Wed Aug 18 22:33:44 2010

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 'file' Colp <jcolp@digium.com>
00024  */
00025 
00026 #include "asterisk.h"
00027 
00028 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 260052 $")
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    int 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 
00131    /* Update last feeding time to be current */
00132    *rwtime = ast_tvnow();
00133 
00134    our_factory_samples = ast_slinfactory_available(factory);
00135    our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / 8);
00136    other_factory_samples = ast_slinfactory_available(other_factory);
00137    other_factory_ms = other_factory_samples / 8;
00138 
00139    if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && other_factory_samples && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
00140       if (option_debug)
00141          ast_log(LOG_DEBUG, "Flushing audiohook %p so it remains in sync\n", audiohook);
00142       ast_slinfactory_flush(factory);
00143       ast_slinfactory_flush(other_factory);
00144    }
00145 
00146    if (ast_test_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE) && (our_factory_samples > 640 || other_factory_samples > 640)) {
00147       if (option_debug) {
00148          ast_log(LOG_DEBUG, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
00149       }
00150       ast_slinfactory_flush(factory);
00151       ast_slinfactory_flush(other_factory);
00152    }
00153 
00154    /* Write frame out to respective factory */
00155    ast_slinfactory_feed(factory, frame);
00156 
00157    /* If we need to notify the respective handler of this audiohook, do so */
00158    if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
00159       ast_cond_signal(&audiohook->trigger);
00160    } else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
00161       ast_cond_signal(&audiohook->trigger);
00162    } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
00163       ast_cond_signal(&audiohook->trigger);
00164    }
00165 
00166    return 0;
00167 }
00168 
00169 static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
00170 {
00171    struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
00172    int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
00173    short buf[samples];
00174    struct ast_frame frame = {
00175       .frametype = AST_FRAME_VOICE,
00176       .subclass = AST_FORMAT_SLINEAR,
00177       .data.ptr = buf,
00178       .datalen = sizeof(buf),
00179       .samples = samples,
00180    };
00181 
00182    /* Ensure the factory is able to give us the samples we want */
00183    if (samples > ast_slinfactory_available(factory))
00184       return NULL;
00185    
00186    /* Read data in from factory */
00187    if (!ast_slinfactory_read(factory, buf, samples))
00188       return NULL;
00189 
00190    /* If a volume adjustment needs to be applied apply it */
00191    if (vol)
00192       ast_frame_adjust_volume(&frame, vol);
00193 
00194    return ast_frdup(&frame);
00195 }
00196 
00197 static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
00198 {
00199    int i = 0, usable_read, usable_write;
00200    short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
00201    struct ast_frame frame = {
00202       .frametype = AST_FRAME_VOICE,
00203       .subclass = AST_FORMAT_SLINEAR,
00204       .data.ptr = NULL,
00205       .datalen = sizeof(buf1),
00206       .samples = samples,
00207    };
00208 
00209    /* Make sure both factories have the required samples */
00210    usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
00211    usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
00212 
00213    if (!usable_read && !usable_write) {
00214       /* If both factories are unusable bail out */
00215       ast_debug(1, "Read factory %p and write factory %p both fail to provide %zd samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
00216       return NULL;
00217    }
00218 
00219    /* If we want to provide only a read factory make sure we aren't waiting for other audio */
00220    if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
00221       ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
00222       return NULL;
00223    }
00224 
00225    /* If we want to provide only a write factory make sure we aren't waiting for other audio */
00226    if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
00227       ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
00228       return NULL;
00229    }
00230 
00231    /* Start with the read factory... if there are enough samples, read them in */
00232    if (usable_read) {
00233       if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
00234          read_buf = buf1;
00235          /* Adjust read volume if need be */
00236          if (audiohook->options.read_volume) {
00237             int count = 0;
00238             short adjust_value = abs(audiohook->options.read_volume);
00239             for (count = 0; count < samples; count++) {
00240                if (audiohook->options.read_volume > 0)
00241                   ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
00242                else if (audiohook->options.read_volume < 0)
00243                   ast_slinear_saturated_divide(&buf1[count], &adjust_value);
00244             }
00245          }
00246       }
00247    } else if (option_debug)
00248       ast_log(LOG_DEBUG, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
00249 
00250    /* Move on to the write factory... if there are enough samples, read them in */
00251    if (usable_write) {
00252       if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
00253          write_buf = buf2;
00254          /* Adjust write volume if need be */
00255          if (audiohook->options.write_volume) {
00256             int count = 0;
00257             short adjust_value = abs(audiohook->options.write_volume);
00258             for (count = 0; count < samples; count++) {
00259                if (audiohook->options.write_volume > 0)
00260                   ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
00261                else if (audiohook->options.write_volume < 0)
00262                   ast_slinear_saturated_divide(&buf2[count], &adjust_value);
00263             }
00264          }
00265       }
00266    } else if (option_debug)
00267       ast_log(LOG_DEBUG, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
00268 
00269    /* Basically we figure out which buffer to use... and if mixing can be done here */
00270    if (!read_buf && !write_buf)
00271       return NULL;
00272    else if (read_buf && write_buf) {
00273       for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
00274          ast_slinear_saturated_add(data1, data2);
00275       final_buf = buf1;
00276    } else if (read_buf)
00277       final_buf = buf1;
00278    else if (write_buf)
00279       final_buf = buf2;
00280 
00281    /* Make the final buffer part of the frame, so it gets duplicated fine */
00282    frame.data.ptr = final_buf;
00283 
00284    /* Yahoo, a combined copy of the audio! */
00285    return ast_frdup(&frame);
00286 }
00287 
00288 /*! \brief Reads a frame in from the audiohook structure
00289  * \param audiohook Audiohook structure
00290  * \param samples Number of samples wanted
00291  * \param direction Direction the audio frame came from
00292  * \param format Format of frame remote side wants back
00293  * \return Returns frame on success, NULL on failure
00294  */
00295 struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format)
00296 {
00297    struct ast_frame *read_frame = NULL, *final_frame = NULL;
00298 
00299    if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
00300       return NULL;
00301 
00302    /* If they don't want signed linear back out, we'll have to send it through the translation path */
00303    if (format != AST_FORMAT_SLINEAR) {
00304       /* Rebuild translation path if different format then previously */
00305       if (audiohook->format != format) {
00306          if (audiohook->trans_pvt) {
00307             ast_translator_free_path(audiohook->trans_pvt);
00308             audiohook->trans_pvt = NULL;
00309          }
00310          /* 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 */
00311          if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
00312             ast_frfree(read_frame);
00313             return NULL;
00314          }
00315       }
00316       /* Convert to requested format, and allow the read in frame to be freed */
00317       final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
00318    } else {
00319       final_frame = read_frame;
00320    }
00321 
00322    return final_frame;
00323 }
00324 
00325 /*! \brief Attach audiohook to channel
00326  * \param chan Channel
00327  * \param audiohook Audiohook structure
00328  * \return Returns 0 on success, -1 on failure
00329  */
00330 int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
00331 {
00332    ast_channel_lock(chan);
00333 
00334    if (!chan->audiohooks) {
00335       /* Whoops... allocate a new structure */
00336       if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
00337          ast_channel_unlock(chan);
00338          return -1;
00339       }
00340       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
00341       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
00342       AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
00343    }
00344 
00345    /* Drop into respective list */
00346    if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
00347       AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
00348    else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
00349       AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
00350    else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
00351       AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
00352 
00353    /* Change status over to running since it is now attached */
00354    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
00355 
00356    ast_channel_unlock(chan);
00357 
00358    return 0;
00359 }
00360 
00361 /*! \brief Update audiohook's status
00362  * \param audiohook status enum
00363  * \param audiohook Audiohook structure
00364  *
00365  * \note once status is updated to DONE, this function can not be used to set the
00366  * status back to any other setting.  Setting DONE effectively locks the status as such.
00367  */
00368 
00369 void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
00370 {
00371    ast_audiohook_lock(audiohook);
00372    if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
00373       audiohook->status = status;
00374       ast_cond_signal(&audiohook->trigger);
00375    }
00376    ast_audiohook_unlock(audiohook);
00377 }
00378 
00379 /*! \brief Detach audiohook from channel
00380  * \param audiohook Audiohook structure
00381  * \return Returns 0 on success, -1 on failure
00382  */
00383 int ast_audiohook_detach(struct ast_audiohook *audiohook)
00384 {
00385    if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
00386       return 0;
00387 
00388    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
00389 
00390    while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
00391       ast_audiohook_trigger_wait(audiohook);
00392 
00393    return 0;
00394 }
00395 
00396 /*! \brief Detach audiohooks from list and destroy said list
00397  * \param audiohook_list List of audiohooks
00398  * \return Returns 0 on success, -1 on failure
00399  */
00400 int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
00401 {
00402    int i = 0;
00403    struct ast_audiohook *audiohook = NULL;
00404 
00405    /* Drop any spies */
00406    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
00407       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00408    }
00409 
00410    /* Drop any whispering sources */
00411    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
00412       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00413    }
00414 
00415    /* Drop any manipulaters */
00416    while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
00417       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00418       audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
00419    }
00420 
00421    /* Drop translation paths if present */
00422    for (i = 0; i < 2; i++) {
00423       if (audiohook_list->in_translate[i].trans_pvt)
00424          ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
00425       if (audiohook_list->out_translate[i].trans_pvt)
00426          ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
00427    }
00428    
00429    /* Free ourselves */
00430    ast_free(audiohook_list);
00431 
00432    return 0;
00433 }
00434 
00435 /*! \brief find an audiohook based on its source
00436  * \param audiohook_list The list of audiohooks to search in
00437  * \param source The source of the audiohook we wish to find
00438  * \return Return the corresponding audiohook or NULL if it cannot be found.
00439  */
00440 static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
00441 {
00442    struct ast_audiohook *audiohook = NULL;
00443 
00444    AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
00445       if (!strcasecmp(audiohook->source, source))
00446          return audiohook;
00447    }
00448 
00449    AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
00450       if (!strcasecmp(audiohook->source, source))
00451          return audiohook;
00452    }
00453 
00454    AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
00455       if (!strcasecmp(audiohook->source, source))
00456          return audiohook;
00457    }
00458 
00459    return NULL;
00460 }
00461 
00462 void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
00463 {
00464    struct ast_audiohook *audiohook;
00465    enum ast_audiohook_status oldstatus;
00466 
00467    if (!old_chan->audiohooks || !(audiohook = find_audiohook_by_source(old_chan->audiohooks, source))) {
00468       return;
00469    }
00470 
00471    /* By locking both channels and the audiohook, we can assure that
00472     * another thread will not have a chance to read the audiohook's status
00473     * as done, even though ast_audiohook_remove signals the trigger
00474     * condition.
00475     */
00476    ast_audiohook_lock(audiohook);
00477    oldstatus = audiohook->status;
00478 
00479    ast_audiohook_remove(old_chan, audiohook);
00480    ast_audiohook_attach(new_chan, audiohook);
00481 
00482    audiohook->status = oldstatus;
00483    ast_audiohook_unlock(audiohook);
00484 }
00485 
00486 /*! \brief Detach specified source audiohook from channel
00487  * \param chan Channel to detach from
00488  * \param source Name of source to detach
00489  * \return Returns 0 on success, -1 on failure
00490  */
00491 int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
00492 {
00493    struct ast_audiohook *audiohook = NULL;
00494 
00495    ast_channel_lock(chan);
00496 
00497    /* Ensure the channel has audiohooks on it */
00498    if (!chan->audiohooks) {
00499       ast_channel_unlock(chan);
00500       return -1;
00501    }
00502 
00503    audiohook = find_audiohook_by_source(chan->audiohooks, source);
00504 
00505    ast_channel_unlock(chan);
00506 
00507    if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
00508       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
00509 
00510    return (audiohook ? 0 : -1);
00511 }
00512 
00513 /*!
00514  * \brief Remove an audiohook from a specified channel
00515  *
00516  * \param chan Channel to remove from
00517  * \param audiohook Audiohook to remove
00518  *
00519  * \return Returns 0 on success, -1 on failure
00520  *
00521  * \note The channel does not need to be locked before calling this function
00522  */
00523 int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
00524 {
00525    ast_channel_lock(chan);
00526 
00527    if (!chan->audiohooks) {
00528       ast_channel_unlock(chan);
00529       return -1;
00530    }
00531 
00532    if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
00533       AST_LIST_REMOVE(&chan->audiohooks->spy_list, audiohook, list);
00534    else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
00535       AST_LIST_REMOVE(&chan->audiohooks->whisper_list, audiohook, list);
00536    else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
00537       AST_LIST_REMOVE(&chan->audiohooks->manipulate_list, audiohook, list);
00538 
00539    ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00540 
00541    ast_channel_unlock(chan);
00542 
00543    return 0;
00544 }
00545 
00546 /*! \brief Pass a DTMF frame off to be handled by the audiohook core
00547  * \param chan Channel that the list is coming off of
00548  * \param audiohook_list List of audiohooks
00549  * \param direction Direction frame is coming in from
00550  * \param frame The frame itself
00551  * \return Return frame on success, NULL on failure
00552  */
00553 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)
00554 {
00555    struct ast_audiohook *audiohook = NULL;
00556 
00557    AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
00558       ast_audiohook_lock(audiohook);
00559       if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00560          AST_LIST_REMOVE_CURRENT(list);
00561          ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00562          ast_audiohook_unlock(audiohook);
00563          audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
00564          continue;
00565       }
00566       if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
00567          audiohook->manipulate_callback(audiohook, chan, frame, direction);
00568       ast_audiohook_unlock(audiohook);
00569    }
00570    AST_LIST_TRAVERSE_SAFE_END;
00571 
00572    return frame;
00573 }
00574 
00575 /*!
00576  * \brief Pass an AUDIO frame off to be handled by the audiohook core
00577  *
00578  * \details
00579  * This function has 3 ast_frames and 3 parts to handle each.  At the beginning of this
00580  * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
00581  * input frame.
00582  *
00583  * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
00584  *         format.  The result of this part is middle_frame is guaranteed to be in
00585  *         SLINEAR format for Part_2.
00586  * Part_2: Send middle_frame off to spies and manipulators.  At this point middle_frame is
00587  *         either a new frame as result of the translation, or points directly to the start_frame
00588  *         because no translation to SLINEAR audio was required.  The result of this part
00589  *         is end_frame will be updated to point to middle_frame if any audiohook manipulation
00590  *         took place.
00591  * Part_3: Translate end_frame's audio back into the format of start frame if necessary.
00592  *         At this point if middle_frame != end_frame, we are guaranteed that no manipulation
00593  *         took place and middle_frame can be freed as it was translated... If middle_frame was
00594  *         not translated and still pointed to start_frame, it would be equal to end_frame as well
00595  *         regardless if manipulation took place which would not result in this free.  The result
00596  *         of this part is end_frame is guaranteed to be the format of start_frame for the return.
00597  *         
00598  * \param chan Channel that the list is coming off of
00599  * \param audiohook_list List of audiohooks
00600  * \param direction Direction frame is coming in from
00601  * \param frame The frame itself
00602  * \return Return frame on success, NULL on failure
00603  */
00604 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)
00605 {
00606    struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
00607    struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
00608    struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
00609    struct ast_audiohook *audiohook = NULL;
00610    int samples = frame->samples;
00611 
00612    /* ---Part_1. translate start_frame to SLINEAR if necessary. */
00613    /* If the frame coming in is not signed linear we have to send it through the in_translate path */
00614    if (frame->subclass != AST_FORMAT_SLINEAR) {
00615       if (in_translate->format != frame->subclass) {
00616          if (in_translate->trans_pvt)
00617             ast_translator_free_path(in_translate->trans_pvt);
00618          if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass)))
00619             return frame;
00620          in_translate->format = frame->subclass;
00621       }
00622       if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
00623          return frame;
00624       samples = middle_frame->samples;
00625    }
00626 
00627    /* ---Part_2: Send middle_frame to spy and manipulator lists.  middle_frame is guaranteed to be SLINEAR here.*/
00628    /* Queue up signed linear frame to each spy */
00629    AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
00630       ast_audiohook_lock(audiohook);
00631       if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00632          AST_LIST_REMOVE_CURRENT(list);
00633          ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00634          ast_audiohook_unlock(audiohook);
00635          continue;
00636       }
00637       ast_audiohook_write_frame(audiohook, direction, middle_frame);
00638       ast_audiohook_unlock(audiohook);
00639    }
00640    AST_LIST_TRAVERSE_SAFE_END;
00641 
00642    /* If this frame is being written out to the channel then we need to use whisper sources */
00643    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
00644       int i = 0;
00645       short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
00646       memset(&combine_buf, 0, sizeof(combine_buf));
00647       AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_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          if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
00656             /* Take audio from this whisper source and combine it into our main buffer */
00657             for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
00658                ast_slinear_saturated_add(data1, data2);
00659          }
00660          ast_audiohook_unlock(audiohook);
00661       }
00662       AST_LIST_TRAVERSE_SAFE_END;
00663       /* We take all of the combined whisper sources and combine them into the audio being written out */
00664       for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++)
00665          ast_slinear_saturated_add(data1, data2);
00666       end_frame = middle_frame;
00667    }
00668 
00669    /* Pass off frame to manipulate audiohooks */
00670    if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
00671       AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
00672          ast_audiohook_lock(audiohook);
00673          if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
00674             AST_LIST_REMOVE_CURRENT(list);
00675             ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
00676             ast_audiohook_unlock(audiohook);
00677             /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
00678             audiohook->manipulate_callback(audiohook, chan, NULL, direction);
00679             continue;
00680          }
00681          /* Feed in frame to manipulation. */
00682          if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
00683             /* XXX IGNORE FAILURE */
00684 
00685             /* If the manipulation fails then the frame will be returned in its original state.
00686              * Since there are potentially more manipulator callbacks in the list, no action should
00687              * be taken here to exit early. */
00688          }
00689          ast_audiohook_unlock(audiohook);
00690       }
00691       AST_LIST_TRAVERSE_SAFE_END;
00692       end_frame = middle_frame;
00693    }
00694 
00695    /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
00696    if (middle_frame == end_frame) {
00697       /* Middle frame was modified and became the end frame... let's see if we need to transcode */
00698       if (end_frame->subclass != start_frame->subclass) {
00699          if (out_translate->format != start_frame->subclass) {
00700             if (out_translate->trans_pvt)
00701                ast_translator_free_path(out_translate->trans_pvt);
00702             if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass, AST_FORMAT_SLINEAR))) {
00703                /* We can't transcode this... drop our middle frame and return the original */
00704                ast_frfree(middle_frame);
00705                return start_frame;
00706             }
00707             out_translate->format = start_frame->subclass;
00708          }
00709          /* Transcode from our middle (signed linear) frame to new format of the frame that came in */
00710          if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
00711             /* Failed to transcode the frame... drop it and return the original */
00712             ast_frfree(middle_frame);
00713             return start_frame;
00714          }
00715          /* Here's the scoop... middle frame is no longer of use to us */
00716          ast_frfree(middle_frame);
00717       }
00718    } else {
00719       /* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
00720       ast_frfree(middle_frame);
00721    }
00722 
00723    return end_frame;
00724 }
00725 
00726 /*! \brief Pass a frame off to be handled by the audiohook core
00727  * \param chan Channel that the list is coming off of
00728  * \param audiohook_list List of audiohooks
00729  * \param direction Direction frame is coming in from
00730  * \param frame The frame itself
00731  * \return Return frame on success, NULL on failure
00732  */
00733 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)
00734 {
00735    /* Pass off frame to it's respective list write function */
00736    if (frame->frametype == AST_FRAME_VOICE)
00737       return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
00738    else if (frame->frametype == AST_FRAME_DTMF)
00739       return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
00740    else
00741       return frame;
00742 }
00743          
00744 
00745 /*! \brief Wait for audiohook trigger to be triggered
00746  * \param audiohook Audiohook to wait on
00747  */
00748 void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
00749 {
00750    struct timeval wait;
00751    struct timespec ts;
00752 
00753    wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
00754    ts.tv_sec = wait.tv_sec;
00755    ts.tv_nsec = wait.tv_usec * 1000;
00756    
00757    ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
00758    
00759    return;
00760 }
00761 
00762 /* Count number of channel audiohooks by type, regardless of type */
00763 int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
00764 {
00765    int count = 0;
00766    struct ast_audiohook *ah = NULL;
00767 
00768    if (!chan->audiohooks)
00769       return -1;
00770 
00771    switch (type) {
00772       case AST_AUDIOHOOK_TYPE_SPY:
00773          AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->spy_list, ah, list) {
00774             if (!strcmp(ah->source, source)) {
00775                count++;
00776             }
00777          }
00778          AST_LIST_TRAVERSE_SAFE_END;
00779          break;
00780       case AST_AUDIOHOOK_TYPE_WHISPER:
00781          AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->whisper_list, ah, list) {
00782             if (!strcmp(ah->source, source)) {
00783                count++;
00784             }
00785          }
00786          AST_LIST_TRAVERSE_SAFE_END;
00787          break;
00788       case AST_AUDIOHOOK_TYPE_MANIPULATE:
00789          AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->manipulate_list, ah, list) {
00790             if (!strcmp(ah->source, source)) {
00791                count++;
00792             }
00793          }
00794          AST_LIST_TRAVERSE_SAFE_END;
00795          break;
00796       default:
00797          ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%d)\n", type);
00798          return -1;
00799    }
00800 
00801    return count;
00802 }
00803 
00804 /* Count number of channel audiohooks by type that are running */
00805 int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
00806 {
00807    int count = 0;
00808    struct ast_audiohook *ah = NULL;
00809    if (!chan->audiohooks)
00810       return -1;
00811 
00812    switch (type) {
00813       case AST_AUDIOHOOK_TYPE_SPY:
00814          AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->spy_list, ah, list) {
00815             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00816                count++;
00817          }
00818          AST_LIST_TRAVERSE_SAFE_END;
00819          break;
00820       case AST_AUDIOHOOK_TYPE_WHISPER:
00821          AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->whisper_list, ah, list) {
00822             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00823                count++;
00824          }
00825          AST_LIST_TRAVERSE_SAFE_END;
00826          break;
00827       case AST_AUDIOHOOK_TYPE_MANIPULATE:
00828          AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->manipulate_list, ah, list) {
00829             if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
00830                count++;
00831          }
00832          AST_LIST_TRAVERSE_SAFE_END;
00833          break;
00834       default:
00835          ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%d)\n", type);
00836          return -1;
00837    }
00838    return count;
00839 }
00840 
00841 /*! \brief Audiohook volume adjustment structure */
00842 struct audiohook_volume {
00843    struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
00844    int read_adjustment;            /*!< Value to adjust frames read from the channel by */
00845    int write_adjustment;           /*!< Value to adjust frames written to the channel by */
00846 };
00847 
00848 /*! \brief Callback used to destroy the audiohook volume datastore
00849  * \param data Volume information structure
00850  * \return Returns nothing
00851  */
00852 static void audiohook_volume_destroy(void *data)
00853 {
00854    struct audiohook_volume *audiohook_volume = data;
00855 
00856    /* Destroy the audiohook as it is no longer in use */
00857    ast_audiohook_destroy(&audiohook_volume->audiohook);
00858 
00859    /* Finally free ourselves, we are of no more use */
00860    ast_free(audiohook_volume);
00861 
00862    return;
00863 }
00864 
00865 /*! \brief Datastore used to store audiohook volume information */
00866 static const struct ast_datastore_info audiohook_volume_datastore = {
00867    .type = "Volume",
00868    .destroy = audiohook_volume_destroy,
00869 };
00870 
00871 /*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
00872  * \param audiohook Audiohook attached to the channel
00873  * \param chan Channel we are attached to
00874  * \param frame Frame of audio we want to manipulate
00875  * \param direction Direction the audio came in from
00876  * \return Returns 0 on success, -1 on failure
00877  */
00878 static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
00879 {
00880    struct ast_datastore *datastore = NULL;
00881    struct audiohook_volume *audiohook_volume = NULL;
00882    int *gain = NULL;
00883 
00884    /* If the audiohook is shutting down don't even bother */
00885    if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
00886       return 0;
00887    }
00888 
00889    /* Try to find the datastore containg adjustment information, if we can't just bail out */
00890    if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
00891       return 0;
00892    }
00893 
00894    audiohook_volume = datastore->data;
00895 
00896    /* Based on direction grab the appropriate adjustment value */
00897    if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
00898       gain = &audiohook_volume->read_adjustment;
00899    } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
00900       gain = &audiohook_volume->write_adjustment;
00901    }
00902 
00903    /* If an adjustment value is present modify the frame */
00904    if (gain && *gain) {
00905       ast_frame_adjust_volume(frame, *gain);
00906    }
00907 
00908    return 0;
00909 }
00910 
00911 /*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
00912  * \param chan Channel to look on
00913  * \param create Whether to create the datastore if not found
00914  * \return Returns audiohook_volume structure on success, NULL on failure
00915  */
00916 static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
00917 {
00918    struct ast_datastore *datastore = NULL;
00919    struct audiohook_volume *audiohook_volume = NULL;
00920 
00921    /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
00922    if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
00923       return datastore->data;
00924    }
00925 
00926    /* 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 */
00927    if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
00928       return NULL;
00929    }
00930 
00931    /* Create a new audiohook_volume structure to contain our adjustments and audiohook */
00932    if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
00933       ast_datastore_free(datastore);
00934       return NULL;
00935    }
00936 
00937    /* Setup our audiohook structure so we can manipulate the audio */
00938    ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume");
00939    audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback;
00940 
00941    /* Attach the audiohook_volume blob to the datastore and attach to the channel */
00942    datastore->data = audiohook_volume;
00943    ast_channel_datastore_add(chan, datastore);
00944 
00945    /* All is well... put the audiohook into motion */
00946    ast_audiohook_attach(chan, &audiohook_volume->audiohook);
00947 
00948    return audiohook_volume;
00949 }
00950 
00951 /*! \brief Adjust the volume on frames read from or written to a channel
00952  * \param chan Channel to muck with
00953  * \param direction Direction to set on
00954  * \param volume Value to adjust the volume by
00955  * \return Returns 0 on success, -1 on failure
00956  */
00957 int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
00958 {
00959    struct audiohook_volume *audiohook_volume = NULL;
00960 
00961    /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
00962    if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
00963       return -1;
00964    }
00965 
00966    /* Now based on the direction set the proper value */
00967    if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
00968       audiohook_volume->read_adjustment = volume;
00969    }
00970    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
00971       audiohook_volume->write_adjustment = volume;
00972    }
00973 
00974    return 0;
00975 }
00976 
00977 /*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
00978  * \param chan Channel to retrieve volume adjustment from
00979  * \param direction Direction to retrieve
00980  * \return Returns adjustment value
00981  */
00982 int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
00983 {
00984    struct audiohook_volume *audiohook_volume = NULL;
00985    int adjustment = 0;
00986 
00987    /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
00988    if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
00989       return 0;
00990    }
00991 
00992    /* Grab the adjustment value based on direction given */
00993    if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
00994       adjustment = audiohook_volume->read_adjustment;
00995    } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
00996       adjustment = audiohook_volume->write_adjustment;
00997    }
00998 
00999    return adjustment;
01000 }
01001 
01002 /*! \brief Adjust the volume on frames read from or written to a channel
01003  * \param chan Channel to muck with
01004  * \param direction Direction to increase
01005  * \param volume Value to adjust the adjustment by
01006  * \return Returns 0 on success, -1 on failure
01007  */
01008 int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
01009 {
01010    struct audiohook_volume *audiohook_volume = NULL;
01011 
01012    /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
01013    if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
01014       return -1;
01015    }
01016 
01017    /* Based on the direction change the specific adjustment value */
01018    if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
01019       audiohook_volume->read_adjustment += volume;
01020    }
01021    if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
01022       audiohook_volume->write_adjustment += volume;
01023    }
01024 
01025    return 0;
01026 }

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