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         int samples_converted;
 364
 365         /* the number of samples requested is based on the format they are requesting.  Inorder
 366          * to process this correctly samples must be converted to our internal sample rate */
 367         if (audiohook->hook_internal_samp_rate == ast_format_get_sample_rate(format)) {
 368                 samples_converted = samples;
 369         } else if (audiohook->hook_internal_samp_rate > ast_format_get_sample_rate(format)) {
 370                 samples_converted = samples * (audiohook->hook_internal_samp_rate / (float) ast_format_get_sample_rate(format));
 371         } else {
 372                 samples_converted = samples * (ast_format_get_sample_rate(format) / (float) audiohook->hook_internal_samp_rate);
 373         }
 374
 375         if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
 376                 audiohook_read_frame_both(audiohook, samples_converted, read_reference, write_reference) :
 377                 audiohook_read_frame_single(audiohook, samples_converted, direction)))) {
 378                 return NULL;
 379         }
 380
 381         slin = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
 382
 383         /* If they don't want signed linear back out, we'll have to send it through the translation path */
 384         if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) {
 385                 /* Rebuild translation path if different format then previously */
 386                 if (ast_format_cmp(format, audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) {
 387                         if (audiohook->trans_pvt) {
 388                                 ast_translator_free_path(audiohook->trans_pvt);
 389                                 audiohook->trans_pvt = NULL;
 390                         }
 391
 392                         /* 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 */
 393                         if (!(audiohook->trans_pvt = ast_translator_build_path(format, slin))) {
 394                                 ast_frfree(read_frame);
 395                                 return NULL;
 396                         }
 397                         ao2_replace(audiohook->format, format);
 398                 }
 399                 /* Convert to requested format, and allow the read in frame to be freed */
 400                 final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
 401         } else {
 402                 final_frame = read_frame;
 403         }
 404
 405         return final_frame;
 406 }
 407
 408 /*! \brief Reads a frame in from the audiohook structure
 409  * \param audiohook Audiohook structure
 410  * \param samples Number of samples wanted in requested output format
 411  * \param direction Direction the audio frame came from
 412  * \param format Format of frame remote side wants back
 413  * \return Returns frame on success, NULL on failure
 414  */
 415 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format)
 416 {
 417         return audiohook_read_frame_helper(audiohook, samples, direction, format, NULL, NULL);
 418 }
 419
 420 /*! \brief Reads a frame in from the audiohook structure
 421  * \param audiohook Audiohook structure
 422  * \param samples Number of samples wanted
 423  * \param direction Direction the audio frame came from
 424  * \param format Format of frame remote side wants back
 425  * \param read_frame frame pointer for copying read frame data
 426  * \param write_frame frame pointer for copying write frame data
 427  * \return Returns frame on success, NULL on failure
 428  */
 429 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)
 430 {
 431         return audiohook_read_frame_helper(audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, format, read_frame, write_frame);
 432 }
 433
 434 static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list)
 435 {
 436         struct ast_audiohook *ah = NULL;
 437         audiohook_list->native_slin_compatible = 1;
 438         AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, ah, list) {
 439                 if (!(ah->init_flags & AST_AUDIOHOOK_MANIPULATE_ALL_RATES)) {
 440                         audiohook_list->native_slin_compatible = 0;
 441                         return;
 442                 }
 443         }
 444 }
 445
 446 /*! \brief Attach audiohook to channel
 447  * \param chan Channel
 448  * \param audiohook Audiohook structure
 449  * \return Returns 0 on success, -1 on failure
 450  */
 451 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
 452 {
 453         ast_channel_lock(chan);
 454
 455         if (!ast_channel_audiohooks(chan)) {
 456                 struct ast_audiohook_list *ahlist;
 457                 /* Whoops... allocate a new structure */
 458                 if (!(ahlist = ast_calloc(1, sizeof(*ahlist)))) {
 459                         ast_channel_unlock(chan);
 460                         return -1;
 461                 }
 462                 ast_channel_audiohooks_set(chan, ahlist);
 463                 AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->spy_list);
 464                 AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->whisper_list);
 465                 AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->manipulate_list);
 466                 /* This sample rate will adjust as necessary when writing to the list. */
 467                 ast_channel_audiohooks(chan)->list_internal_samp_rate = 8000;
 468         }
 469
 470         /* Drop into respective list */
 471         if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
 472                 AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
 473         } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
 474                 AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
 475         } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
 476                 AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
 477         }
 478
 479
 480         audiohook_set_internal_rate(audiohook, ast_channel_audiohooks(chan)->list_internal_samp_rate, 1);
 481         audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
 482
 483         /* Change status over to running since it is now attached */
 484         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
 485
 486         if (ast_channel_is_bridged(chan)) {
 487                 ast_channel_set_unbridged_nolock(chan, 1);
 488         }
 489
 490         ast_channel_unlock(chan);
 491
 492         return 0;
 493 }
 494
 495 /*! \brief Update audiohook's status
 496  * \param audiohook Audiohook structure
 497  * \param status Audiohook status enum
 498  *
 499  * \note once status is updated to DONE, this function can not be used to set the
 500  * status back to any other setting.  Setting DONE effectively locks the status as such.
 501  */
 502
 503 void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
 504 {
 505         ast_audiohook_lock(audiohook);
 506         if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
 507                 audiohook->status = status;
 508                 ast_cond_signal(&audiohook->trigger);
 509         }
 510         ast_audiohook_unlock(audiohook);
 511 }
 512
 513 /*! \brief Detach audiohook from channel
 514  * \param audiohook Audiohook structure
 515  * \return Returns 0 on success, -1 on failure
 516  */
 517 int ast_audiohook_detach(struct ast_audiohook *audiohook)
 518 {
 519         if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
 520                 return 0;
 521         }
 522
 523         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
 524
 525         while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
 526                 ast_audiohook_trigger_wait(audiohook);
 527         }
 528
 529         return 0;
 530 }
 531
 532 void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
 533 {
 534         int i;
 535         struct ast_audiohook *audiohook;
 536
 537         if (!audiohook_list) {
 538                 return;
 539         }
 540
 541         /* Drop any spies */
 542         while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
 543                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 544         }
 545
 546         /* Drop any whispering sources */
 547         while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
 548                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 549         }
 550
 551         /* Drop any manipulaters */
 552         while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
 553                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 554                 audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
 555         }
 556
 557         /* Drop translation paths if present */
 558         for (i = 0; i < 2; i++) {
 559                 if (audiohook_list->in_translate[i].trans_pvt) {
 560                         ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
 561                 }
 562                 if (audiohook_list->out_translate[i].trans_pvt) {
 563                         ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
 564                 }
 565         }
 566
 567         /* Free ourselves */
 568         ast_free(audiohook_list);
 569 }
 570
 571 /*! \brief find an audiohook based on its source
 572  * \param audiohook_list The list of audiohooks to search in
 573  * \param source The source of the audiohook we wish to find
 574  * \return Return the corresponding audiohook or NULL if it cannot be found.
 575  */
 576 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
 577 {
 578         struct ast_audiohook *audiohook = NULL;
 579
 580         AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
 581                 if (!strcasecmp(audiohook->source, source)) {
 582                         return audiohook;
 583                 }
 584         }
 585
 586         AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
 587                 if (!strcasecmp(audiohook->source, source)) {
 588                         return audiohook;
 589                 }
 590         }
 591
 592         AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
 593                 if (!strcasecmp(audiohook->source, source)) {
 594                         return audiohook;
 595                 }
 596         }
 597
 598         return NULL;
 599 }
 600
 601 static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook)
 602 {
 603         enum ast_audiohook_status oldstatus;
 604
 605         /* By locking both channels and the audiohook, we can assure that
 606          * another thread will not have a chance to read the audiohook's status
 607          * as done, even though ast_audiohook_remove signals the trigger
 608          * condition.
 609          */
 610         ast_audiohook_lock(audiohook);
 611         oldstatus = audiohook->status;
 612
 613         ast_audiohook_remove(old_chan, audiohook);
 614         ast_audiohook_attach(new_chan, audiohook);
 615
 616         audiohook->status = oldstatus;
 617         ast_audiohook_unlock(audiohook);
 618 }
 619
 620 void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
 621 {
 622         struct ast_audiohook *audiohook;
 623
 624         if (!ast_channel_audiohooks(old_chan)) {
 625                 return;
 626         }
 627
 628         audiohook = find_audiohook_by_source(ast_channel_audiohooks(old_chan), source);
 629         if (!audiohook) {
 630                 return;
 631         }
 632
 633         audiohook_move(old_chan, new_chan, audiohook);
 634 }
 635
 636 void ast_audiohook_move_all(struct ast_channel *old_chan, struct ast_channel *new_chan)
 637 {
 638         struct ast_audiohook *audiohook;
 639         struct ast_audiohook_list *audiohook_list;
 640
 641         audiohook_list = ast_channel_audiohooks(old_chan);
 642         if (!audiohook_list) {
 643                 return;
 644         }
 645
 646         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
 647                 audiohook_move(old_chan, new_chan, audiohook);
 648         }
 649         AST_LIST_TRAVERSE_SAFE_END;
 650
 651         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
 652                 audiohook_move(old_chan, new_chan, audiohook);
 653         }
 654         AST_LIST_TRAVERSE_SAFE_END;
 655
 656         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
 657                 audiohook_move(old_chan, new_chan, audiohook);
 658         }
 659         AST_LIST_TRAVERSE_SAFE_END;
 660 }
 661
 662 /*! \brief Detach specified source audiohook from channel
 663  * \param chan Channel to detach from
 664  * \param source Name of source to detach
 665  * \return Returns 0 on success, -1 on failure
 666  */
 667 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
 668 {
 669         struct ast_audiohook *audiohook = NULL;
 670
 671         ast_channel_lock(chan);
 672
 673         /* Ensure the channel has audiohooks on it */
 674         if (!ast_channel_audiohooks(chan)) {
 675                 ast_channel_unlock(chan);
 676                 return -1;
 677         }
 678
 679         audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
 680
 681         ast_channel_unlock(chan);
 682
 683         if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
 684                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
 685         }
 686
 687         return (audiohook ? 0 : -1);
 688 }
 689
 690 /*!
 691  * \brief Remove an audiohook from a specified channel
 692  *
 693  * \param chan Channel to remove from
 694  * \param audiohook Audiohook to remove
 695  *
 696  * \return Returns 0 on success, -1 on failure
 697  *
 698  * \note The channel does not need to be locked before calling this function
 699  */
 700 int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
 701 {
 702         ast_channel_lock(chan);
 703
 704         if (!ast_channel_audiohooks(chan)) {
 705                 ast_channel_unlock(chan);
 706                 return -1;
 707         }
 708
 709         if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
 710                 AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
 711         } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
 712                 AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
 713         } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
 714                 AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
 715         }
 716
 717         audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
 718         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 719
 720         if (ast_channel_is_bridged(chan)) {
 721                 ast_channel_set_unbridged_nolock(chan, 1);
 722         }
 723
 724         ast_channel_unlock(chan);
 725
 726         return 0;
 727 }
 728
 729 /*! \brief Pass a DTMF frame off to be handled by the audiohook core
 730  * \param chan Channel that the list is coming off of
 731  * \param audiohook_list List of audiohooks
 732  * \param direction Direction frame is coming in from
 733  * \param frame The frame itself
 734  * \return Return frame on success, NULL on failure
 735  */
 736 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)
 737 {
 738         struct ast_audiohook *audiohook = NULL;
 739         int removed = 0;
 740
 741         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
 742                 ast_audiohook_lock(audiohook);
 743                 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 744                         AST_LIST_REMOVE_CURRENT(list);
 745                         removed = 1;
 746                         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 747                         ast_audiohook_unlock(audiohook);
 748                         audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
 749                         if (ast_channel_is_bridged(chan)) {
 750                                 ast_channel_set_unbridged_nolock(chan, 1);
 751                         }
 752                         continue;
 753                 }
 754                 if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF)) {
 755                         audiohook->manipulate_callback(audiohook, chan, frame, direction);
 756                 }
 757                 ast_audiohook_unlock(audiohook);
 758         }
 759         AST_LIST_TRAVERSE_SAFE_END;
 760
 761         /* if an audiohook got removed, reset samplerate compatibility */
 762         if (removed) {
 763                 audiohook_list_set_samplerate_compatibility(audiohook_list);
 764         }
 765         return frame;
 766 }
 767
 768 static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_list *audiohook_list,
 769         enum ast_audiohook_direction direction, struct ast_frame *frame)
 770 {
 771         struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ?
 772                 &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
 773         struct ast_frame *new_frame = frame;
 774         struct ast_format *slin;
 775
 776         /* If we are capable of maintaining doing samplerates other that 8khz, update
 777          * the internal audiohook_list's rate and higher samplerate audio arrives. By
 778          * updating the list's rate, all the audiohooks in the list will be updated as well
 779          * as the are written and read from. */
 780         if (audiohook_list->native_slin_compatible) {
 781                 audiohook_list->list_internal_samp_rate =
 782                         MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate);
 783         }
 784
 785         slin = ast_format_cache_get_slin_by_rate(audiohook_list->list_internal_samp_rate);
 786         if (ast_format_cmp(frame->subclass.format, slin) == AST_FORMAT_CMP_EQUAL) {
 787                 return new_frame;
 788         }
 789
 790         if (ast_format_cmp(frame->subclass.format, in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) {
 791                 if (in_translate->trans_pvt) {
 792                         ast_translator_free_path(in_translate->trans_pvt);
 793                 }
 794                 if (!(in_translate->trans_pvt = ast_translator_build_path(slin, frame->subclass.format))) {
 795                         return NULL;
 796                 }
 797                 ao2_replace(in_translate->format, frame->subclass.format);
 798         }
 799
 800         if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) {
 801                 return NULL;
 802         }
 803
 804         return new_frame;
 805 }
 806
 807 static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook_list *audiohook_list,
 808         enum ast_audiohook_direction direction, struct ast_frame *slin_frame, struct ast_format *outformat)
 809 {
 810         struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
 811         struct ast_frame *outframe = NULL;
 812         if (ast_format_cmp(slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
 813                 /* rebuild translators if necessary */
 814                 if (ast_format_cmp(out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
 815                         if (out_translate->trans_pvt) {
 816                                 ast_translator_free_path(out_translate->trans_pvt);
 817                         }
 818                         if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) {
 819                                 return NULL;
 820                         }
 821                         ao2_replace(out_translate->format, outformat);
 822                 }
 823                 /* translate back to the format the frame came in as. */
 824                 if (!(outframe = ast_translate(out_translate->trans_pvt, slin_frame, 0))) {
 825                         return NULL;
 826                 }
 827         }
 828         return outframe;
 829 }
 830
 831 /*!
 832  * \brief Pass an AUDIO frame off to be handled by the audiohook core
 833  *
 834  * \details
 835  * This function has 3 ast_frames and 3 parts to handle each.  At the beginning of this
 836  * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
 837  * input frame.
 838  *
 839  * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
 840  *         format.  The result of this part is middle_frame is guaranteed to be in
 841  *         SLINEAR format for Part_2.
 842  * Part_2: Send middle_frame off to spies and manipulators.  At this point middle_frame is
 843  *         either a new frame as result of the translation, or points directly to the start_frame
 844  *         because no translation to SLINEAR audio was required.
 845  * Part_3: Translate end_frame's audio back into the format of start frame if necessary.  This
 846  *         is only necessary if manipulation of middle_frame occurred.
 847  *
 848  * \param chan Channel that the list is coming off of
 849  * \param audiohook_list List of audiohooks
 850  * \param direction Direction frame is coming in from
 851  * \param frame The frame itself
 852  * \return Return frame on success, NULL on failure
 853  */
 854 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)
 855 {
 856         struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
 857         struct ast_audiohook *audiohook = NULL;
 858         int samples;
 859         int middle_frame_manipulated = 0;
 860         int removed = 0;
 861
 862         /* ---Part_1. translate start_frame to SLINEAR if necessary. */
 863         if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
 864                 return frame;
 865         }
 866         samples = middle_frame->samples;
 867
 868         /* ---Part_2: Send middle_frame to spy and manipulator lists.  middle_frame is guaranteed to be SLINEAR here.*/
 869         /* Queue up signed linear frame to each spy */
 870         AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
 871                 ast_audiohook_lock(audiohook);
 872                 if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 873                         AST_LIST_REMOVE_CURRENT(list);
 874                         removed = 1;
 875                         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 876                         ast_audiohook_unlock(audiohook);
 877                         if (ast_channel_is_bridged(chan)) {
 878                                 ast_channel_set_unbridged_nolock(chan, 1);
 879                         }
 880                         continue;
 881                 }
 882                 audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
 883                 ast_audiohook_write_frame(audiohook, direction, middle_frame);
 884                 ast_audiohook_unlock(audiohook);
 885         }
 886         AST_LIST_TRAVERSE_SAFE_END;
 887
 888         /* If this frame is being written out to the channel then we need to use whisper sources */
 889         if (!AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
 890                 int i = 0;
 891                 short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
 892                 memset(&combine_buf, 0, sizeof(combine_buf));
 893                 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
 894                         struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
 895                         ast_audiohook_lock(audiohook);
 896                         if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 897                                 AST_LIST_REMOVE_CURRENT(list);
 898                                 removed = 1;
 899                                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 900                                 ast_audiohook_unlock(audiohook);
 901                                 if (ast_channel_is_bridged(chan)) {
 902                                         ast_channel_set_unbridged_nolock(chan, 1);
 903                                 }
 904                                 continue;
 905                         }
 906                         audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
 907                         if (ast_slinfactory_available(factory) >= samples && ast_slinfactory_read(factory, read_buf, samples)) {
 908                                 /* Take audio from this whisper source and combine it into our main buffer */
 909                                 for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++) {
 910                                         ast_slinear_saturated_add(data1, data2);
 911                                 }
 912                         }
 913                         ast_audiohook_unlock(audiohook);
 914                 }
 915                 AST_LIST_TRAVERSE_SAFE_END;
 916                 /* We take all of the combined whisper sources and combine them into the audio being written out */
 917                 for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++) {
 918                         ast_slinear_saturated_add(data1, data2);
 919                 }
 920                 middle_frame_manipulated = 1;
 921         }
 922
 923         /* Pass off frame to manipulate audiohooks */
 924         if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
 925                 AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
 926                         ast_audiohook_lock(audiohook);
 927                         if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 928                                 AST_LIST_REMOVE_CURRENT(list);
 929                                 removed = 1;
 930                                 ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
 931                                 ast_audiohook_unlock(audiohook);
 932                                 /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
 933                                 audiohook->manipulate_callback(audiohook, chan, NULL, direction);
 934                                 if (ast_channel_is_bridged(chan)) {
 935                                         ast_channel_set_unbridged_nolock(chan, 1);
 936                                 }
 937                                 continue;
 938                         }
 939                         audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
 940                         /* Feed in frame to manipulation. */
 941                         if (!audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
 942                                 /* If the manipulation fails then the frame will be returned in its original state.
 943                                  * Since there are potentially more manipulator callbacks in the list, no action should
 944                                  * be taken here to exit early. */
 945                                  middle_frame_manipulated = 1;
 946                         }
 947                         ast_audiohook_unlock(audiohook);
 948                 }
 949                 AST_LIST_TRAVERSE_SAFE_END;
 950         }
 951
 952         /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
 953         if (middle_frame_manipulated) {
 954                 if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, start_frame->subclass.format))) {
 955                         /* translation failed, so just pass back the input frame */
 956                         end_frame = start_frame;
 957                 }
 958         } else {
 959                 end_frame = start_frame;
 960         }
 961         /* clean up our middle_frame if required */
 962         if (middle_frame != end_frame) {
 963                 ast_frfree(middle_frame);
 964                 middle_frame = NULL;
 965         }
 966
 967         /* Before returning, if an audiohook got removed, reset samplerate compatibility */
 968         if (removed) {
 969                 audiohook_list_set_samplerate_compatibility(audiohook_list);
 970         }
 971
 972         return end_frame;
 973 }
 974
 975 int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
 976 {
 977         return !audiohook_list
 978                 || (AST_LIST_EMPTY(&audiohook_list->spy_list)
 979                         && AST_LIST_EMPTY(&audiohook_list->whisper_list)
 980                         && AST_LIST_EMPTY(&audiohook_list->manipulate_list));
 981 }
 982
 983 /*! \brief Pass a frame off to be handled by the audiohook core
 984  * \param chan Channel that the list is coming off of
 985  * \param audiohook_list List of audiohooks
 986  * \param direction Direction frame is coming in from
 987  * \param frame The frame itself
 988  * \return Return frame on success, NULL on failure
 989  */
 990 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)
 991 {
 992         /* Pass off frame to it's respective list write function */
 993         if (frame->frametype == AST_FRAME_VOICE) {
 994                 return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
 995         } else if (frame->frametype == AST_FRAME_DTMF) {
 996                 return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
 997         } else {
 998                 return frame;
 999         }
1000 }
1001
1002 /*! \brief Wait for audiohook trigger to be triggered
1003  * \param audiohook Audiohook to wait on
1004  */
1005 void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
1006 {
1007         struct timeval wait;
1008         struct timespec ts;
1009
1010         wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
1011         ts.tv_sec = wait.tv_sec;
1012         ts.tv_nsec = wait.tv_usec * 1000;
1013
1014         ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
1015
1016         return;
1017 }
1018
1019 /* Count number of channel audiohooks by type, regardless of type */
1020 int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
1021 {
1022         int count = 0;
1023         struct ast_audiohook *ah = NULL;
1024
1025         if (!ast_channel_audiohooks(chan)) {
1026                 return -1;
1027         }
1028
1029         switch (type) {
1030                 case AST_AUDIOHOOK_TYPE_SPY:
1031                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
1032                                 if (!strcmp(ah->source, source)) {
1033                                         count++;
1034                                 }
1035                         }
1036                         break;
1037                 case AST_AUDIOHOOK_TYPE_WHISPER:
1038                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
1039                                 if (!strcmp(ah->source, source)) {
1040                                         count++;
1041                                 }
1042                         }
1043                         break;
1044                 case AST_AUDIOHOOK_TYPE_MANIPULATE:
1045                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
1046                                 if (!strcmp(ah->source, source)) {
1047                                         count++;
1048                                 }
1049                         }
1050                         break;
1051                 default:
1052                         ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
1053                         return -1;
1054         }
1055
1056         return count;
1057 }
1058
1059 /* Count number of channel audiohooks by type that are running */
1060 int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
1061 {
1062         int count = 0;
1063         struct ast_audiohook *ah = NULL;
1064         if (!ast_channel_audiohooks(chan))
1065                 return -1;
1066
1067         switch (type) {
1068                 case AST_AUDIOHOOK_TYPE_SPY:
1069                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
1070                                 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1071                                         count++;
1072                         }
1073                         break;
1074                 case AST_AUDIOHOOK_TYPE_WHISPER:
1075                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
1076                                 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1077                                         count++;
1078                         }
1079                         break;
1080                 case AST_AUDIOHOOK_TYPE_MANIPULATE:
1081                         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
1082                                 if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
1083                                         count++;
1084                         }
1085                         break;
1086                 default:
1087                         ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
1088                         return -1;
1089         }
1090         return count;
1091 }
1092
1093 /*! \brief Audiohook volume adjustment structure */
1094 struct audiohook_volume {
1095         struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
1096         int read_adjustment;            /*!< Value to adjust frames read from the channel by */
1097         int write_adjustment;           /*!< Value to adjust frames written to the channel by */
1098 };
1099
1100 /*! \brief Callback used to destroy the audiohook volume datastore
1101  * \param data Volume information structure
1102  * \return Returns nothing
1103  */
1104 static void audiohook_volume_destroy(void *data)
1105 {
1106         struct audiohook_volume *audiohook_volume = data;
1107
1108         /* Destroy the audiohook as it is no longer in use */
1109         ast_audiohook_destroy(&audiohook_volume->audiohook);
1110
1111         /* Finally free ourselves, we are of no more use */
1112         ast_free(audiohook_volume);
1113
1114         return;
1115 }
1116
1117 /*! \brief Datastore used to store audiohook volume information */
1118 static const struct ast_datastore_info audiohook_volume_datastore = {
1119         .type = "Volume",
1120         .destroy = audiohook_volume_destroy,
1121 };
1122
1123 /*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
1124  * \param audiohook Audiohook attached to the channel
1125  * \param chan Channel we are attached to
1126  * \param frame Frame of audio we want to manipulate
1127  * \param direction Direction the audio came in from
1128  * \return Returns 0 on success, -1 on failure
1129  */
1130 static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
1131 {
1132         struct ast_datastore *datastore = NULL;
1133         struct audiohook_volume *audiohook_volume = NULL;
1134         int *gain = NULL;
1135
1136         /* If the audiohook is shutting down don't even bother */
1137         if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
1138                 return 0;
1139         }
1140
1141         /* Try to find the datastore containg adjustment information, if we can't just bail out */
1142         if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
1143                 return 0;
1144         }
1145
1146         audiohook_volume = datastore->data;
1147
1148         /* Based on direction grab the appropriate adjustment value */
1149         if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
1150                 gain = &audiohook_volume->read_adjustment;
1151         } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
1152                 gain = &audiohook_volume->write_adjustment;
1153         }
1154
1155         /* If an adjustment value is present modify the frame */
1156         if (gain && *gain) {
1157                 ast_frame_adjust_volume(frame, *gain);
1158         }
1159
1160         return 0;
1161 }
1162
1163 /*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
1164  * \param chan Channel to look on
1165  * \param create Whether to create the datastore if not found
1166  * \return Returns audiohook_volume structure on success, NULL on failure
1167  */
1168 static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
1169 {
1170         struct ast_datastore *datastore = NULL;
1171         struct audiohook_volume *audiohook_volume = NULL;
1172
1173         /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
1174         if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
1175                 return datastore->data;
1176         }
1177
1178         /* 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 */
1179         if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
1180                 return NULL;
1181         }
1182
1183         /* Create a new audiohook_volume structure to contain our adjustments and audiohook */
1184         if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
1185                 ast_datastore_free(datastore);
1186                 return NULL;
1187         }
1188
1189         /* Setup our audiohook structure so we can manipulate the audio */
1190         ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
1191         audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback;
1192
1193         /* Attach the audiohook_volume blob to the datastore and attach to the channel */
1194         datastore->data = audiohook_volume;
1195         ast_channel_datastore_add(chan, datastore);
1196
1197         /* All is well... put the audiohook into motion */
1198         ast_audiohook_attach(chan, &audiohook_volume->audiohook);
1199
1200         return audiohook_volume;
1201 }
1202
1203 /*! \brief Adjust the volume on frames read from or written to a channel
1204  * \param chan Channel to muck with
1205  * \param direction Direction to set on
1206  * \param volume Value to adjust the volume by
1207  * \return Returns 0 on success, -1 on failure
1208  */
1209 int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
1210 {
1211         struct audiohook_volume *audiohook_volume = NULL;
1212
1213         /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
1214         if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
1215                 return -1;
1216         }
1217
1218         /* Now based on the direction set the proper value */
1219         if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1220                 audiohook_volume->read_adjustment = volume;
1221         }
1222         if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1223                 audiohook_volume->write_adjustment = volume;
1224         }
1225
1226         return 0;
1227 }
1228
1229 /*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
1230  * \param chan Channel to retrieve volume adjustment from
1231  * \param direction Direction to retrieve
1232  * \return Returns adjustment value
1233  */
1234 int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
1235 {
1236         struct audiohook_volume *audiohook_volume = NULL;
1237         int adjustment = 0;
1238
1239         /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
1240         if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
1241                 return 0;
1242         }
1243
1244         /* Grab the adjustment value based on direction given */
1245         if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
1246                 adjustment = audiohook_volume->read_adjustment;
1247         } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
1248                 adjustment = audiohook_volume->write_adjustment;
1249         }
1250
1251         return adjustment;
1252 }
1253
1254 /*! \brief Adjust the volume on frames read from or written to a channel
1255  * \param chan Channel to muck with
1256  * \param direction Direction to increase
1257  * \param volume Value to adjust the adjustment by
1258  * \return Returns 0 on success, -1 on failure
1259  */
1260 int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
1261 {
1262         struct audiohook_volume *audiohook_volume = NULL;
1263
1264         /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
1265         if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
1266                 return -1;
1267         }
1268
1269         /* Based on the direction change the specific adjustment value */
1270         if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1271                 audiohook_volume->read_adjustment += volume;
1272         }
1273         if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
1274                 audiohook_volume->write_adjustment += volume;
1275         }
1276
1277         return 0;
1278 }
1279
1280 /*! \brief Mute frames read from or written to a channel
1281  * \param chan Channel to muck with
1282  * \param source Type of audiohook
1283  * \param flag which flag to set / clear
1284  * \param clear set or clear
1285  * \return Returns 0 on success, -1 on failure
1286  */
1287 int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
1288 {
1289         struct ast_audiohook *audiohook = NULL;
1290
1291         ast_channel_lock(chan);
1292
1293         /* Ensure the channel has audiohooks on it */
1294         if (!ast_channel_audiohooks(chan)) {
1295                 ast_channel_unlock(chan);
1296                 return -1;
1297         }
1298
1299         audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
1300
1301         if (audiohook) {
1302                 if (clear) {
1303                         ast_clear_flag(audiohook, flag);
1304                 } else {
1305                         ast_set_flag(audiohook, flag);
1306                 }
1307         }
1308
1309         ast_channel_unlock(chan);
1310
1311         return (audiohook ? 0 : -1);
1312 }