Thu Sep 7 01:02:54 2017

Asterisk developer's documentation


audiohook.c

Go to the documentation of this file.
00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 1999 - 2007, Digium, Inc.
00005  *
00006  * Joshua Colp <jcolp@digium.com>
00007  *
00008  * See http://www.asterisk.org for more information about
00009  * the Asterisk project. Please do not directly contact
00010  * any of the maintainers of this project for assistance;
00011  * the project provides a web site, mailing lists and IRC
00012  * channels for your use.
00013  *
00014  * This program is free software, distributed under the terms of
00015  * the GNU General Public License Version 2. See the LICENSE file
00016  * at the top of the source tree.
00017  */
00018 
00019 /*! \file
00020  *
00021  * \brief Audiohooks Architecture
00022  *
00023  * \author Joshua Colp <jcolp@digium.com>
00024  */
00025 
00026 /*** MODULEINFO
00027    <support_level>core</support_level>
00028  ***/
00029 
00030 #include "asterisk.h"
00031 
00032 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 413586 $")
00033 
00034 #include <signal.h>
00035 
00036 #include "asterisk/channel.h"
00037 #include "asterisk/utils.h"
00038 #include "asterisk/lock.h"
00039 #include "asterisk/linkedlists.h"
00040 #include "asterisk/audiohook.h"
00041 #include "asterisk/slinfactory.h"
00042 #include "asterisk/frame.h"
00043 #include "asterisk/translate.h"
00044 
00045 struct ast_audiohook_translate {
00046    struct ast_trans_pvt *trans_pvt;
00047    format_t format;
00048 };
00049 
00050 struct ast_audiohook_list {
00051    struct ast_audiohook_translate in_translate[2];
00052    struct ast_audiohook_translate out_translate[2];
00053    AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
00054    AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
00055    AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
00056 };
00057 
00058 /*! \brief Initialize an audiohook structure
00059  * \param audiohook Audiohook structure
00060  * \param type
00061  * \param source
00062  * \return Returns 0 on success, -1 on failure
00063  */
00064 int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source)
00065 {
00066    /* Need to keep the type and source */
00067    audiohook->type = type;
00068    audiohook->source = source;
00069 
00070    /* Initialize lock that protects our audiohook */
00071    ast_mutex_init(&audiohook->lock);
00072    ast_cond_init(&audiohook->trigger, NULL);
00073 
00074    /* Setup the factories that are needed for this audiohook type */
00075    switch (type) {
00076    case AST_AUDIOHOOK_TYPE_SPY:
00077       ast_slinfactory_init(&audiohook->read_factory);
00078       /* Fall through intentionally */
00079    case AST_AUDIOHOOK_TYPE_WHISPER:
00080       ast_slinfactory_init(&audiohook->write_factory);
00081       break;
00082    default:
00083       break;
00084    }
00085 
00086    /* Since we are just starting out... this audiohook is new */
00087    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
00088 
00089    return 0;
00090 }
00091 
00092 /*! \brief Destroys an audiohook structure
00093  * \param audiohook Audiohook structure
00094  * \return Returns 0 on success, -1 on failure
00095  */
00096 int ast_audiohook_destroy(struct ast_audiohook *audiohook)
00097 {
00098    /* Drop the factories used by this audiohook type */
00099    switch (audiohook->type) {
00100    case AST_AUDIOHOOK_TYPE_SPY:
00101       ast_slinfactory_destroy(&audiohook->read_factory);
00102       /* Fall through intentionally */
00103    case AST_AUDIOHOOK_TYPE_WHISPER:
00104       ast_slinfactory_destroy(&audiohook->write_factory);
00105       break;
00106    default:
00107       break;
00108    }
00109 
00110    /* Destroy translation path if present */
00111    if (audiohook->trans_pvt)
00112       ast_translator_free_path(audiohook->trans_pvt);
00113 
00114    /* Lock and trigger be gone! */
00115    ast_cond_destroy(&audiohook->trigger);
00116    ast_mutex_destroy(&audiohook->lock);
00117 
00118    return 0;
00119 }
00120 
00121 /*! \brief Writes a frame into the audiohook structure
00122  * \param audiohook Audiohook structure
00123  * \param direction Direction the audio frame came from
00124  * \param frame Frame to write in
00125  * \return Returns 0 on success, -1 on failure
00126  */
00127 int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
00128 {
00129    struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
00130    struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
00131    struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime;
00132    int our_factory_samples;
00133    int our_factory_ms;
00134    int other_factory_samples;
00135    int other_factory_ms;
00136    int muteme = 0;
00137 
00138    /* Update last feeding time to be current */
00139    *rwtime = ast_tvnow();
00140 
00141    our_factory_samples = ast_slinfactory_available(factory);
00142    our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / 8);
00143    other_factory_samples = ast_slinfactory_available(other_factory);
00144    other_factory_ms = other_factory_samples / 8;
00145 
00146    if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && other_factory_samples && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
00147       if (option_debug)
00148          ast_log(LOG_DEBUG, "Flushing audiohook %p so it remains in sync\n", audiohook);
00149       ast_slinfactory_flush(factory);
00150       ast_slinfactory_flush(other_factory);
00151    }
00152 
00153    if (ast_test_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE) && (our_factory_samples > 640 || other_factory_samples > 640)) {
00154       if (option_debug) {
00155          ast_log(LOG_DEBUG, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
00156       }
00157       ast_slinfactory_flush(factory);
00158       ast_slinfactory_flush(other_factory);
00159    }
00160 
00161    /* swap frame data for zeros if mute is required */
00162    if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) ||
00163       (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) ||
00164       (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE))) {
00165          muteme = 1;
00166    }
00167 
00168    if (muteme && frame->datalen > 0) {
00169       ast_frame_clear(frame);
00170    }
00171 
00172    /* Write frame out to respective factory */
00173    ast_slinfactory_feed(factory, frame);
00174 
00175    /* If we need to notify the respective handler of this audiohook, do so */
00176    if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
00177       ast_cond_signal(&audiohook->trigger);
00178    } else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
00179       ast_cond_signal(&audiohook->trigger);
00180    } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
00181       ast_cond_signal(&audiohook->trigger);
00182    }
00183 
00184    return 0;
00185 }
00186 
00187 static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
00188 {
00189    struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
00190    int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
00191    short buf[samples];
00192    struct ast_frame frame = {
00193       .frametype = AST_FRAME_VOICE,
00194       .subclass.codec = AST_FORMAT_SLINEAR,
00195       .data.ptr = buf,
00196       .datalen = sizeof(buf),
00197       .samples = samples,
00198    };
00199 
00200    /* Ensure the factory is able to give us the samples we want */
00201    if (samples > ast_slinfactory_available(factory))
00202       return NULL;
00203    
00204    /* Read data in from factory */
00205    if (!ast_slinfactory_read(factory, buf, samples))
00206       return NULL;
00207 
00208    /* If a volume adjustment needs to be applied apply it */
00209    if (vol)
00210       ast_frame_adjust_volume(&frame, vol);
00211 
00212    return ast_frdup(&frame);
00213 }
00214 
00215 static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
00216 {
00217    int i = 0, usable_read, usable_write;
00218    short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
00219    struct ast_frame frame = {
00220       .frametype = AST_FRAME_VOICE,
00221       .subclass.codec = AST_FORMAT_SLINEAR,
00222       .data.ptr = NULL,
00223       .datalen = sizeof(buf1),
00224       .samples = samples,
00225    };
00226 
00227    /* Make sure both factories have the required samples */
00228    usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
00229    usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
00230 
00231    if (!usable_read && !usable_write) {
00232       /* If both factories are unusable bail out */
00233       ast_debug(2, "Read factory %p and write factory %p both fail to provide %zu samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
00234       return NULL;
00235    }
00236 
00237    /* If we want to provide only a read factory make sure we aren't waiting for other audio */
00238    if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
00239       ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
00240       return NULL;
00241    }
00242 
00243    /* If we want to provide only a write factory make sure we aren't waiting for other audio */
00244    if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
00245       ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
00246       return NULL;
00247    }
00248 
00249    /* Start with the read factory... if there are enough samples, read them in */
00250    if (usable_read) {
00251       if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
00252          read_buf = buf1;
00253          /* Adjust read volume if need be */
00254          if (audiohook->options.read_volume) {
00255             int count = 0;
00256             short adjust_value = abs(audiohook->options.read_volume);
00257             for (count = 0; count < samples; count++) {
00258                if (audiohook->options.read_volume > 0)
00259                   ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
00260                else if (audiohook->options.read_volume < 0)
00261                   ast_slinear_saturated_divide(&buf1[count], &adjust_value);
00262             }
00263          }
00264       }
00265    } else if (option_debug > 2)
00266       ast_log(LOG_DEBUG, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
00267 
00268    /* Move on to the write factory... if there are enough samples, read them in */
00269    if (usable_write) {
00270       if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
00271          write_buf = buf2;
00272          /* Adjust write volume if need be */
00273          if (audiohook->options.write_volume) {
00274             int count = 0;
00275             short adjust_value = abs(audiohook->options.write_volume);
00276             for (count = 0; count < samples; count++) {
00277                if (audiohook->options.write_volume > 0)
00278                   ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
00279                else if (audiohook->options.write_volume < 0)
00280                   ast_slinear_saturated_divide(&buf2[count], &adjust_value);
00281             }
00282          }
00283       }
00284    } else if (option_debug > 2)
00285       ast_log(LOG_DEBUG, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
00286 
00287    /* Basically we figure out which buffer to use... and if mixing can be done here */
00288    if (!read_buf && !write_buf)
00289       return NULL;
00290    else if (read_buf && write_buf) {
00291       for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
00292          ast_slinear_saturated_add(data1, data2);
00293       final_buf = buf1;
00294    } else if (read_buf)
00295       final_buf = buf1;
00296    else if (write_buf)
00297       final_buf = buf2;
00298 
00299    /* Make the final buffer part of the frame, so it gets duplicated fine */
00300    frame.data.ptr = final_buf;
00301 
00302    /* Yahoo, a combined copy of the audio! */
00303    return ast_frdup(&frame);
00304 }
00305 
00306 /*! \brief Reads a frame in from the audiohook structure
00307  * \param audiohook Audiohook structure
00308  * \param samples Number of samples wanted
00309  * \param direction Direction the audio frame came from
00310  * \param format Format of frame remote side wants back
00311  * \return Returns frame on success, NULL on failure
00312  */
00313 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, format_t format)
00314 {
00315    struct ast_frame *read_frame = NULL, *final_frame = NULL;
00316 
00317    if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
00318       return NULL;
00319 
00320    /* If they don't want signed linear back out, we'll have to send it through the translation path */
00321    if (format != AST_FORMAT_SLINEAR) {
00322       /* Rebuild translation path if different format then previously */
00323       if (audiohook->format != format) {
00324          if (audiohook->trans_pvt) {
00325             ast_translator_free_path(audiohook->trans_pvt);
00326             audiohook->trans_pvt = NULL;
00327          }
00328          /* 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 */
00329          if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
00330             ast_frfree(read_frame);
00331             return NULL;
00332          }
00333       }
00334       /* Convert to requested format, and allow the read in frame to be freed */
00335       final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
00336    } else {
00337       final_frame = read_frame;
00338    }
00339 
00340    return final_frame;
00341 }
00342 
00343 /*! \brief Attach audiohook to channel
00344  * \param chan Channel
00345  * \param audiohook Audiohook structure
00346  * \return Returns 0 on success, -1 on failure
00347  */
00348 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
00349 {
00350    ast_channel_lock(chan);
00351 
00352    if (!chan->audiohooks) {
00353       /* Whoops... allocate a new structure */
00354       if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
00355          ast_channel_unlock(chan);
00356          return -1;
00357       }
00358       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
00359       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
00360       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
00361    }
00362 
00363    /* Drop into respective list */
00364    if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
00365       AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
00366    else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
00367       AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
00368    else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
00369       AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
00370 
00371    /* Change status over to running since it is now attached */
00372    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
00373 
00374    ast_channel_unlock(chan);
00375 
00376    return 0;
00377 }
00378 
00379 /*! \brief Update audiohook's status
00380  * \param audiohook Audiohook structure
00381  * \param status Audiohook status enum
00382  *
00383  * \note once status is updated to DONE, this function can not be used to set the
00384  * status back to any other setting.  Setting DONE effectively locks the status as such.
00385  */
00386 
00387 void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
00388 {
00389    ast_audiohook_lock(audiohook);
00390    if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
00391       audiohook->status = status;
00392       ast_cond_signal(&audiohook->trigger);
00393    }
00394    ast_audiohook_unlock(audiohook);
00395 }
00396 
00397 /*! \brief Detach audiohook from channel
00398  * \param audiohook Audiohook structure
00399  * \return Returns 0 on success, -1 on failure
00400  */
00401 int ast_audiohook_detach(struct ast_audiohook *audiohook)
00402 {
00403    if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
00404       return 0;
00405 
00406    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
00407 
00408    while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
00409       ast_audiohook_trigger_wait(audiohook);
00410 
00411    return 0;
00412 }
00413 
00414 /*! \brief Detach audiohooks from list and destroy said list
00415  * \param audiohook_list List of audiohooks
00416  * \return Returns 0 on success, -1 on failure
00417  */
00418 int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
00419 {
00420    int i = 0;
00421    struct ast_audiohook *audiohook = NULL;
00422 
00423    /* Drop any spies */
00424    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
00425       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00426    }
00427 
00428    /* Drop any whispering sources */
00429    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
00430       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00431    }
00432 
00433    /* Drop any manipulaters */
00434    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
00435       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00436       audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
00437    }
00438 
00439    /* Drop translation paths if present */
00440    for (i = 0; i < 2; i++) {
00441       if (audiohook_list->in_translate[i].trans_pvt)
00442          ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
00443       if (audiohook_list->out_translate[i].trans_pvt)
00444          ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
00445    }
00446    
00447    /* Free ourselves */
00448    ast_free(audiohook_list);
00449 
00450    return 0;
00451 }
00452 
00453 /*! \brief find an audiohook based on its source
00454  * \param audiohook_list The list of audiohooks to search in
00455  * \param source The source of the audiohook we wish to find
00456  * \return Return the corresponding audiohook or NULL if it cannot be found.
00457  */
00458 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
00459 {
00460    struct ast_audiohook *audiohook = NULL;
00461 
00462    AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
00463       if (!strcasecmp(audiohook->source, source))
00464          return audiohook;
00465    }
00466 
00467    AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
00468       if (!strcasecmp(audiohook->source, source))
00469          return audiohook;
00470    }
00471 
00472    AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
00473       if (!strcasecmp(audiohook->source, source))
00474          return audiohook;
00475    }
00476 
00477    return NULL;
00478 }
00479 
00480 void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
00481 {
00482    struct ast_audiohook *audiohook;
00483    enum ast_audiohook_status oldstatus;
00484 
00485    if (!old_chan->audiohooks || !(audiohook = find_audiohook_by_source(old_chan->audiohooks, source))) {
00486       return;
00487    }
00488 
00489    /* By locking both channels and the audiohook, we can assure that
00490     * another thread will not have a chance to read the audiohook's status
00491     * as done, even though ast_audiohook_remove signals the trigger
00492     * condition.
00493     */
00494    ast_audiohook_lock(audiohook);
00495    oldstatus = audiohook->status;
00496 
00497    ast_audiohook_remove(old_chan, audiohook);
00498    ast_audiohook_attach(new_chan, audiohook);
00499 
00500    audiohook->status = oldstatus;
00501    ast_audiohook_unlock(audiohook);
00502 }
00503 
00504 /*! \brief Detach specified source audiohook from channel
00505  * \param chan Channel to detach from
00506  * \param source Name of source to detach
00507  * \return Returns 0 on success, -1 on failure
00508  */
00509 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
00510 {
00511    struct ast_audiohook *audiohook = NULL;
00512 
00513    ast_channel_lock(chan);
00514 
00515    /* Ensure the channel has audiohooks on it */
00516    if (!chan->audiohooks) {
00517       ast_channel_unlock(chan);
00518       return -1;
00519    }
00520 
00521    audiohook = find_audiohook_by_source(chan->audiohooks, source);
00522 
00523    ast_channel_unlock(chan);
00524 
00525    if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
00526       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
00527 
00528    return (audiohook ? 0 : -1);
00529 }
00530 
00531 /*!
00532  * \brief Remove an audiohook from a specified channel
00533  *
00534  * \param chan Channel to remove from
00535  * \param audiohook Audiohook to remove
00536  *
00537  * \return Returns 0 on success, -1 on failure
00538  *
00539  * \note The channel does not need to be locked before calling this function
00540  */
00541 int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
00542 {
00543    ast_channel_lock(chan);
00544 
00545    if (!chan->audiohooks) {
00546       ast_channel_unlock(chan);
00547       return -1;
00548    }
00549 
00550    if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
00551       AST_LIST_REMOVE(&chan->audiohooks->spy_list, audiohook, list);
00552    else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
00553       AST_LIST_REMOVE(&chan->audiohooks->whisper_list, audiohook, list);
00554    else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
00555       AST_LIST_REMOVE(&chan->audiohooks->manipulate_list, audiohook, list);
00556 
00557    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00558 
00559    ast_channel_unlock(chan);
00560 
00561    return 0;
00562 }
00563 
00564 /*! \brief Pass a DTMF frame off to be handled by the audiohook core
00565  * \param chan Channel that the list is coming off of
00566  * \param audiohook_list List of audiohooks
00567  * \param direction Direction frame is coming in from
00568  * \param frame The frame itself
00569  * \return Return frame on success, NULL on failure
00570  */
00571 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)
00572 {
00573    struct ast_audiohook *audiohook = NULL;
00574 
00575    AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
00576       ast_audiohook_lock(audiohook);
00577       if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00578          AST_LIST_REMOVE_CURRENT(list);
00579          ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00580          ast_audiohook_unlock(audiohook);
00581          audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
00582          continue;
00583       }
00584       if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
00585          audiohook->manipulate_callback(audiohook, chan, frame, direction);
00586       ast_audiohook_unlock(audiohook);
00587    }
00588    AST_LIST_TRAVERSE_SAFE_END;
00589 
00590    return frame;
00591 }
00592 
00593 /*!
00594  * \brief Pass an AUDIO frame off to be handled by the audiohook core
00595  *
00596  * \details
00597  * This function has 3 ast_frames and 3 parts to handle each.  At the beginning of this
00598  * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
00599  * input frame.
00600  *
00601  * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
00602  *         format.  The result of this part is middle_frame is guaranteed to be in
00603  *         SLINEAR format for Part_2.
00604  * Part_2: Send middle_frame off to spies and manipulators.  At this point middle_frame is
00605  *         either a new frame as result of the translation, or points directly to the start_frame
00606  *         because no translation to SLINEAR audio was required.  The result of this part
00607  *         is end_frame will be updated to point to middle_frame if any audiohook manipulation
00608  *         took place.
00609  * Part_3: Translate end_frame's audio back into the format of start frame if necessary.
00610  *         At this point if middle_frame != end_frame, we are guaranteed that no manipulation
00611  *         took place and middle_frame can be freed as it was translated... If middle_frame was
00612  *         not translated and still pointed to start_frame, it would be equal to end_frame as well
00613  *         regardless if manipulation took place which would not result in this free.  The result
00614  *         of this part is end_frame is guaranteed to be the format of start_frame for the return.
00615  *         
00616  * \param chan Channel that the list is coming off of
00617  * \param audiohook_list List of audiohooks
00618  * \param direction Direction frame is coming in from
00619  * \param frame The frame itself
00620  * \return Return frame on success, NULL on failure
00621  */
00622 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)
00623 {
00624    struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
00625    struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
00626    struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
00627    struct ast_audiohook *audiohook = NULL;
00628    int samples = frame->samples;
00629 
00630    /* ---Part_1. translate start_frame to SLINEAR if necessary. */
00631    /* If the frame coming in is not signed linear we have to send it through the in_translate path */
00632    if (frame->subclass.codec != AST_FORMAT_SLINEAR) {
00633       if (in_translate->format != frame->subclass.codec) {
00634          if (in_translate->trans_pvt)
00635             ast_translator_free_path(in_translate->trans_pvt);
00636          if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass.codec)))
00637             return frame;
00638          in_translate->format = frame->subclass.codec;
00639       }
00640       if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
00641          return frame;
00642       samples = middle_frame->samples;
00643    }
00644 
00645    /* ---Part_2: Send middle_frame to spy and manipulator lists.  middle_frame is guaranteed to be SLINEAR here.*/
00646    /* Queue up signed linear frame to each spy */
00647    AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
00648       ast_audiohook_lock(audiohook);
00649       if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00650          AST_LIST_REMOVE_CURRENT(list);
00651          ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00652          ast_audiohook_unlock(audiohook);
00653          continue;
00654       }
00655       ast_audiohook_write_frame(audiohook, direction, middle_frame);
00656       ast_audiohook_unlock(audiohook);
00657    }
00658    AST_LIST_TRAVERSE_SAFE_END;
00659 
00660    /* If this frame is being written out to the channel then we need to use whisper sources */
00661    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
00662       int i = 0;
00663       short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
00664       memset(&combine_buf, 0, sizeof(combine_buf));
00665       AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
00666          ast_audiohook_lock(audiohook);
00667          if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00668             AST_LIST_REMOVE_CURRENT(list);
00669             ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00670             ast_audiohook_unlock(audiohook);
00671             continue;
00672          }
00673          if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
00674             /* Take audio from this whisper source and combine it into our main buffer */
00675             for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
00676                ast_slinear_saturated_add(data1, data2);
00677          }
00678          ast_audiohook_unlock(audiohook);
00679       }
00680       AST_LIST_TRAVERSE_SAFE_END;
00681       /* We take all of the combined whisper sources and combine them into the audio being written out */
00682       for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++)
00683          ast_slinear_saturated_add(data1, data2);
00684       end_frame = middle_frame;
00685    }
00686 
00687    /* Pass off frame to manipulate audiohooks */
00688    if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
00689       AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
00690          ast_audiohook_lock(audiohook);
00691          if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00692             AST_LIST_REMOVE_CURRENT(list);
00693             ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00694             ast_audiohook_unlock(audiohook);
00695             /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
00696             audiohook->manipulate_callback(audiohook, chan, NULL, direction);
00697             continue;
00698          }
00699          /* Feed in frame to manipulation. */
00700          if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
00701             /* XXX IGNORE FAILURE */
00702 
00703             /* If the manipulation fails then the frame will be returned in its original state.
00704              * Since there are potentially more manipulator callbacks in the list, no action should
00705              * be taken here to exit early. */
00706          }
00707          ast_audiohook_unlock(audiohook);
00708       }
00709       AST_LIST_TRAVERSE_SAFE_END;
00710       end_frame = middle_frame;
00711    }
00712 
00713    /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
00714    if (middle_frame == end_frame) {
00715       /* Middle frame was modified and became the end frame... let's see if we need to transcode */
00716       if (end_frame->subclass.codec != start_frame->subclass.codec) {
00717          if (out_translate->format != start_frame->subclass.codec) {
00718             if (out_translate->trans_pvt)
00719                ast_translator_free_path(out_translate->trans_pvt);
00720             if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass.codec, AST_FORMAT_SLINEAR))) {
00721                /* We can't transcode this... drop our middle frame and return the original */
00722                ast_frfree(middle_frame);
00723                return start_frame;
00724             }
00725             out_translate->format = start_frame->subclass.codec;
00726          }
00727          /* Transcode from our middle (signed linear) frame to new format of the frame that came in */
00728          if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
00729             /* Failed to transcode the frame... drop it and return the original */
00730             ast_frfree(middle_frame);
00731             return start_frame;
00732          }
00733          /* Here's the scoop... middle frame is no longer of use to us */
00734          ast_frfree(middle_frame);
00735       }
00736    } else {
00737       /* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
00738       ast_frfree(middle_frame);
00739    }
00740 
00741    return end_frame;
00742 }
00743 
00744 int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
00745 {
00746    if (AST_LIST_EMPTY(&audiohook_list->spy_list) &&
00747       AST_LIST_EMPTY(&audiohook_list->whisper_list) &&
00748       AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
00749 
00750       return 1;
00751    }
00752    return 0;
00753 }
00754 
00755 /*! \brief Pass a frame off to be handled by the audiohook core
00756  * \param chan Channel that the list is coming off of
00757  * \param audiohook_list List of audiohooks
00758  * \param direction Direction frame is coming in from
00759  * \param frame The frame itself
00760  * \return Return frame on success, NULL on failure
00761  */
00762 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)
00763 {
00764    /* Pass off frame to it's respective list write function */
00765    if (frame->frametype == AST_FRAME_VOICE)
00766       return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
00767    else if (frame->frametype == AST_FRAME_DTMF)
00768       return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
00769    else
00770       return frame;
00771 }
00772 
00773 /*! \brief Wait for audiohook trigger to be triggered
00774  * \param audiohook Audiohook to wait on
00775  */
00776 void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
00777 {
00778    struct timeval wait;
00779    struct timespec ts;
00780 
00781    wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
00782    ts.tv_sec = wait.tv_sec;
00783    ts.tv_nsec = wait.tv_usec * 1000;
00784    
00785    ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
00786    
00787    return;
00788 }
00789 
00790 /* Count number of channel audiohooks by type, regardless of type */
00791 int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
00792 {
00793    int count = 0;
00794    struct ast_audiohook *ah = NULL;
00795 
00796    if (!chan->audiohooks)
00797       return -1;
00798 
00799    switch (type) {
00800       case AST_AUDIOHOOK_TYPE_SPY:
00801          AST_LIST_TRAVERSE(&chan->audiohooks->spy_list, ah, list) {
00802             if (!strcmp(ah->source, source)) {
00803                count++;
00804             }
00805          }
00806          break;
00807       case AST_AUDIOHOOK_TYPE_WHISPER:
00808          AST_LIST_TRAVERSE(&chan->audiohooks->whisper_list, ah, list) {
00809             if (!strcmp(ah->source, source)) {
00810                count++;
00811             }
00812          }
00813          break;
00814       case AST_AUDIOHOOK_TYPE_MANIPULATE:
00815          AST_LIST_TRAVERSE(&chan->audiohooks->manipulate_list, ah, list) {
00816             if (!strcmp(ah->source, source)) {
00817                count++;
00818             }
00819          }
00820          break;
00821       default:
00822          ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%u)\n", type);
00823          return -1;
00824    }
00825 
00826    return count;
00827 }
00828 
00829 /* Count number of channel audiohooks by type that are running */
00830 int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
00831 {
00832    int count = 0;
00833    struct ast_audiohook *ah = NULL;
00834    if (!chan->audiohooks)
00835       return -1;
00836 
00837    switch (type) {
00838       case AST_AUDIOHOOK_TYPE_SPY:
00839          AST_LIST_TRAVERSE(&chan->audiohooks->spy_list, ah, list) {
00840             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00841                count++;
00842          }
00843          break;
00844       case AST_AUDIOHOOK_TYPE_WHISPER:
00845          AST_LIST_TRAVERSE(&chan->audiohooks->whisper_list, ah, list) {
00846             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00847                count++;
00848          }
00849          break;
00850       case AST_AUDIOHOOK_TYPE_MANIPULATE:
00851          AST_LIST_TRAVERSE(&chan->audiohooks->manipulate_list, ah, list) {
00852             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00853                count++;
00854          }
00855          break;
00856       default:
00857          ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%u)\n", type);
00858          return -1;
00859    }
00860    return count;
00861 }
00862 
00863 /*! \brief Audiohook volume adjustment structure */
00864 struct audiohook_volume {
00865    struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
00866    int read_adjustment;            /*!< Value to adjust frames read from the channel by */
00867    int write_adjustment;           /*!< Value to adjust frames written to the channel by */
00868 };
00869 
00870 /*! \brief Callback used to destroy the audiohook volume datastore
00871  * \param data Volume information structure
00872  * \return Returns nothing
00873  */
00874 static void audiohook_volume_destroy(void *data)
00875 {
00876    struct audiohook_volume *audiohook_volume = data;
00877 
00878    /* Destroy the audiohook as it is no longer in use */
00879    ast_audiohook_destroy(&audiohook_volume->audiohook);
00880 
00881    /* Finally free ourselves, we are of no more use */
00882    ast_free(audiohook_volume);
00883 
00884    return;
00885 }
00886 
00887 /*! \brief Datastore used to store audiohook volume information */
00888 static const struct ast_datastore_info audiohook_volume_datastore = {
00889    .type = "Volume",
00890    .destroy = audiohook_volume_destroy,
00891 };
00892 
00893 /*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
00894  * \param audiohook Audiohook attached to the channel
00895  * \param chan Channel we are attached to
00896  * \param frame Frame of audio we want to manipulate
00897  * \param direction Direction the audio came in from
00898  * \return Returns 0 on success, -1 on failure
00899  */
00900 static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
00901 {
00902    struct ast_datastore *datastore = NULL;
00903    struct audiohook_volume *audiohook_volume = NULL;
00904    int *gain = NULL;
00905 
00906    /* If the audiohook is shutting down don't even bother */
00907    if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
00908       return 0;
00909    }
00910 
00911    /* Try to find the datastore containg adjustment information, if we can't just bail out */
00912    if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
00913       return 0;
00914    }
00915 
00916    audiohook_volume = datastore->data;
00917 
00918    /* Based on direction grab the appropriate adjustment value */
00919    if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
00920       gain = &audiohook_volume->read_adjustment;
00921    } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
00922       gain = &audiohook_volume->write_adjustment;
00923    }
00924 
00925    /* If an adjustment value is present modify the frame */
00926    if (gain && *gain) {
00927       ast_frame_adjust_volume(frame, *gain);
00928    }
00929 
00930    return 0;
00931 }
00932 
00933 /*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
00934  * \param chan Channel to look on
00935  * \param create Whether to create the datastore if not found
00936  * \return Returns audiohook_volume structure on success, NULL on failure
00937  */
00938 static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
00939 {
00940    struct ast_datastore *datastore = NULL;
00941    struct audiohook_volume *audiohook_volume = NULL;
00942 
00943    /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
00944    if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
00945       return datastore->data;
00946    }
00947 
00948    /* 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 */
00949    if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
00950       return NULL;
00951    }
00952 
00953    /* Create a new audiohook_volume structure to contain our adjustments and audiohook */
00954    if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
00955       ast_datastore_free(datastore);
00956       return NULL;
00957    }
00958 
00959    /* Setup our audiohook structure so we can manipulate the audio */
00960    ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume");
00961    audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback;
00962 
00963    /* Attach the audiohook_volume blob to the datastore and attach to the channel */
00964    datastore->data = audiohook_volume;
00965    ast_channel_datastore_add(chan, datastore);
00966 
00967    /* All is well... put the audiohook into motion */
00968    ast_audiohook_attach(chan, &audiohook_volume->audiohook);
00969 
00970    return audiohook_volume;
00971 }
00972 
00973 /*! \brief Adjust the volume on frames read from or written to a channel
00974  * \param chan Channel to muck with
00975  * \param direction Direction to set on
00976  * \param volume Value to adjust the volume by
00977  * \return Returns 0 on success, -1 on failure
00978  */
00979 int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
00980 {
00981    struct audiohook_volume *audiohook_volume = NULL;
00982 
00983    /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
00984    if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
00985       return -1;
00986    }
00987 
00988    /* Now based on the direction set the proper value */
00989    if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
00990       audiohook_volume->read_adjustment = volume;
00991    }
00992    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
00993       audiohook_volume->write_adjustment = volume;
00994    }
00995 
00996    return 0;
00997 }
00998 
00999 /*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
01000  * \param chan Channel to retrieve volume adjustment from
01001  * \param direction Direction to retrieve
01002  * \return Returns adjustment value
01003  */
01004 int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
01005 {
01006    struct audiohook_volume *audiohook_volume = NULL;
01007    int adjustment = 0;
01008 
01009    /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
01010    if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
01011       return 0;
01012    }
01013 
01014    /* Grab the adjustment value based on direction given */
01015    if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
01016       adjustment = audiohook_volume->read_adjustment;
01017    } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
01018       adjustment = audiohook_volume->write_adjustment;
01019    }
01020 
01021    return adjustment;
01022 }
01023 
01024 /*! \brief Adjust the volume on frames read from or written to a channel
01025  * \param chan Channel to muck with
01026  * \param direction Direction to increase
01027  * \param volume Value to adjust the adjustment by
01028  * \return Returns 0 on success, -1 on failure
01029  */
01030 int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
01031 {
01032    struct audiohook_volume *audiohook_volume = NULL;
01033 
01034    /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
01035    if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
01036       return -1;
01037    }
01038 
01039    /* Based on the direction change the specific adjustment value */
01040    if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
01041       audiohook_volume->read_adjustment += volume;
01042    }
01043    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
01044       audiohook_volume->write_adjustment += volume;
01045    }
01046 
01047    return 0;
01048 }
01049 
01050 /*! \brief Mute frames read from or written to a channel
01051  * \param chan Channel to muck with
01052  * \param source Type of audiohook
01053  * \param flag which flag to set / clear
01054  * \param clear set or clear
01055  * \return Returns 0 on success, -1 on failure
01056  */
01057 int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
01058 {
01059    struct ast_audiohook *audiohook = NULL;
01060 
01061    ast_channel_lock(chan);
01062 
01063    /* Ensure the channel has audiohooks on it */
01064    if (!chan->audiohooks) {
01065       ast_channel_unlock(chan);
01066       return -1;
01067    }
01068 
01069    audiohook = find_audiohook_by_source(chan->audiohooks, source);
01070 
01071    if (audiohook) {
01072       if (clear) {
01073          ast_clear_flag(audiohook, flag);
01074       } else {
01075          ast_set_flag(audiohook, flag);
01076       }
01077    }
01078 
01079    ast_channel_unlock(chan);
01080 
01081    return (audiohook ? 0 : -1);
01082 }

Generated on 7 Sep 2017 for Asterisk - The Open Source Telephony Project by  doxygen 1.6.1