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Asterisk developer's documentation


app_speech_utils.c

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00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 2006, 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 Speech Recognition Utility Applications
00022  *
00023  * \author Joshua Colp <jcolp@digium.com>
00024  *
00025  * \ingroup applications
00026  */
00027 
00028 #include "asterisk.h"
00029 
00030 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 264334 $");
00031 
00032 #include <stdio.h>
00033 #include <stdlib.h>
00034 #include <unistd.h>
00035 #include <string.h>
00036 
00037 #include "asterisk/file.h"
00038 #include "asterisk/logger.h"
00039 #include "asterisk/channel.h"
00040 #include "asterisk/pbx.h"
00041 #include "asterisk/module.h"
00042 #include "asterisk/lock.h"
00043 #include "asterisk/app.h"
00044 #include "asterisk/speech.h"
00045 
00046 /* Descriptions for each application */
00047 static char *speechcreate_descrip =
00048 "SpeechCreate(engine name)\n"
00049 "This application creates information to be used by all the other applications. It must be called before doing any speech recognition activities such as activating a grammar.\n"
00050 "It takes the engine name to use as the argument, if not specified the default engine will be used.\n";
00051 
00052 static char *speechactivategrammar_descrip =
00053 "SpeechActivateGrammar(Grammar Name)\n"
00054 "This activates the specified grammar to be recognized by the engine. A grammar tells the speech recognition engine what to recognize, \n"
00055    "and how to portray it back to you in the dialplan. The grammar name is the only argument to this application.\n";
00056 
00057 static char *speechstart_descrip =
00058 "SpeechStart()\n"
00059    "Tell the speech recognition engine that it should start trying to get results from audio being fed to it. This has no arguments.\n";
00060 
00061 static char *speechbackground_descrip =
00062 "SpeechBackground(Sound File|Timeout)\n"
00063 "This application plays a sound file and waits for the person to speak. Once they start speaking playback of the file stops, and silence is heard.\n"
00064 "Once they stop talking the processing sound is played to indicate the speech recognition engine is working.\n"
00065 "Once results are available the application returns and results (score and text) are available using dialplan functions.\n"
00066 "The first text and score are ${SPEECH_TEXT(0)} AND ${SPEECH_SCORE(0)} while the second are ${SPEECH_TEXT(1)} and ${SPEECH_SCORE(1)}.\n"
00067 "The first argument is the sound file and the second is the timeout. Note the timeout will only start once the sound file has stopped playing.\n";
00068 
00069 static char *speechdeactivategrammar_descrip =
00070 "SpeechDeactivateGrammar(Grammar Name)\n"
00071    "This deactivates the specified grammar so that it is no longer recognized. The only argument is the grammar name to deactivate.\n";
00072 
00073 static char *speechprocessingsound_descrip =
00074 "SpeechProcessingSound(Sound File)\n"
00075 "This changes the processing sound that SpeechBackground plays back when the speech recognition engine is processing and working to get results.\n"
00076    "It takes the sound file as the only argument.\n";
00077 
00078 static char *speechdestroy_descrip =
00079 "SpeechDestroy()\n"
00080 "This destroys the information used by all the other speech recognition applications.\n"
00081 "If you call this application but end up wanting to recognize more speech, you must call SpeechCreate\n"
00082    "again before calling any other application. It takes no arguments.\n";
00083 
00084 static char *speechload_descrip =
00085 "SpeechLoadGrammar(Grammar Name|Path)\n"
00086 "Load a grammar only on the channel, not globally.\n"
00087 "It takes the grammar name as first argument and path as second.\n";
00088 
00089 static char *speechunload_descrip =
00090 "SpeechUnloadGrammar(Grammar Name)\n"
00091 "Unload a grammar. It takes the grammar name as the only argument.\n";
00092 
00093 /*! \brief Helper function used by datastores to destroy the speech structure upon hangup */
00094 static void destroy_callback(void *data)
00095 {
00096    struct ast_speech *speech = (struct ast_speech*)data;
00097 
00098    if (speech == NULL) {
00099       return;
00100    }
00101 
00102    /* Deallocate now */
00103    ast_speech_destroy(speech);
00104 
00105    return;
00106 }
00107 
00108 /*! \brief Static structure for datastore information */
00109 static const struct ast_datastore_info speech_datastore = {
00110    .type = "speech",
00111    .destroy = destroy_callback
00112 };
00113 
00114 /*! \brief Helper function used to find the speech structure attached to a channel */
00115 static struct ast_speech *find_speech(struct ast_channel *chan)
00116 {
00117    struct ast_speech *speech = NULL;
00118    struct ast_datastore *datastore = NULL;
00119    
00120    datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
00121    if (datastore == NULL) {
00122       return NULL;
00123    }
00124    speech = datastore->data;
00125 
00126    return speech;
00127 }
00128 
00129 /* Helper function to find a specific speech recognition result by number and nbest alternative */
00130 static struct ast_speech_result *find_result(struct ast_speech_result *results, char *result_num)
00131 {
00132    struct ast_speech_result *result = results;
00133    char *tmp = NULL;
00134    int nbest_num = 0, wanted_num = 0, i = 0;
00135 
00136    if (!result)
00137       return NULL;
00138 
00139    if ((tmp = strchr(result_num, '/'))) {
00140       *tmp++ = '\0';
00141       nbest_num = atoi(result_num);
00142       wanted_num = atoi(tmp);
00143    } else {
00144       wanted_num = atoi(result_num);
00145    }
00146 
00147    do {
00148       if (result->nbest_num != nbest_num)
00149          continue;
00150       if (i == wanted_num)
00151          break;
00152       i++;
00153    } while ((result = result->next));
00154 
00155    return result;
00156 }
00157 
00158 /*! \brief SPEECH_SCORE() Dialplan Function */
00159 static int speech_score(struct ast_channel *chan, char *cmd, char *data,
00160              char *buf, size_t len)
00161 {
00162    struct ast_speech_result *result = NULL;
00163    struct ast_speech *speech = find_speech(chan);
00164    char tmp[128] = "";
00165 
00166    if (data == NULL || speech == NULL || !(result = find_result(speech->results, data)))
00167       return -1;
00168    
00169    snprintf(tmp, sizeof(tmp), "%d", result->score);
00170    
00171    ast_copy_string(buf, tmp, len);
00172 
00173    return 0;
00174 }
00175 
00176 static struct ast_custom_function speech_score_function = {
00177         .name = "SPEECH_SCORE",
00178         .synopsis = "Gets the confidence score of a result.",
00179         .syntax = "SPEECH_SCORE([nbest number/]result number)",
00180         .desc =
00181         "Gets the confidence score of a result.\n",
00182         .read = speech_score,
00183         .write = NULL,
00184 };
00185 
00186 /*! \brief SPEECH_TEXT() Dialplan Function */
00187 static int speech_text(struct ast_channel *chan, char *cmd, char *data,
00188          char *buf, size_t len)
00189 {
00190         struct ast_speech_result *result = NULL;
00191         struct ast_speech *speech = find_speech(chan);
00192 
00193    if (data == NULL || speech == NULL || !(result = find_result(speech->results, data)))
00194                 return -1;
00195 
00196    if (result->text != NULL) {
00197       ast_copy_string(buf, result->text, len);
00198    } else {
00199       buf[0] = '\0';
00200    }
00201 
00202         return 0;
00203 }
00204 
00205 static struct ast_custom_function speech_text_function = {
00206         .name = "SPEECH_TEXT",
00207         .synopsis = "Gets the recognized text of a result.",
00208         .syntax = "SPEECH_TEXT([nbest number/]result number)",
00209         .desc =
00210         "Gets the recognized text of a result.\n",
00211         .read = speech_text,
00212         .write = NULL,
00213 };
00214 
00215 /*! \brief SPEECH_GRAMMAR() Dialplan Function */
00216 static int speech_grammar(struct ast_channel *chan, char *cmd, char *data,
00217          char *buf, size_t len)
00218 {
00219         struct ast_speech_result *result = NULL;
00220         struct ast_speech *speech = find_speech(chan);
00221 
00222    if (data == NULL || speech == NULL || !(result = find_result(speech->results, data)))
00223                 return -1;
00224 
00225    if (result->grammar != NULL) {
00226       ast_copy_string(buf, result->grammar, len);
00227    } else {
00228       buf[0] = '\0';
00229    }
00230 
00231         return 0;
00232 }
00233 
00234 static struct ast_custom_function speech_grammar_function = {
00235         .name = "SPEECH_GRAMMAR",
00236         .synopsis = "Gets the matched grammar of a result if available.",
00237         .syntax = "SPEECH_GRAMMAR([nbest number/]result number)",
00238         .desc =
00239         "Gets the matched grammar of a result if available.\n",
00240         .read = speech_grammar,
00241         .write = NULL,
00242 };
00243 
00244 /*! \brief SPEECH_ENGINE() Dialplan Function */
00245 static int speech_engine_write(struct ast_channel *chan, char *cmd, char *data, const char *value)
00246 {
00247    struct ast_speech *speech = find_speech(chan);
00248 
00249    if (data == NULL || speech == NULL)
00250       return -1;
00251 
00252    ast_speech_change(speech, data, value);
00253 
00254    return 0;
00255 }
00256 
00257 static struct ast_custom_function speech_engine_function = {
00258    .name = "SPEECH_ENGINE",
00259    .synopsis = "Change a speech engine specific attribute.",
00260    .syntax = "SPEECH_ENGINE(name)=value",
00261    .desc =
00262    "Changes a speech engine specific attribute.\n",
00263    .read = NULL,
00264    .write = speech_engine_write,
00265 };
00266 
00267 /*! \brief SPEECH_RESULTS_TYPE() Dialplan Function */
00268 static int speech_results_type_write(struct ast_channel *chan, char *cmd, char *data, const char *value)
00269 {
00270    struct ast_speech *speech = find_speech(chan);
00271 
00272    if (data == NULL || speech == NULL)
00273       return -1;
00274 
00275    if (!strcasecmp(value, "normal"))
00276       ast_speech_change_results_type(speech, AST_SPEECH_RESULTS_TYPE_NORMAL);
00277    else if (!strcasecmp(value, "nbest"))
00278       ast_speech_change_results_type(speech, AST_SPEECH_RESULTS_TYPE_NBEST);
00279 
00280    return 0;
00281 }
00282 
00283 static struct ast_custom_function speech_results_type_function = {
00284    .name = "SPEECH_RESULTS_TYPE",
00285    .synopsis = "Sets the type of results that will be returned.",
00286    .syntax = "SPEECH_RESULTS_TYPE()=results type",
00287    .desc =
00288    "Sets the type of results that will be returned. Valid options are normal or nbest.",
00289    .read = NULL,
00290    .write = speech_results_type_write,
00291 };
00292 
00293 /*! \brief SPEECH() Dialplan Function */
00294 static int speech_read(struct ast_channel *chan, char *cmd, char *data,
00295          char *buf, size_t len)
00296 {
00297    int results = 0;
00298    struct ast_speech_result *result = NULL;
00299    struct ast_speech *speech = find_speech(chan);
00300    char tmp[128] = "";
00301 
00302    /* Now go for the various options */
00303    if (!strcasecmp(data, "status")) {
00304       if (speech != NULL)
00305          ast_copy_string(buf, "1", len);
00306       else
00307          ast_copy_string(buf, "0", len);
00308       return 0;
00309    }
00310 
00311    /* Make sure we have a speech structure for everything else */
00312    if (speech == NULL) {
00313       return -1;
00314    }
00315 
00316    /* Check to see if they are checking for silence */
00317    if (!strcasecmp(data, "spoke")) {
00318       if (ast_test_flag(speech, AST_SPEECH_SPOKE))
00319          ast_copy_string(buf, "1", len);
00320       else
00321          ast_copy_string(buf, "0", len);
00322    } else if (!strcasecmp(data, "results")) {
00323       /* Count number of results */
00324       result = speech->results;
00325       while (result) {
00326          results++;
00327          result = result->next;
00328       }
00329       snprintf(tmp, sizeof(tmp), "%d", results);
00330       ast_copy_string(buf, tmp, len);
00331    } else {
00332       buf[0] = '\0';
00333    }
00334 
00335    return 0;
00336 }
00337 
00338 static struct ast_custom_function speech_function = {
00339         .name = "SPEECH",
00340         .synopsis = "Gets information about speech recognition results.",
00341         .syntax = "SPEECH(argument)",
00342         .desc =
00343    "Gets information about speech recognition results.\n"
00344    "status:   Returns 1 upon speech object existing, or 0 if not\n"
00345    "spoke:  Returns 1 if spoker spoke, or 0 if not\n"
00346    "results:  Returns number of results that were recognized\n",
00347         .read = speech_read,
00348         .write = NULL,
00349 };
00350 
00351 
00352 
00353 /*! \brief SpeechCreate() Dialplan Application */
00354 static int speech_create(struct ast_channel *chan, void *data)
00355 {
00356    struct ast_module_user *u = NULL;
00357    struct ast_speech *speech = NULL;
00358    struct ast_datastore *datastore = NULL;
00359 
00360    u = ast_module_user_add(chan);
00361 
00362    /* Request a speech object */
00363    speech = ast_speech_new(data, AST_FORMAT_SLINEAR);
00364    if (speech == NULL) {
00365       /* Not available */
00366       pbx_builtin_setvar_helper(chan, "ERROR", "1");
00367       ast_module_user_remove(u);
00368       return 0;
00369    }
00370 
00371    datastore = ast_channel_datastore_alloc(&speech_datastore, NULL);
00372    if (datastore == NULL) {
00373       ast_speech_destroy(speech);
00374       pbx_builtin_setvar_helper(chan, "ERROR", "1");
00375       ast_module_user_remove(u);
00376       return 0;
00377    }
00378    datastore->data = speech;
00379    ast_channel_datastore_add(chan, datastore);
00380 
00381    pbx_builtin_setvar_helper(chan, "ERROR", NULL);
00382 
00383    ast_module_user_remove(u);
00384 
00385    return 0;
00386 }
00387 
00388 /*! \brief SpeechLoadGrammar(Grammar Name|Path) Dialplan Application */
00389 static int speech_load(struct ast_channel *chan, void *data)
00390 {
00391    int res = 0, argc = 0;
00392    struct ast_module_user *u = NULL;
00393    struct ast_speech *speech = find_speech(chan);
00394    char *argv[2], *args = NULL, *name = NULL, *path = NULL;
00395 
00396    args = ast_strdupa(data);
00397 
00398    u = ast_module_user_add(chan);
00399 
00400    if (speech == NULL) {
00401       ast_module_user_remove(u);
00402                 return -1;
00403         }
00404 
00405    /* Parse out arguments */
00406    argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0]));
00407    if (argc != 2) {
00408       ast_module_user_remove(u);
00409       return -1;
00410    }
00411    name = argv[0];
00412    path = argv[1];
00413 
00414         /* Load the grammar locally on the object */
00415         res = ast_speech_grammar_load(speech, name, path);
00416 
00417         ast_module_user_remove(u);
00418 
00419         return res;
00420 }
00421 
00422 /*! \brief SpeechUnloadGrammar(Grammar Name) Dialplan Application */
00423 static int speech_unload(struct ast_channel *chan, void *data)
00424 {
00425         int res = 0;
00426         struct ast_module_user *u = NULL;
00427         struct ast_speech *speech = find_speech(chan);
00428 
00429         u = ast_module_user_add(chan);
00430 
00431         if (speech == NULL) {
00432                 ast_module_user_remove(u);
00433                 return -1;
00434         }
00435 
00436         /* Unload the grammar */
00437         res = ast_speech_grammar_unload(speech, data);
00438 
00439         ast_module_user_remove(u);
00440 
00441         return res;
00442 }
00443 
00444 /*! \brief SpeechDeactivateGrammar(Grammar Name) Dialplan Application */
00445 static int speech_deactivate(struct ast_channel *chan, void *data)
00446 {
00447         int res = 0;
00448         struct ast_module_user *u = NULL;
00449         struct ast_speech *speech = find_speech(chan);
00450 
00451         u = ast_module_user_add(chan);
00452 
00453         if (speech == NULL) {
00454                 ast_module_user_remove(u);
00455                 return -1;
00456         }
00457 
00458         /* Deactivate the grammar on the speech object */
00459         res = ast_speech_grammar_deactivate(speech, data);
00460 
00461         ast_module_user_remove(u);
00462 
00463         return res;
00464 }
00465 
00466 /*! \brief SpeechActivateGrammar(Grammar Name) Dialplan Application */
00467 static int speech_activate(struct ast_channel *chan, void *data)
00468 {
00469    int res = 0;
00470    struct ast_module_user *u = NULL;
00471    struct ast_speech *speech = find_speech(chan);
00472 
00473    u = ast_module_user_add(chan);
00474 
00475    if (speech == NULL) {
00476       ast_module_user_remove(u);
00477       return -1;
00478    }
00479 
00480    /* Activate the grammar on the speech object */
00481    res = ast_speech_grammar_activate(speech, data);
00482 
00483    ast_module_user_remove(u);
00484 
00485    return res;
00486 }
00487 
00488 /*! \brief SpeechStart() Dialplan Application */
00489 static int speech_start(struct ast_channel *chan, void *data)
00490 {
00491    int res = 0;
00492         struct ast_module_user *u = NULL;
00493    struct ast_speech *speech = find_speech(chan);
00494 
00495    u = ast_module_user_add(chan);
00496 
00497    if (speech == NULL) {
00498       ast_module_user_remove(u);
00499       return -1;
00500    }
00501 
00502    ast_speech_start(speech);
00503 
00504    ast_module_user_remove(u);
00505 
00506    return res;
00507 }
00508 
00509 /*! \brief SpeechProcessingSound(Sound File) Dialplan Application */
00510 static int speech_processing_sound(struct ast_channel *chan, void *data)
00511 {
00512         int res = 0;
00513         struct ast_module_user *u = NULL;
00514         struct ast_speech *speech = find_speech(chan);
00515 
00516         u = ast_module_user_add(chan);
00517 
00518         if (speech == NULL) {
00519                 ast_module_user_remove(u);
00520                 return -1;
00521         }
00522 
00523    if (speech->processing_sound != NULL) {
00524       free(speech->processing_sound);
00525       speech->processing_sound = NULL;
00526    }
00527 
00528    speech->processing_sound = strdup(data);
00529 
00530         ast_module_user_remove(u);
00531 
00532         return res;
00533 }
00534 
00535 /*! \brief Helper function used by speech_background to playback a soundfile */
00536 static int speech_streamfile(struct ast_channel *chan, const char *filename, const char *preflang)
00537 {
00538         struct ast_filestream *fs = NULL;
00539 
00540    if (!(fs = ast_openstream(chan, filename, preflang)))
00541       return -1;
00542    
00543    if (ast_applystream(chan, fs))
00544       return -1;
00545    
00546    ast_playstream(fs);
00547 
00548         return 0;
00549 }
00550 
00551 /*! \brief SpeechBackground(Sound File|Timeout) Dialplan Application */
00552 static int speech_background(struct ast_channel *chan, void *data)
00553 {
00554         unsigned int timeout = 0;
00555         int res = 0, done = 0, argc = 0, started = 0, quieted = 0, max_dtmf_len = 0;
00556         struct ast_module_user *u = NULL;
00557         struct ast_speech *speech = find_speech(chan);
00558         struct ast_frame *f = NULL;
00559         int oldreadformat = AST_FORMAT_SLINEAR;
00560         char dtmf[AST_MAX_EXTENSION] = "";
00561         time_t start, current;
00562         struct ast_datastore *datastore = NULL;
00563         char *argv[2], *args = NULL, *filename_tmp = NULL, *filename = NULL, tmp[2] = "", dtmf_terminator = '#';
00564    const char *tmp2 = NULL;
00565 
00566         args = ast_strdupa(data);
00567 
00568         u = ast_module_user_add(chan);
00569 
00570         if (speech == NULL) {
00571                 ast_module_user_remove(u);
00572                 return -1;
00573         }
00574 
00575    /* If channel is not already answered, then answer it */
00576    if (chan->_state != AST_STATE_UP && ast_answer(chan)) {
00577       ast_module_user_remove(u);
00578       return -1;
00579    }
00580 
00581         /* Record old read format */
00582         oldreadformat = chan->readformat;
00583 
00584         /* Change read format to be signed linear */
00585         if (ast_set_read_format(chan, AST_FORMAT_SLINEAR)) {
00586                 ast_module_user_remove(u);
00587                 return -1;
00588         }
00589 
00590         /* Parse out options */
00591         argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0]));
00592         if (argc > 0) {
00593                 /* Yay sound file */
00594                 filename_tmp = ast_strdupa(argv[0]);
00595       if (!ast_strlen_zero(argv[1])) {
00596          if ((timeout = atoi(argv[1])) == 0)
00597             timeout = -1;
00598       } else
00599          timeout = 0;
00600         }
00601 
00602    /* See if the maximum DTMF length variable is set... we use a variable in case they want to carry it through their entire dialplan */
00603    if ((tmp2 = pbx_builtin_getvar_helper(chan, "SPEECH_DTMF_MAXLEN")) && !ast_strlen_zero(tmp2))
00604       max_dtmf_len = atoi(tmp2);
00605 
00606    /* See if a terminator is specified */
00607    if ((tmp2 = pbx_builtin_getvar_helper(chan, "SPEECH_DTMF_TERMINATOR"))) {
00608       if (ast_strlen_zero(tmp2))
00609          dtmf_terminator = '\0';
00610       else
00611          dtmf_terminator = tmp2[0];
00612    }
00613 
00614         /* Before we go into waiting for stuff... make sure the structure is ready, if not - start it again */
00615         if (speech->state == AST_SPEECH_STATE_NOT_READY || speech->state == AST_SPEECH_STATE_DONE) {
00616       ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
00617                 ast_speech_start(speech);
00618         }
00619 
00620    /* Ensure no streams are currently running */
00621    ast_stopstream(chan);
00622 
00623         /* Okay it's streaming so go into a loop grabbing frames! */
00624         while (done == 0) {
00625       /* If the filename is null and stream is not running, start up a new sound file */
00626       if (!quieted && (chan->streamid == -1 && chan->timingfunc == NULL) && (filename = strsep(&filename_tmp, "&"))) {
00627          /* Discard old stream information */
00628          ast_stopstream(chan);
00629          /* Start new stream */
00630          speech_streamfile(chan, filename, chan->language);
00631       }
00632 
00633                 /* Run scheduled stuff */
00634                 ast_sched_runq(chan->sched);
00635 
00636                 /* Yay scheduling */
00637                 res = ast_sched_wait(chan->sched);
00638                 if (res < 0) {
00639                         res = 1000;
00640                 }
00641 
00642                 /* If there is a frame waiting, get it - if not - oh well */
00643                 if (ast_waitfor(chan, res) > 0) {
00644                         f = ast_read(chan);
00645                         if (f == NULL) {
00646                                 /* The channel has hung up most likely */
00647                                 done = 3;
00648                                 break;
00649                         }
00650                 }
00651 
00652       /* Do timeout check (shared between audio/dtmf) */
00653       if ((!quieted || strlen(dtmf)) && started == 1) {
00654          time(&current);
00655          if ((current-start) >= timeout) {
00656             done = 1;
00657             if (f)
00658                ast_frfree(f);
00659             break;
00660          }
00661       }
00662 
00663                 /* Do checks on speech structure to see if it's changed */
00664                 ast_mutex_lock(&speech->lock);
00665                 if (ast_test_flag(speech, AST_SPEECH_QUIET)) {
00666          if (chan->stream)
00667             ast_stopstream(chan);
00668          ast_clear_flag(speech, AST_SPEECH_QUIET);
00669          quieted = 1;
00670                 }
00671                 /* Check state so we can see what to do */
00672                 switch (speech->state) {
00673                 case AST_SPEECH_STATE_READY:
00674                         /* If audio playback has stopped do a check for timeout purposes */
00675                         if (chan->streamid == -1 && chan->timingfunc == NULL)
00676                                 ast_stopstream(chan);
00677                         if (!quieted && chan->stream == NULL && timeout && started == 0 && !filename_tmp) {
00678             if (timeout == -1) {
00679                done = 1;
00680                if (f)
00681                   ast_frfree(f);
00682                break;
00683             }
00684             time(&start);
00685             started = 1;
00686                         }
00687                         /* Write audio frame out to speech engine if no DTMF has been received */
00688                         if (!strlen(dtmf) && f != NULL && f->frametype == AST_FRAME_VOICE) {
00689                                 ast_speech_write(speech, f->data, f->datalen);
00690                         }
00691                         break;
00692                 case AST_SPEECH_STATE_WAIT:
00693                         /* Cue up waiting sound if not already playing */
00694          if (!strlen(dtmf)) {
00695             if (chan->stream == NULL) {
00696                if (speech->processing_sound != NULL) {
00697                   if (strlen(speech->processing_sound) > 0 && strcasecmp(speech->processing_sound,"none")) {
00698                      speech_streamfile(chan, speech->processing_sound, chan->language);
00699                   }
00700                }
00701             } else if (chan->streamid == -1 && chan->timingfunc == NULL) {
00702                ast_stopstream(chan);
00703                if (speech->processing_sound != NULL) {
00704                   if (strlen(speech->processing_sound) > 0 && strcasecmp(speech->processing_sound,"none")) {
00705                      speech_streamfile(chan, speech->processing_sound, chan->language);
00706                   }
00707                }
00708             }
00709          }
00710                         break;
00711                 case AST_SPEECH_STATE_DONE:
00712          /* Now that we are done... let's switch back to not ready state */
00713          ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
00714          if (!strlen(dtmf)) {
00715             /* Copy to speech structure the results, if available */
00716             speech->results = ast_speech_results_get(speech);
00717             /* Break out of our background too */
00718             done = 1;
00719             /* Stop audio playback */
00720             if (chan->stream != NULL) {
00721                ast_stopstream(chan);
00722             }
00723          }
00724                         break;
00725                 default:
00726                         break;
00727                 }
00728                 ast_mutex_unlock(&speech->lock);
00729 
00730                 /* Deal with other frame types */
00731                 if (f != NULL) {
00732                         /* Free the frame we received */
00733                         switch (f->frametype) {
00734                         case AST_FRAME_DTMF:
00735             if (dtmf_terminator != '\0' && f->subclass == dtmf_terminator) {
00736                done = 1;
00737             } else {
00738                quieted = 1;
00739                if (chan->stream != NULL) {
00740                   ast_stopstream(chan);
00741                }
00742                if (!started) {
00743                   /* Change timeout to be 5 seconds for DTMF input */
00744                   timeout = (chan->pbx && chan->pbx->dtimeout) ? chan->pbx->dtimeout : 5;
00745                   started = 1;
00746                }
00747                time(&start);
00748                snprintf(tmp, sizeof(tmp), "%c", f->subclass);
00749                strncat(dtmf, tmp, sizeof(dtmf) - strlen(dtmf) - 1);
00750                /* If the maximum length of the DTMF has been reached, stop now */
00751                if (max_dtmf_len && strlen(dtmf) == max_dtmf_len)
00752                   done = 1;
00753             }
00754                                 break;
00755                         case AST_FRAME_CONTROL:
00756                                 switch (f->subclass) {
00757                                 case AST_CONTROL_HANGUP:
00758                                         /* Since they hung up we should destroy the speech structure */
00759                                         done = 3;
00760                                 default:
00761                                         break;
00762                                 }
00763                         default:
00764                                 break;
00765                         }
00766                         ast_frfree(f);
00767                         f = NULL;
00768                 }
00769         }
00770 
00771    if (strlen(dtmf)) {
00772       /* We sort of make a results entry */
00773       speech->results = ast_calloc(1, sizeof(*speech->results));
00774       if (speech->results != NULL) {
00775          ast_speech_dtmf(speech, dtmf);
00776          speech->results->score = 1000;
00777          speech->results->text = strdup(dtmf);
00778          speech->results->grammar = strdup("dtmf");
00779       }
00780       ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
00781    }
00782 
00783         /* See if it was because they hung up */
00784         if (done == 3) {
00785                 /* Destroy speech structure */
00786                 ast_speech_destroy(speech);
00787                 datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
00788                 if (datastore != NULL) {
00789                         ast_channel_datastore_remove(chan, datastore);
00790                 }
00791         } else {
00792                 /* Channel is okay so restore read format */
00793                 ast_set_read_format(chan, oldreadformat);
00794         }
00795 
00796         ast_module_user_remove(u);
00797 
00798         return 0;
00799 }
00800 
00801 
00802 /*! \brief SpeechDestroy() Dialplan Application */
00803 static int speech_destroy(struct ast_channel *chan, void *data)
00804 {
00805    int res = 0;
00806         struct ast_module_user *u = NULL;
00807    struct ast_speech *speech = find_speech(chan);
00808    struct ast_datastore *datastore = NULL;
00809 
00810    u = ast_module_user_add(chan);
00811 
00812    if (speech == NULL) {
00813       ast_module_user_remove(u);
00814       return -1;
00815    }
00816 
00817    /* Destroy speech structure */
00818    ast_speech_destroy(speech);
00819 
00820    datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
00821    if (datastore != NULL) {
00822       ast_channel_datastore_remove(chan, datastore);
00823    }
00824 
00825    ast_module_user_remove(u);
00826 
00827    return res;
00828 }
00829 
00830 static int unload_module(void)
00831 {
00832    int res = 0;
00833 
00834    res = ast_unregister_application("SpeechCreate");
00835    res |= ast_unregister_application("SpeechLoadGrammar");
00836    res |= ast_unregister_application("SpeechUnloadGrammar");
00837    res |= ast_unregister_application("SpeechActivateGrammar");
00838         res |= ast_unregister_application("SpeechDeactivateGrammar");
00839    res |= ast_unregister_application("SpeechStart");
00840    res |= ast_unregister_application("SpeechBackground");
00841    res |= ast_unregister_application("SpeechDestroy");
00842    res |= ast_unregister_application("SpeechProcessingSound");
00843    res |= ast_custom_function_unregister(&speech_function);
00844    res |= ast_custom_function_unregister(&speech_score_function);
00845    res |= ast_custom_function_unregister(&speech_text_function);
00846    res |= ast_custom_function_unregister(&speech_grammar_function);
00847    res |= ast_custom_function_unregister(&speech_engine_function);
00848    res |= ast_custom_function_unregister(&speech_results_type_function);
00849 
00850    ast_module_user_hangup_all();
00851 
00852    return res; 
00853 }
00854 
00855 static int load_module(void)
00856 {
00857    int res = 0;
00858 
00859    res = ast_register_application("SpeechCreate", speech_create, "Create a Speech Structure", speechcreate_descrip);
00860    res |= ast_register_application("SpeechLoadGrammar", speech_load, "Load a Grammar", speechload_descrip);
00861    res |= ast_register_application("SpeechUnloadGrammar", speech_unload, "Unload a Grammar", speechunload_descrip);
00862    res |= ast_register_application("SpeechActivateGrammar", speech_activate, "Activate a Grammar", speechactivategrammar_descrip);
00863         res |= ast_register_application("SpeechDeactivateGrammar", speech_deactivate, "Deactivate a Grammar", speechdeactivategrammar_descrip);
00864    res |= ast_register_application("SpeechStart", speech_start, "Start recognizing voice in the audio stream", speechstart_descrip);
00865    res |= ast_register_application("SpeechBackground", speech_background, "Play a sound file and wait for speech to be recognized", speechbackground_descrip);
00866    res |= ast_register_application("SpeechDestroy", speech_destroy, "End speech recognition", speechdestroy_descrip);
00867    res |= ast_register_application("SpeechProcessingSound", speech_processing_sound, "Change background processing sound", speechprocessingsound_descrip);
00868    res |= ast_custom_function_register(&speech_function);
00869    res |= ast_custom_function_register(&speech_score_function);
00870    res |= ast_custom_function_register(&speech_text_function);
00871    res |= ast_custom_function_register(&speech_grammar_function);
00872    res |= ast_custom_function_register(&speech_engine_function);
00873    res |= ast_custom_function_register(&speech_results_type_function);
00874 
00875    return res;
00876 }
00877 
00878 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialplan Speech Applications");

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