Sat Mar 10 01:54:04 2012

Asterisk developer's documentation


audiohook.c

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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 #include "asterisk.h"
00027 
00028 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 349289 $")
00029 
00030 #include <signal.h>
00031 
00032 #include "asterisk/channel.h"
00033 #include "asterisk/utils.h"
00034 #include "asterisk/lock.h"
00035 #include "asterisk/linkedlists.h"
00036 #include "asterisk/audiohook.h"
00037 #include "asterisk/slinfactory.h"
00038 #include "asterisk/frame.h"
00039 #include "asterisk/translate.h"
00040 
00041 struct ast_audiohook_translate {
00042    struct ast_trans_pvt *trans_pvt;
00043    format_t format;
00044 };
00045 
00046 struct ast_audiohook_list {
00047    struct ast_audiohook_translate in_translate[2];
00048    struct ast_audiohook_translate out_translate[2];
00049    AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
00050    AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
00051    AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
00052 };
00053 
00054 /*! \brief Initialize an audiohook structure
00055  * \param audiohook Audiohook structure
00056  * \param type
00057  * \param source
00058  * \return Returns 0 on success, -1 on failure
00059  */
00060 int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source)
00061 {
00062    /* Need to keep the type and source */
00063    audiohook->type = type;
00064    audiohook->source = source;
00065 
00066    /* Initialize lock that protects our audiohook */
00067    ast_mutex_init(&audiohook->lock);
00068    ast_cond_init(&audiohook->trigger, NULL);
00069 
00070    /* Setup the factories that are needed for this audiohook type */
00071    switch (type) {
00072    case AST_AUDIOHOOK_TYPE_SPY:
00073       ast_slinfactory_init(&audiohook->read_factory);
00074    case AST_AUDIOHOOK_TYPE_WHISPER:
00075       ast_slinfactory_init(&audiohook->write_factory);
00076       break;
00077    default:
00078       break;
00079    }
00080 
00081    /* Since we are just starting out... this audiohook is new */
00082    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
00083 
00084    return 0;
00085 }
00086 
00087 /*! \brief Destroys an audiohook structure
00088  * \param audiohook Audiohook structure
00089  * \return Returns 0 on success, -1 on failure
00090  */
00091 int ast_audiohook_destroy(struct ast_audiohook *audiohook)
00092 {
00093    /* Drop the factories used by this audiohook type */
00094    switch (audiohook->type) {
00095    case AST_AUDIOHOOK_TYPE_SPY:
00096       ast_slinfactory_destroy(&audiohook->read_factory);
00097    case AST_AUDIOHOOK_TYPE_WHISPER:
00098       ast_slinfactory_destroy(&audiohook->write_factory);
00099       break;
00100    default:
00101       break;
00102    }
00103 
00104    /* Destroy translation path if present */
00105    if (audiohook->trans_pvt)
00106       ast_translator_free_path(audiohook->trans_pvt);
00107 
00108    /* Lock and trigger be gone! */
00109    ast_cond_destroy(&audiohook->trigger);
00110    ast_mutex_destroy(&audiohook->lock);
00111 
00112    return 0;
00113 }
00114 
00115 /*! \brief Writes a frame into the audiohook structure
00116  * \param audiohook Audiohook structure
00117  * \param direction Direction the audio frame came from
00118  * \param frame Frame to write in
00119  * \return Returns 0 on success, -1 on failure
00120  */
00121 int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
00122 {
00123    struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
00124    struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
00125    struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime;
00126    int our_factory_samples;
00127    int our_factory_ms;
00128    int other_factory_samples;
00129    int other_factory_ms;
00130    int muteme = 0;
00131 
00132    /* Update last feeding time to be current */
00133    *rwtime = ast_tvnow();
00134 
00135    our_factory_samples = ast_slinfactory_available(factory);
00136    our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / 8);
00137    other_factory_samples = ast_slinfactory_available(other_factory);
00138    other_factory_ms = other_factory_samples / 8;
00139 
00140    if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && other_factory_samples && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
00141       if (option_debug)
00142          ast_log(LOG_DEBUG, "Flushing audiohook %p so it remains in sync\n", audiohook);
00143       ast_slinfactory_flush(factory);
00144       ast_slinfactory_flush(other_factory);
00145    }
00146 
00147    if (ast_test_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE) && (our_factory_samples > 640 || other_factory_samples > 640)) {
00148       if (option_debug) {
00149          ast_log(LOG_DEBUG, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
00150       }
00151       ast_slinfactory_flush(factory);
00152       ast_slinfactory_flush(other_factory);
00153    }
00154 
00155    /* swap frame data for zeros if mute is required */
00156    if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) ||
00157       (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) ||
00158       (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE))) {
00159          muteme = 1;
00160    }
00161 
00162    if (muteme && frame->datalen > 0) {
00163       ast_frame_clear(frame);
00164    }
00165 
00166    /* Write frame out to respective factory */
00167    ast_slinfactory_feed(factory, frame);
00168 
00169    /* If we need to notify the respective handler of this audiohook, do so */
00170    if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
00171       ast_cond_signal(&audiohook->trigger);
00172    } else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
00173       ast_cond_signal(&audiohook->trigger);
00174    } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
00175       ast_cond_signal(&audiohook->trigger);
00176    }
00177 
00178    return 0;
00179 }
00180 
00181 static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
00182 {
00183    struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
00184    int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
00185    short buf[samples];
00186    struct ast_frame frame = {
00187       .frametype = AST_FRAME_VOICE,
00188       .subclass.codec = AST_FORMAT_SLINEAR,
00189       .data.ptr = buf,
00190       .datalen = sizeof(buf),
00191       .samples = samples,
00192    };
00193 
00194    /* Ensure the factory is able to give us the samples we want */
00195    if (samples > ast_slinfactory_available(factory))
00196       return NULL;
00197    
00198    /* Read data in from factory */
00199    if (!ast_slinfactory_read(factory, buf, samples))
00200       return NULL;
00201 
00202    /* If a volume adjustment needs to be applied apply it */
00203    if (vol)
00204       ast_frame_adjust_volume(&frame, vol);
00205 
00206    return ast_frdup(&frame);
00207 }
00208 
00209 static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
00210 {
00211    int i = 0, usable_read, usable_write;
00212    short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
00213    struct ast_frame frame = {
00214       .frametype = AST_FRAME_VOICE,
00215       .subclass.codec = AST_FORMAT_SLINEAR,
00216       .data.ptr = NULL,
00217       .datalen = sizeof(buf1),
00218       .samples = samples,
00219    };
00220 
00221    /* Make sure both factories have the required samples */
00222    usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
00223    usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
00224 
00225    if (!usable_read && !usable_write) {
00226       /* If both factories are unusable bail out */
00227       ast_debug(1, "Read factory %p and write factory %p both fail to provide %zd samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
00228       return NULL;
00229    }
00230 
00231    /* If we want to provide only a read factory make sure we aren't waiting for other audio */
00232    if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
00233       ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
00234       return NULL;
00235    }
00236 
00237    /* If we want to provide only a write factory make sure we aren't waiting for other audio */
00238    if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
00239       ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
00240       return NULL;
00241    }
00242 
00243    /* Start with the read factory... if there are enough samples, read them in */
00244    if (usable_read) {
00245       if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
00246          read_buf = buf1;
00247          /* Adjust read volume if need be */
00248          if (audiohook->options.read_volume) {
00249             int count = 0;
00250             short adjust_value = abs(audiohook->options.read_volume);
00251             for (count = 0; count < samples; count++) {
00252                if (audiohook->options.read_volume > 0)
00253                   ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
00254                else if (audiohook->options.read_volume < 0)
00255                   ast_slinear_saturated_divide(&buf1[count], &adjust_value);
00256             }
00257          }
00258       }
00259    } else if (option_debug)
00260       ast_log(LOG_DEBUG, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
00261 
00262    /* Move on to the write factory... if there are enough samples, read them in */
00263    if (usable_write) {
00264       if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
00265          write_buf = buf2;
00266          /* Adjust write volume if need be */
00267          if (audiohook->options.write_volume) {
00268             int count = 0;
00269             short adjust_value = abs(audiohook->options.write_volume);
00270             for (count = 0; count < samples; count++) {
00271                if (audiohook->options.write_volume > 0)
00272                   ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
00273                else if (audiohook->options.write_volume < 0)
00274                   ast_slinear_saturated_divide(&buf2[count], &adjust_value);
00275             }
00276          }
00277       }
00278    } else if (option_debug)
00279       ast_log(LOG_DEBUG, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
00280 
00281    /* Basically we figure out which buffer to use... and if mixing can be done here */
00282    if (!read_buf && !write_buf)
00283       return NULL;
00284    else if (read_buf && write_buf) {
00285       for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
00286          ast_slinear_saturated_add(data1, data2);
00287       final_buf = buf1;
00288    } else if (read_buf)
00289       final_buf = buf1;
00290    else if (write_buf)
00291       final_buf = buf2;
00292 
00293    /* Make the final buffer part of the frame, so it gets duplicated fine */
00294    frame.data.ptr = final_buf;
00295 
00296    /* Yahoo, a combined copy of the audio! */
00297    return ast_frdup(&frame);
00298 }
00299 
00300 /*! \brief Reads a frame in from the audiohook structure
00301  * \param audiohook Audiohook structure
00302  * \param samples Number of samples wanted
00303  * \param direction Direction the audio frame came from
00304  * \param format Format of frame remote side wants back
00305  * \return Returns frame on success, NULL on failure
00306  */
00307 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, format_t format)
00308 {
00309    struct ast_frame *read_frame = NULL, *final_frame = NULL;
00310 
00311    if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
00312       return NULL;
00313 
00314    /* If they don't want signed linear back out, we'll have to send it through the translation path */
00315    if (format != AST_FORMAT_SLINEAR) {
00316       /* Rebuild translation path if different format then previously */
00317       if (audiohook->format != format) {
00318          if (audiohook->trans_pvt) {
00319             ast_translator_free_path(audiohook->trans_pvt);
00320             audiohook->trans_pvt = NULL;
00321          }
00322          /* 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 */
00323          if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
00324             ast_frfree(read_frame);
00325             return NULL;
00326          }
00327       }
00328       /* Convert to requested format, and allow the read in frame to be freed */
00329       final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
00330    } else {
00331       final_frame = read_frame;
00332    }
00333 
00334    return final_frame;
00335 }
00336 
00337 /*! \brief Attach audiohook to channel
00338  * \param chan Channel
00339  * \param audiohook Audiohook structure
00340  * \return Returns 0 on success, -1 on failure
00341  */
00342 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
00343 {
00344    ast_channel_lock(chan);
00345 
00346    if (!chan->audiohooks) {
00347       /* Whoops... allocate a new structure */
00348       if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
00349          ast_channel_unlock(chan);
00350          return -1;
00351       }
00352       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
00353       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
00354       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
00355    }
00356 
00357    /* Drop into respective list */
00358    if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
00359       AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
00360    else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
00361       AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
00362    else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
00363       AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
00364 
00365    /* Change status over to running since it is now attached */
00366    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
00367 
00368    ast_channel_unlock(chan);
00369 
00370    return 0;
00371 }
00372 
00373 /*! \brief Update audiohook's status
00374  * \param audiohook Audiohook structure
00375  * \param status Audiohook status enum
00376  *
00377  * \note once status is updated to DONE, this function can not be used to set the
00378  * status back to any other setting.  Setting DONE effectively locks the status as such.
00379  */
00380 
00381 void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
00382 {
00383    ast_audiohook_lock(audiohook);
00384    if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
00385       audiohook->status = status;
00386       ast_cond_signal(&audiohook->trigger);
00387    }
00388    ast_audiohook_unlock(audiohook);
00389 }
00390 
00391 /*! \brief Detach audiohook from channel
00392  * \param audiohook Audiohook structure
00393  * \return Returns 0 on success, -1 on failure
00394  */
00395 int ast_audiohook_detach(struct ast_audiohook *audiohook)
00396 {
00397    if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
00398       return 0;
00399 
00400    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
00401 
00402    while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
00403       ast_audiohook_trigger_wait(audiohook);
00404 
00405    return 0;
00406 }
00407 
00408 /*! \brief Detach audiohooks from list and destroy said list
00409  * \param audiohook_list List of audiohooks
00410  * \return Returns 0 on success, -1 on failure
00411  */
00412 int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
00413 {
00414    int i = 0;
00415    struct ast_audiohook *audiohook = NULL;
00416 
00417    /* Drop any spies */
00418    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
00419       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00420    }
00421 
00422    /* Drop any whispering sources */
00423    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
00424       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00425    }
00426 
00427    /* Drop any manipulaters */
00428    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
00429       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00430       audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
00431    }
00432 
00433    /* Drop translation paths if present */
00434    for (i = 0; i < 2; i++) {
00435       if (audiohook_list->in_translate[i].trans_pvt)
00436          ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
00437       if (audiohook_list->out_translate[i].trans_pvt)
00438          ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
00439    }
00440    
00441    /* Free ourselves */
00442    ast_free(audiohook_list);
00443 
00444    return 0;
00445 }
00446 
00447 /*! \brief find an audiohook based on its source
00448  * \param audiohook_list The list of audiohooks to search in
00449  * \param source The source of the audiohook we wish to find
00450  * \return Return the corresponding audiohook or NULL if it cannot be found.
00451  */
00452 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
00453 {
00454    struct ast_audiohook *audiohook = NULL;
00455 
00456    AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
00457       if (!strcasecmp(audiohook->source, source))
00458          return audiohook;
00459    }
00460 
00461    AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
00462       if (!strcasecmp(audiohook->source, source))
00463          return audiohook;
00464    }
00465 
00466    AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
00467       if (!strcasecmp(audiohook->source, source))
00468          return audiohook;
00469    }
00470 
00471    return NULL;
00472 }
00473 
00474 void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
00475 {
00476    struct ast_audiohook *audiohook;
00477    enum ast_audiohook_status oldstatus;
00478 
00479    if (!old_chan->audiohooks || !(audiohook = find_audiohook_by_source(old_chan->audiohooks, source))) {
00480       return;
00481    }
00482 
00483    /* By locking both channels and the audiohook, we can assure that
00484     * another thread will not have a chance to read the audiohook's status
00485     * as done, even though ast_audiohook_remove signals the trigger
00486     * condition.
00487     */
00488    ast_audiohook_lock(audiohook);
00489    oldstatus = audiohook->status;
00490 
00491    ast_audiohook_remove(old_chan, audiohook);
00492    ast_audiohook_attach(new_chan, audiohook);
00493 
00494    audiohook->status = oldstatus;
00495    ast_audiohook_unlock(audiohook);
00496 }
00497 
00498 /*! \brief Detach specified source audiohook from channel
00499  * \param chan Channel to detach from
00500  * \param source Name of source to detach
00501  * \return Returns 0 on success, -1 on failure
00502  */
00503 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
00504 {
00505    struct ast_audiohook *audiohook = NULL;
00506 
00507    ast_channel_lock(chan);
00508 
00509    /* Ensure the channel has audiohooks on it */
00510    if (!chan->audiohooks) {
00511       ast_channel_unlock(chan);
00512       return -1;
00513    }
00514 
00515    audiohook = find_audiohook_by_source(chan->audiohooks, source);
00516 
00517    ast_channel_unlock(chan);
00518 
00519    if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
00520       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
00521 
00522    return (audiohook ? 0 : -1);
00523 }
00524 
00525 /*!
00526  * \brief Remove an audiohook from a specified channel
00527  *
00528  * \param chan Channel to remove from
00529  * \param audiohook Audiohook to remove
00530  *
00531  * \return Returns 0 on success, -1 on failure
00532  *
00533  * \note The channel does not need to be locked before calling this function
00534  */
00535 int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
00536 {
00537    ast_channel_lock(chan);
00538 
00539    if (!chan->audiohooks) {
00540       ast_channel_unlock(chan);
00541       return -1;
00542    }
00543 
00544    if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
00545       AST_LIST_REMOVE(&chan->audiohooks->spy_list, audiohook, list);
00546    else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
00547       AST_LIST_REMOVE(&chan->audiohooks->whisper_list, audiohook, list);
00548    else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
00549       AST_LIST_REMOVE(&chan->audiohooks->manipulate_list, audiohook, list);
00550 
00551    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00552 
00553    ast_channel_unlock(chan);
00554 
00555    return 0;
00556 }
00557 
00558 /*! \brief Pass a DTMF frame off to be handled by the audiohook core
00559  * \param chan Channel that the list is coming off of
00560  * \param audiohook_list List of audiohooks
00561  * \param direction Direction frame is coming in from
00562  * \param frame The frame itself
00563  * \return Return frame on success, NULL on failure
00564  */
00565 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)
00566 {
00567    struct ast_audiohook *audiohook = NULL;
00568 
00569    AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
00570       ast_audiohook_lock(audiohook);
00571       if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00572          AST_LIST_REMOVE_CURRENT(list);
00573          ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00574          ast_audiohook_unlock(audiohook);
00575          audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
00576          continue;
00577       }
00578       if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
00579          audiohook->manipulate_callback(audiohook, chan, frame, direction);
00580       ast_audiohook_unlock(audiohook);
00581    }
00582    AST_LIST_TRAVERSE_SAFE_END;
00583 
00584    return frame;
00585 }
00586 
00587 /*!
00588  * \brief Pass an AUDIO frame off to be handled by the audiohook core
00589  *
00590  * \details
00591  * This function has 3 ast_frames and 3 parts to handle each.  At the beginning of this
00592  * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
00593  * input frame.
00594  *
00595  * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
00596  *         format.  The result of this part is middle_frame is guaranteed to be in
00597  *         SLINEAR format for Part_2.
00598  * Part_2: Send middle_frame off to spies and manipulators.  At this point middle_frame is
00599  *         either a new frame as result of the translation, or points directly to the start_frame
00600  *         because no translation to SLINEAR audio was required.  The result of this part
00601  *         is end_frame will be updated to point to middle_frame if any audiohook manipulation
00602  *         took place.
00603  * Part_3: Translate end_frame's audio back into the format of start frame if necessary.
00604  *         At this point if middle_frame != end_frame, we are guaranteed that no manipulation
00605  *         took place and middle_frame can be freed as it was translated... If middle_frame was
00606  *         not translated and still pointed to start_frame, it would be equal to end_frame as well
00607  *         regardless if manipulation took place which would not result in this free.  The result
00608  *         of this part is end_frame is guaranteed to be the format of start_frame for the return.
00609  *         
00610  * \param chan Channel that the list is coming off of
00611  * \param audiohook_list List of audiohooks
00612  * \param direction Direction frame is coming in from
00613  * \param frame The frame itself
00614  * \return Return frame on success, NULL on failure
00615  */
00616 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)
00617 {
00618    struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
00619    struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
00620    struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
00621    struct ast_audiohook *audiohook = NULL;
00622    int samples = frame->samples;
00623 
00624    /* Don't translate our frame if we aren't going to bother to use it */
00625    if (ast_audiohook_write_list_empty(audiohook_list))
00626       return end_frame;
00627 
00628    /* ---Part_1. translate start_frame to SLINEAR if necessary. */
00629    /* If the frame coming in is not signed linear we have to send it through the in_translate path */
00630    if (frame->subclass.codec != AST_FORMAT_SLINEAR) {
00631       if (in_translate->format != frame->subclass.codec) {
00632          if (in_translate->trans_pvt)
00633             ast_translator_free_path(in_translate->trans_pvt);
00634          if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass.codec)))
00635             return frame;
00636          in_translate->format = frame->subclass.codec;
00637       }
00638       if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
00639          return frame;
00640       samples = middle_frame->samples;
00641    }
00642 
00643    /* ---Part_2: Send middle_frame to spy and manipulator lists.  middle_frame is guaranteed to be SLINEAR here.*/
00644    /* Queue up signed linear frame to each spy */
00645    AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
00646       ast_audiohook_lock(audiohook);
00647       if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00648          AST_LIST_REMOVE_CURRENT(list);
00649          ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00650          ast_audiohook_unlock(audiohook);
00651          continue;
00652       }
00653       ast_audiohook_write_frame(audiohook, direction, middle_frame);
00654       ast_audiohook_unlock(audiohook);
00655    }
00656    AST_LIST_TRAVERSE_SAFE_END;
00657 
00658    /* If this frame is being written out to the channel then we need to use whisper sources */
00659    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
00660       int i = 0;
00661       short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
00662       memset(&combine_buf, 0, sizeof(combine_buf));
00663       AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
00664          ast_audiohook_lock(audiohook);
00665          if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00666             AST_LIST_REMOVE_CURRENT(list);
00667             ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00668             ast_audiohook_unlock(audiohook);
00669             continue;
00670          }
00671          if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
00672             /* Take audio from this whisper source and combine it into our main buffer */
00673             for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
00674                ast_slinear_saturated_add(data1, data2);
00675          }
00676          ast_audiohook_unlock(audiohook);
00677       }
00678       AST_LIST_TRAVERSE_SAFE_END;
00679       /* We take all of the combined whisper sources and combine them into the audio being written out */
00680       for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++)
00681          ast_slinear_saturated_add(data1, data2);
00682       end_frame = middle_frame;
00683    }
00684 
00685    /* Pass off frame to manipulate audiohooks */
00686    if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
00687       AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
00688          ast_audiohook_lock(audiohook);
00689          if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00690             AST_LIST_REMOVE_CURRENT(list);
00691             ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00692             ast_audiohook_unlock(audiohook);
00693             /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
00694             audiohook->manipulate_callback(audiohook, chan, NULL, direction);
00695             continue;
00696          }
00697          /* Feed in frame to manipulation. */
00698          if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
00699             /* XXX IGNORE FAILURE */
00700 
00701             /* If the manipulation fails then the frame will be returned in its original state.
00702              * Since there are potentially more manipulator callbacks in the list, no action should
00703              * be taken here to exit early. */
00704          }
00705          ast_audiohook_unlock(audiohook);
00706       }
00707       AST_LIST_TRAVERSE_SAFE_END;
00708       end_frame = middle_frame;
00709    }
00710 
00711    /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
00712    if (middle_frame == end_frame) {
00713       /* Middle frame was modified and became the end frame... let's see if we need to transcode */
00714       if (end_frame->subclass.codec != start_frame->subclass.codec) {
00715          if (out_translate->format != start_frame->subclass.codec) {
00716             if (out_translate->trans_pvt)
00717                ast_translator_free_path(out_translate->trans_pvt);
00718             if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass.codec, AST_FORMAT_SLINEAR))) {
00719                /* We can't transcode this... drop our middle frame and return the original */
00720                ast_frfree(middle_frame);
00721                return start_frame;
00722             }
00723             out_translate->format = start_frame->subclass.codec;
00724          }
00725          /* Transcode from our middle (signed linear) frame to new format of the frame that came in */
00726          if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
00727             /* Failed to transcode the frame... drop it and return the original */
00728             ast_frfree(middle_frame);
00729             return start_frame;
00730          }
00731          /* Here's the scoop... middle frame is no longer of use to us */
00732          ast_frfree(middle_frame);
00733       }
00734    } else {
00735       /* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
00736       ast_frfree(middle_frame);
00737    }
00738 
00739    return end_frame;
00740 }
00741 
00742 int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
00743 {
00744    if (AST_LIST_EMPTY(&audiohook_list->spy_list) &&
00745       AST_LIST_EMPTY(&audiohook_list->whisper_list) &&
00746       AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
00747 
00748       return 1;
00749    }
00750    return 0;
00751 }
00752 
00753 /*! \brief Pass a frame off to be handled by the audiohook core
00754  * \param chan Channel that the list is coming off of
00755  * \param audiohook_list List of audiohooks
00756  * \param direction Direction frame is coming in from
00757  * \param frame The frame itself
00758  * \return Return frame on success, NULL on failure
00759  */
00760 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)
00761 {
00762    /* Pass off frame to it's respective list write function */
00763    if (frame->frametype == AST_FRAME_VOICE)
00764       return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
00765    else if (frame->frametype == AST_FRAME_DTMF)
00766       return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
00767    else
00768       return frame;
00769 }
00770 
00771 /*! \brief Wait for audiohook trigger to be triggered
00772  * \param audiohook Audiohook to wait on
00773  */
00774 void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
00775 {
00776    struct timeval wait;
00777    struct timespec ts;
00778 
00779    wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
00780    ts.tv_sec = wait.tv_sec;
00781    ts.tv_nsec = wait.tv_usec * 1000;
00782    
00783    ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
00784    
00785    return;
00786 }
00787 
00788 /* Count number of channel audiohooks by type, regardless of type */
00789 int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
00790 {
00791    int count = 0;
00792    struct ast_audiohook *ah = NULL;
00793 
00794    if (!chan->audiohooks)
00795       return -1;
00796 
00797    switch (type) {
00798       case AST_AUDIOHOOK_TYPE_SPY:
00799          AST_LIST_TRAVERSE(&chan->audiohooks->spy_list, ah, list) {
00800             if (!strcmp(ah->source, source)) {
00801                count++;
00802             }
00803          }
00804          break;
00805       case AST_AUDIOHOOK_TYPE_WHISPER:
00806          AST_LIST_TRAVERSE(&chan->audiohooks->whisper_list, ah, list) {
00807             if (!strcmp(ah->source, source)) {
00808                count++;
00809             }
00810          }
00811          break;
00812       case AST_AUDIOHOOK_TYPE_MANIPULATE:
00813          AST_LIST_TRAVERSE(&chan->audiohooks->manipulate_list, ah, list) {
00814             if (!strcmp(ah->source, source)) {
00815                count++;
00816             }
00817          }
00818          break;
00819       default:
00820          ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%d)\n", type);
00821          return -1;
00822    }
00823 
00824    return count;
00825 }
00826 
00827 /* Count number of channel audiohooks by type that are running */
00828 int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
00829 {
00830    int count = 0;
00831    struct ast_audiohook *ah = NULL;
00832    if (!chan->audiohooks)
00833       return -1;
00834 
00835    switch (type) {
00836       case AST_AUDIOHOOK_TYPE_SPY:
00837          AST_LIST_TRAVERSE(&chan->audiohooks->spy_list, ah, list) {
00838             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00839                count++;
00840          }
00841          break;
00842       case AST_AUDIOHOOK_TYPE_WHISPER:
00843          AST_LIST_TRAVERSE(&chan->audiohooks->whisper_list, ah, list) {
00844             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00845                count++;
00846          }
00847          break;
00848       case AST_AUDIOHOOK_TYPE_MANIPULATE:
00849          AST_LIST_TRAVERSE(&chan->audiohooks->manipulate_list, ah, list) {
00850             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00851                count++;
00852          }
00853          break;
00854       default:
00855          ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%d)\n", type);
00856          return -1;
00857    }
00858    return count;
00859 }
00860 
00861 /*! \brief Audiohook volume adjustment structure */
00862 struct audiohook_volume {
00863    struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
00864    int read_adjustment;            /*!< Value to adjust frames read from the channel by */
00865    int write_adjustment;           /*!< Value to adjust frames written to the channel by */
00866 };
00867 
00868 /*! \brief Callback used to destroy the audiohook volume datastore
00869  * \param data Volume information structure
00870  * \return Returns nothing
00871  */
00872 static void audiohook_volume_destroy(void *data)
00873 {
00874    struct audiohook_volume *audiohook_volume = data;
00875 
00876    /* Destroy the audiohook as it is no longer in use */
00877    ast_audiohook_destroy(&audiohook_volume->audiohook);
00878 
00879    /* Finally free ourselves, we are of no more use */
00880    ast_free(audiohook_volume);
00881 
00882    return;
00883 }
00884 
00885 /*! \brief Datastore used to store audiohook volume information */
00886 static const struct ast_datastore_info audiohook_volume_datastore = {
00887    .type = "Volume",
00888    .destroy = audiohook_volume_destroy,
00889 };
00890 
00891 /*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
00892  * \param audiohook Audiohook attached to the channel
00893  * \param chan Channel we are attached to
00894  * \param frame Frame of audio we want to manipulate
00895  * \param direction Direction the audio came in from
00896  * \return Returns 0 on success, -1 on failure
00897  */
00898 static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
00899 {
00900    struct ast_datastore *datastore = NULL;
00901    struct audiohook_volume *audiohook_volume = NULL;
00902    int *gain = NULL;
00903 
00904    /* If the audiohook is shutting down don't even bother */
00905    if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
00906       return 0;
00907    }
00908 
00909    /* Try to find the datastore containg adjustment information, if we can't just bail out */
00910    if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
00911       return 0;
00912    }
00913 
00914    audiohook_volume = datastore->data;
00915 
00916    /* Based on direction grab the appropriate adjustment value */
00917    if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
00918       gain = &audiohook_volume->read_adjustment;
00919    } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
00920       gain = &audiohook_volume->write_adjustment;
00921    }
00922 
00923    /* If an adjustment value is present modify the frame */
00924    if (gain && *gain) {
00925       ast_frame_adjust_volume(frame, *gain);
00926    }
00927 
00928    return 0;
00929 }
00930 
00931 /*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
00932  * \param chan Channel to look on
00933  * \param create Whether to create the datastore if not found
00934  * \return Returns audiohook_volume structure on success, NULL on failure
00935  */
00936 static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
00937 {
00938    struct ast_datastore *datastore = NULL;
00939    struct audiohook_volume *audiohook_volume = NULL;
00940 
00941    /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
00942    if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
00943       return datastore->data;
00944    }
00945 
00946    /* 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 */
00947    if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
00948       return NULL;
00949    }
00950 
00951    /* Create a new audiohook_volume structure to contain our adjustments and audiohook */
00952    if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
00953       ast_datastore_free(datastore);
00954       return NULL;
00955    }
00956 
00957    /* Setup our audiohook structure so we can manipulate the audio */
00958    ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume");
00959    audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback;
00960 
00961    /* Attach the audiohook_volume blob to the datastore and attach to the channel */
00962    datastore->data = audiohook_volume;
00963    ast_channel_datastore_add(chan, datastore);
00964 
00965    /* All is well... put the audiohook into motion */
00966    ast_audiohook_attach(chan, &audiohook_volume->audiohook);
00967 
00968    return audiohook_volume;
00969 }
00970 
00971 /*! \brief Adjust the volume on frames read from or written to a channel
00972  * \param chan Channel to muck with
00973  * \param direction Direction to set on
00974  * \param volume Value to adjust the volume by
00975  * \return Returns 0 on success, -1 on failure
00976  */
00977 int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
00978 {
00979    struct audiohook_volume *audiohook_volume = NULL;
00980 
00981    /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
00982    if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
00983       return -1;
00984    }
00985 
00986    /* Now based on the direction set the proper value */
00987    if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
00988       audiohook_volume->read_adjustment = volume;
00989    }
00990    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
00991       audiohook_volume->write_adjustment = volume;
00992    }
00993 
00994    return 0;
00995 }
00996 
00997 /*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
00998  * \param chan Channel to retrieve volume adjustment from
00999  * \param direction Direction to retrieve
01000  * \return Returns adjustment value
01001  */
01002 int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
01003 {
01004    struct audiohook_volume *audiohook_volume = NULL;
01005    int adjustment = 0;
01006 
01007    /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
01008    if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
01009       return 0;
01010    }
01011 
01012    /* Grab the adjustment value based on direction given */
01013    if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
01014       adjustment = audiohook_volume->read_adjustment;
01015    } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
01016       adjustment = audiohook_volume->write_adjustment;
01017    }
01018 
01019    return adjustment;
01020 }
01021 
01022 /*! \brief Adjust the volume on frames read from or written to a channel
01023  * \param chan Channel to muck with
01024  * \param direction Direction to increase
01025  * \param volume Value to adjust the adjustment by
01026  * \return Returns 0 on success, -1 on failure
01027  */
01028 int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
01029 {
01030    struct audiohook_volume *audiohook_volume = NULL;
01031 
01032    /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
01033    if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
01034       return -1;
01035    }
01036 
01037    /* Based on the direction change the specific adjustment value */
01038    if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
01039       audiohook_volume->read_adjustment += volume;
01040    }
01041    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
01042       audiohook_volume->write_adjustment += volume;
01043    }
01044 
01045    return 0;
01046 }
01047 
01048 /*! \brief Mute frames read from or written to a channel
01049  * \param chan Channel to muck with
01050  * \param source Type of audiohook
01051  * \param flag which flag to set / clear
01052  * \param clear set or clear
01053  * \return Returns 0 on success, -1 on failure
01054  */
01055 int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
01056 {
01057    struct ast_audiohook *audiohook = NULL;
01058 
01059    ast_channel_lock(chan);
01060 
01061    /* Ensure the channel has audiohooks on it */
01062    if (!chan->audiohooks) {
01063       ast_channel_unlock(chan);
01064       return -1;
01065    }
01066 
01067    audiohook = find_audiohook_by_source(chan->audiohooks, source);
01068 
01069    if (audiohook) {
01070       if (clear) {
01071          ast_clear_flag(audiohook, flag);
01072       } else {
01073          ast_set_flag(audiohook, flag);
01074       }
01075    }
01076 
01077    ast_channel_unlock(chan);
01078 
01079    return (audiohook ? 0 : -1);
01080 }

Generated on Sat Mar 10 01:54:05 2012 for Asterisk - The Open Source Telephony Project by  doxygen 1.4.7