (in_translate->trans_pvt)
 629                                 ast_translator_free_path(in_translate->trans_pvt);
 630                         if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass.codec)))
 631                                 return frame;
 632                         in_translate->format = frame->subclass.codec;
 633                 }
 634                 if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
 635                         return frame;
 636                 samples = middle_frame->samples;
 637         }
 638
 639         /* ---Part_2: Send middle_frame to spy and manipulator lists.  middle_frame is guaranteed to be SLINEAR here.*/
 640         /* Queue up signed linear frame to each spy */
 641         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
 642                 ast_audiohook_lock(audiohook);
 643                 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 644                         AST_LIST_REMOVE_CURRENT(list);
 645                         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 646                         ast_audiohook_unlock(audiohook);
 647                         continue;
 648                 }
 649                 ast_audiohook_write_frame(audiohook, direction, middle_frame);
 650                 ast_audiohook_unlock(audiohook);
 651         }
 652         AST_LIST_TRAVERSE_SAFE_END;
 653
 654         /* If this frame is being written out to the channel then we need to use whisper sources */
 655         if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
 656                 int i = 0;
 657                 short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
 658                 memset(&combine_buf, 0, sizeof(combine_buf));
 659                 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
 660                         ast_audiohook_lock(audiohook);
 661                         if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 662                                 AST_LIST_REMOVE_CURRENT(list);
 663                                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 664                                 ast_audiohook_unlock(audiohook);
 665                                 continue;
 666                         }
 667                         if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
 668                                 /* Take audio from this whisper source and combine it into our main buffer */
 669                                 for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
 670                                         ast_slinear_saturated_add(data1, data2);
 671                         }
 672                         ast_audiohook_unlock(audiohook);
 673                 }
 674                 AST_LIST_TRAVERSE_SAFE_END;
 675                 /* We take all of the combined whisper sources and combine them into the audio being written out */
 676                 for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++)
 677                         ast_slinear_saturated_add(data1, data2);
 678                 end_frame = middle_frame;
 679         }
 680
 681         /* Pass off frame to manipulate audiohooks */
 682         if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
 683                 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
 684                         ast_audiohook_lock(audiohook);
 685                         if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 686                                 AST_LIST_REMOVE_CURRENT(list);
 687                                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 688                                 ast_audiohook_unlock(audiohook);
 689                                 /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
 690                                 audiohook->manipulate_callback(audiohook, chan, NULL, direction);
 691                                 continue;
 692                         }
 693                         /* Feed in frame to manipulation. */
 694                         if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
 695                                 /* XXX IGNORE FAILURE */
 696
 697                                 /* If the manipulation fails then the frame will be returned in its original state.
 698                                  * Since there are potentially more manipulator callbacks in the list, no action should
 699                                  * be taken here to exit early. */
 700                         }
 701                         ast_audiohook_unlock(audiohook);
 702                 }
 703                 AST_LIST_TRAVERSE_SAFE_END;
 704                 end_frame = middle_frame;
 705         }
 706
 707         /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
 708         if (middle_frame == end_frame) {
 709                 /* Middle frame was modified and became the end frame... let's see if we need to transcode */
 710                 if (end_frame->subclass.codec != start_frame->subclass.codec) {
 711                         if (out_translate->format != start_frame->subclass.codec) {
 712                                 if (out_translate->trans_pvt)
 713                                         ast_translator_free_path(out_translate->trans_pvt);
 714                                 if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass.codec, AST_FORMAT_SLINEAR))) {
 715                                         /* We can't transcode this... drop our middle frame and return the original */
 716                                         ast_frfree(middle_frame);
 717                                         return start_frame;
 718                                 }
 719                                 out_translate->format = start_frame->subclass.codec;
 720                         }
 721                         /* Transcode from our middle (signed linear) frame to new format of the frame that came in */
 722                         if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
 723                                 /* Failed to transcode the frame... drop it and return the original */
 724                                 ast_frfree(middle_frame);
 725                                 return start_frame;
 726                         }
 727                         /* Here's the scoop... middle frame is no longer of use to us */
 728                         ast_frfree(middle_frame);
 729                 }
 730         } else {
 731                 /* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
 732                 ast_frfree(middle_frame);
 733         }
 734
 735         return end_frame;
 736 }
 737
 738 int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
 739 {
 740         if (AST_LIST_EMPTY(&audiohook_list->spy_list) &&
 741                 AST_LIST_EMPTY(&audiohook_list->whisper_list) &&
 742                 AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
 743
 744                 return 1;
 745         }
 746         return 0;
 747 }
 748
 749 /*! \brief Pass a frame off to be handled by the audiohook core
 750  * \param chan Channel that the list is coming off of
 751  * \param audiohook_list List of audiohooks
 752  * \param direction Direction frame is coming in from
 753  * \param frame The frame itself
 754  * \return Return frame on success, NULL on failure
 755  */
 756 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)
 757 {
 758         /* Pass off frame to it's respective list write function */
 759         if (frame->frametype == AST_FRAME_VOICE)
 760                 return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
 761         else if (frame->frametype == AST_FRAME_DTMF)
 762                 return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
 763         else
 764                 return frame;
 765 }
 766
 767 /*! \brief Wait for audiohook trigger to be triggered
 768  * \param audiohook Audiohook to wait on
 769  */
 770 void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
 771 {
 772         struct timeval wait;
 773         struct timespec ts;
 774
 775         wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
 776         ts.tv_sec = wait.tv_sec;
 777         ts.tv_nsec = wait.tv_usec * 1000;
 778
 779         ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
 780
 781         return;
 782 }
 783
 784 /* Count number of channel audiohooks by type, regardless of type */
 785 int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
 786 {
 787         int count = 0;
 788         struct ast_audiohook *ah = NULL;
 789
 790         if (!chan->audiohooks)
 791                 return -1;
 792
 793         switch (type) {
 794                 case AST_AUDIOHOOK_TYPE_SPY:
 795                         AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->spy_list, ah, list) {
 796                                 if (!strcmp(ah->source, source)) {
 797                                         count++;
 798                                 }
 799                         }
 800                         AST_LIST_TRAVERSE_SAFE_END;
 801                         break;
 802                 case AST_AUDIOHOOK_TYPE_WHISPER:
 803                         AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->whisper_list, ah, list) {
 804                                 if (!strcmp(ah->source, source)) {
 805                                         count++;
 806                                 }
 807                         }
 808                         AST_LIST_TRAVERSE_SAFE_END;
 809                         break;
 810                 case AST_AUDIOHOOK_TYPE_MANIPULATE:
 811                         AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->manipulate_list, ah, list) {
 812                                 if (!strcmp(ah->source, source)) {
 813                                         count++;
 814                                 }
 815                         }
 816                         AST_LIST_TRAVERSE_SAFE_END;
 817                         break;
 818                 default:
 819                         ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%d)\n", type);
 820                         return -1;
 821         }
 822
 823         return count;
 824 }
 825
 826 /* Count number of channel audiohooks by type that are running */
 827 int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
 828 {
 829         int count = 0;
 830         struct ast_audiohook *ah = NULL;
 831         if (!chan->audiohooks)
 832                 return -1;
 833
 834         switch (type) {
 835                 case AST_AUDIOHOOK_TYPE_SPY:
 836                         AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->spy_list, ah, list) {
 837                                 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
 838                                         count++;
 839                         }
 840                         AST_LIST_TRAVERSE_SAFE_END;
 841                         break;
 842                 case AST_AUDIOHOOK_TYPE_WHISPER:
 843                         AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->whisper_list, ah, list) {
 844                                 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
 845                                         count++;
 846                         }
 847                         AST_LIST_TRAVERSE_SAFE_END;
 848                         break;
 849                 case AST_AUDIOHOOK_TYPE_MANIPULATE:
 850                         AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->manipulate_list, ah, list) {
 851                                 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
 852                                         count++;
 853                         }
 854                         AST_LIST_TRAVERSE_SAFE_END;
 855                         break;
 856                 default:
 857                         ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%d)\n", type);
 858                         return -1;
 859         }
 860         return count;
 861 }
 862
 863 /*! \brief Audiohook volume adjustment structure */
 864 struct audiohook_volume {
 865         struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
 866         int read_adjustment;            /*!< Value to adjust frames read from the channel by */
 867         int write_adjustment;           /*!< Value to adjust frames written to the channel by */
 868 };
 869
 870 /*! \brief Callback used to destroy the audiohook volume datastore
 871  * \param data Volume information structure
 872  * \return Returns nothing
 873  */
 874 static void audiohook_volume_destroy(void *data)
 875 {
 876         struct audiohook_volume *audiohook_volume = data;
 877
 878         /* Destroy the audiohook as it is no longer in use */
 879         ast_audiohook_destroy(&audiohook_volume->audiohook);
 880
 881         /* Finally free ourselves, we are of no more use */
 882         ast_free(audiohook_volume);
 883
 884         return;
 885 }
 886
 887 /*! \brief Datastore used to store audiohook volume information */
 888 static const struct ast_datastore_info audiohook_volume_datastore = {
 889         .type = "Volume",
 890         .destroy = audiohook_volume_destroy,
 891 };
 892
 893 /*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
 894  * \param audiohook Audiohook attached to the channel
 895  * \param chan Channel we are attached to
 896  * \param frame Frame of audio we want to manipulate
 897  * \param direction Direction the audio came in from
 898  * \return Returns 0 on success, -1 on failure
 899  */
 900 static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
 901 {
 902         struct ast_datastore *datastore = NULL;
 903         struct audiohook_volume *audiohook_volume = NULL;
 904         int *gain = NULL;
 905
 906         /* If the audiohook is shutting down don't even bother */
 907         if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
 908                 return 0;
 909         }
 910
 911         /* Try to find the datastore containg adjustment information, if we can't just bail out */
 912         if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
 913                 return 0;
 914         }
 915
 916         audiohook_volume = datastore->data;
 917
 918         /* Based on direction grab the appropriate adjustment value */
 919         if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
 920                 gain = &audiohook_volume->read_adjustment;
 921         } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
 922                 gain = &audiohook_volume->write_adjustment;
 923         }
 924
 925         /* If an adjustment value is present modify the frame */
 926         if (gain && *gain) {
 927                 ast_frame_adjust_volume(frame, *gain);
 928         }
 929
 930         return 0;
 931 }
 932
 933 /*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
 934  * \param chan Channel to look on
 935  * \param create Whether to create the datastore if not found
 936  * \return Returns audiohook_volume structure on success, NULL on failure
 937  */
 938 static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
 939 {
 940         struct ast_datastore *datastore = NULL;
 941         struct audiohook_volume *audiohook_volume = NULL;
 942
 943         /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
 944         if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
 945                 return datastore->data;
 946         }
 947
 948         /* 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 */
 949         if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
 950                 return NULL;
 951         }
 952
 953         /* Create a new audiohook_volume structure to contain our adjustments and audiohook */
 954         if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
 955                 ast_datastore_free(datastore);
 956                 return NULL;
 957         }
 958
 959         /* Setup our audiohook structure so we can manipulate the audio */
 960         ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume");
 961         audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback;
 962
 963         /* Attach the audiohook_volume blob to the datastore and attach to the channel */
 964         datastore->data = audiohook_volume;
 965         ast_channel_datastore_add(chan, datastore);
 966
 967         /* All is well... put the audiohook into motion */
 968         ast_audiohook_attach(chan, &audiohook_volume->audiohook);
 969
 970         return audiohook_volume;
 971 }
 972
 973 /*! \brief Adjust the volume on frames read from or written to a channel
 974  * \param chan Channel to muck with
 975  * \param direction Direction to set on
 976  * \param volume Value to adjust the volume by
 977  * \return Returns 0 on success, -1 on failure
 978  */
 979 int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
 980 {
 981         struct audiohook_volume *audiohook_volume = NULL;
 982
 983         /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
 984         if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
 985                 return -1;
 986         }
 987
 988         /* Now based on the direction set the proper value */
 989         if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
 990                 audiohook_volume->read_adjustment = volume;
 991         }
 992         if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
 993                 audiohook_volume->write_adjustment = volume;
 994         }
 995
 996         return 0;
 997 }
 998
 999 /*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
1000  * \param chan Channel to retrieve volume adjustment from
1001  * \param direction Direction to retrieve
1002  * \return Returns adjustment value
1003  */
1004 int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
1005 {
1006         struct audiohook_volume *audiohook_volume = NULL;
1007         int adjustment = 0;
1008
1009         /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
1010         if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
1011                 return 0;
1012         }
1013
1014         /* Grab the adjustment value based on direction given */
1015         if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
1016                 adjustment = audiohook_volume->read_adjustment;
1017         } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
1018                 adjustment = audiohook_volume->write_adjustment;
1019         }
1020
1021         return adjustment;
1022 }
1023
1024 /*! \brief Adjust the volume on frames read from or written to a channel
1025  * \param chan Channel to muck with
1026  * \param direction Direction to increase
1027  * \param volume Value to adjust the adjustment by
1028  * \return Returns 0 on success, -1 on failure
1029  */
1030 int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
1031 {
1032         struct audiohook_volume *audiohook_volume = NULL;
1033
1034         /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
1035         if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
1036                 return -1;
1037         }
1038
1039         /* Based on the direction change the specific adjustment value */
1040         if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1041                 audiohook_volume->read_adjustment += volume;
1042         }
1043         if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1044                 audiohook_volume->write_adjustment += volume;
1045         }
1046
1047         return 0;
1048 }
1049
1050 /*! \brief Mute frames read from or written to a channel
1051  * \param chan Channel to muck with
1052  * \param source Type of audiohook
1053  * \param flag which flag to set / clear
1054  * \param clear set or clear
1055  * \return Returns 0 on success, -1 on failure
1056  */
1057 int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
1058 {
1059         struct ast_audiohook *audiohook = NULL;
1060
1061         ast_channel_lock(chan);
1062
1063         /* Ensure the channel has audiohooks on it */
1064         if (!chan->audiohooks) {
1065                 ast_channel_unlock(chan);
1066                 return -1;
1067         }
1068
1069         audiohook = find_audiohook_by_source(chan->audiohooks, source);
1070
1071         if (audiohook) {
1072                 if (clear) {
1073                         ast_clear_flag(audiohook, flag);
1074                 } else {
1075                         ast_set_flag(audiohook, flag);
1076                 }
1077         }
1078
1079         ast_channel_unlock(chan);
1080
1081         return (audiohook ? 0 : -1);
1082 }