[asterisk-commits] tilghman: trunk r106072 - in /trunk: ./ apps/ configs/ include/asterisk/ main...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Mar 5 10:23:45 CST 2008
Author: tilghman
Date: Wed Mar 5 10:23:44 2008
New Revision: 106072
URL: http://svn.digium.com/view/asterisk?view=rev&rev=106072
Log:
Create a centralized configuration option for silencethreshold
(closes issue #11236)
Reported by: philipps
Patches:
20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
Tested by: philipps
Added:
trunk/configs/dsp.conf.sample (with props)
Modified:
trunk/CHANGES
trunk/UPGRADE.txt
trunk/apps/app_amd.c
trunk/apps/app_dial.c
trunk/apps/app_followme.c
trunk/apps/app_meetme.c
trunk/apps/app_minivm.c
trunk/apps/app_record.c
trunk/apps/app_voicemail.c
trunk/apps/app_waitforsilence.c
trunk/include/asterisk/dsp.h
trunk/main/app.c
trunk/main/asterisk.c
trunk/main/dsp.c
trunk/main/loader.c
trunk/res/res_agi.c
Modified: trunk/CHANGES
URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Wed Mar 5 10:23:44 2008
@@ -175,7 +175,7 @@
* Proper codec support in chan_skinny.
* Added settings for IP and Ethernet QoS requests
-
+MGCP changes
------------
* Added separate settings for media QoS in mgcp.conf
@@ -387,6 +387,8 @@
* The ChannelRedirect application no longer exits the dialplan if the given channel
does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
or NOCHANNEL if the given channel was not found.
+ * The silencethreshold setting that was previously configurable in multiple
+ applications is now settable globally via dsp.conf.
Music On Hold Changes
---------------------
Modified: trunk/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/trunk/UPGRADE.txt?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Wed Mar 5 10:23:44 2008
@@ -55,6 +55,9 @@
* The concise versions of various CLI commands are now deprecated. We recommend
using the manager interface (AMI) for application integration with Asterisk.
+
+* The silencethreshold used for various applications is now settable via a
+ centralized config option in dsp.conf.
Voicemail:
Modified: trunk/apps/app_amd.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/apps/app_amd.c (original)
+++ trunk/apps/app_amd.c Wed Mar 5 10:23:44 2008
@@ -371,6 +371,8 @@
struct ast_variable *var = NULL;
struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
+ dfltSilenceThreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
+
if (!(cfg = ast_config_load("amd.conf", config_flags))) {
ast_log(LOG_ERROR, "Configuration file amd.conf missing.\n");
return -1;
Modified: trunk/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_dial.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Wed Mar 5 10:23:44 2008
@@ -60,6 +60,7 @@
#include "asterisk/privacy.h"
#include "asterisk/stringfields.h"
#include "asterisk/global_datastores.h"
+#include "asterisk/dsp.h"
static char *app = "Dial";
@@ -1115,6 +1116,7 @@
char callerid[60];
int res;
char *l;
+ int silencethreshold;
if (!ast_strlen_zero(chan->cid.cid_num)) {
l = ast_strdupa(chan->cid.cid_num);
@@ -1188,8 +1190,9 @@
"At the tone, please say your name:"
*/
+ silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
ast_answer(chan);
- res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "gsm", &duration, 128, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
+ res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "gsm", &duration, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
/* don't think we'll need a lock removed, we took care of
conflicts by naming the pa.privintro file */
if (res == -1) {
Modified: trunk/apps/app_followme.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_followme.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/apps/app_followme.c (original)
+++ trunk/apps/app_followme.c Wed Mar 5 10:23:44 2008
@@ -55,6 +55,7 @@
#include "asterisk/utils.h"
#include "asterisk/causes.h"
#include "asterisk/astdb.h"
+#include "asterisk/dsp.h"
#include "asterisk/app.h"
static char *app = "FollowMe";
@@ -130,7 +131,7 @@
int ynidx;
long digts;
int cleared;
- AST_LIST_ENTRY(findme_user) entry;
+ AST_LIST_ENTRY(findme_user) entry;
};
enum {
@@ -183,7 +184,6 @@
/* Free the whitelisted number */
ast_free(prev);
AST_LIST_HEAD_INIT_NOLOCK(&f->wlnumbers);
-
}
@@ -258,7 +258,6 @@
{
struct number *cur;
char *tmp;
-
if (!(cur = ast_calloc(1, sizeof(*cur))))
return NULL;
@@ -284,7 +283,7 @@
char numberstr[90];
int timeout;
char *timeoutstr;
- int numorder;
+ int numorder;
const char *takecallstr;
const char *declinecallstr;
const char *tmpstr;
@@ -307,7 +306,7 @@
}
featuredigittostr = ast_variable_retrieve(cfg, "general", "featuredigittimeout");
-
+
if (!ast_strlen_zero(featuredigittostr)) {
if (!sscanf(featuredigittostr, "%d", &featuredigittimeout))
featuredigittimeout = 5000;
@@ -316,7 +315,7 @@
takecallstr = ast_variable_retrieve(cfg, "general", "takecall");
if (!ast_strlen_zero(takecallstr))
ast_copy_string(takecall, takecallstr, sizeof(takecall));
-
+
declinecallstr = ast_variable_retrieve(cfg, "general", "declinecall");
if (!ast_strlen_zero(declinecallstr))
ast_copy_string(nextindp, declinecallstr, sizeof(nextindp));
@@ -369,7 +368,7 @@
/* Totally fail if we fail to find/create an entry */
if (!f)
continue;
-
+
if (!new)
ast_mutex_lock(&f->lock);
/* Re-initialize the profile */
@@ -399,8 +398,8 @@
timeout = 25;
numorder = 0;
}
-
- if (!numorder) {
+
+ if (!numorder) {
idx = 1;
AST_LIST_TRAVERSE(&f->numbers, nm, entry)
idx++;
@@ -414,7 +413,7 @@
}
var = var->next;
} /* End while(var) loop */
-
+
if (!new)
ast_mutex_unlock(&f->lock);
else
@@ -431,7 +430,7 @@
static void clear_caller(struct findme_user *tmpuser)
{
struct ast_channel *outbound;
-
+
if (tmpuser && tmpuser->ochan && tmpuser->state >= 0) {
outbound = tmpuser->ochan;
if (!outbound->cdr) {
@@ -460,12 +459,11 @@
static void clear_calling_tree(struct findme_user_listptr *findme_user_list)
{
struct findme_user *tmpuser;
-
+
AST_LIST_TRAVERSE(findme_user_list, tmpuser, entry) {
clear_caller(tmpuser);
tmpuser->cleared = 1;
}
-
}
@@ -489,29 +487,29 @@
/* ------------ wait_for_winner_channel start --------------- */
callfromname = ast_strdupa(tpargs->callfromprompt);
- pressbuttonname = ast_strdupa(tpargs->optionsprompt);
+ pressbuttonname = ast_strdupa(tpargs->optionsprompt);
if (AST_LIST_EMPTY(findme_user_list)) {
ast_verb(3, "couldn't reach at this number.\n");
return NULL;
}
-
+
if (!caller) {
ast_verb(3, "Original caller hungup. Cleanup.\n");
clear_calling_tree(findme_user_list);
return NULL;
}
-
+
totalwait = nm->timeout * 1000;
-
+
while (!ctstatus) {
to = 1000;
pos = 1;
livechannels = 0;
watchers[0] = caller;
-
- dg = 0;
- winner = NULL;
+
+ dg = 0;
+ winner = NULL;
AST_LIST_TRAVERSE(findme_user_list, tmpuser, entry) {
if (tmpuser->state >= 0 && tmpuser->ochan) {
if (tmpuser->state == 3)
@@ -526,7 +524,7 @@
} else {
ast_log(LOG_WARNING, "Unable to playback %s.\n", callfromname);
return NULL;
- }
+ }
} else {
tmpuser->state = 2;
tmpuser->digts = 0;
@@ -566,7 +564,6 @@
tmpuser->ynidx = 0;
if (!ast_streamfile(tmpuser->ochan, pressbuttonname, tmpuser->ochan->language)) {
tmpuser->state = 3;
-
} else {
return NULL;
}
@@ -580,7 +577,7 @@
livechannels++;
}
}
-
+
tmpto = to;
if (to < 0) {
to = 1000;
@@ -590,7 +587,7 @@
winner = ast_waitfor_n(watchers, pos, &to);
tmpto -= to;
totalwait -= tmpto;
- wtd = to;
+ wtd = to;
if (totalwait <= 0) {
ast_verb(3, "We've hit our timeout for this step. Drop everyone and move on to the next one. %ld\n", totalwait);
clear_calling_tree(findme_user_list);
@@ -631,8 +628,8 @@
ast_log(LOG_WARNING, "Unable to playback %s.\n", callfromname);
ast_frfree(f);
return NULL;
- }
- } else {
+ }
+ } else {
tmpuser->state = 2;
if (!ast_streamfile(tmpuser->ochan, tpargs->norecordingprompt, tmpuser->ochan->language))
ast_sched_runq(tmpuser->ochan->sched);
@@ -693,18 +690,18 @@
if (!strcmp(tmpuser->yn, tpargs->takecall)) {
ast_debug(1, "Match to take the call!\n");
ast_frfree(f);
- return tmpuser->ochan;
+ return tmpuser->ochan;
}
if (!strcmp(tmpuser->yn, tpargs->nextindp)) {
ast_debug(1, "Next in dial plan step requested.\n");
*status = 1;
ast_frfree(f);
return NULL;
- }
-
+ }
+
}
}
-
+
ast_frfree(f);
} else {
if (winner) {
@@ -723,12 +720,12 @@
}
}
}
- }
-
+ }
+
} else
ast_debug(1, "timed out waiting for action\n");
}
-
+
/* --- WAIT FOR WINNER NUMBER END! -----------*/
return NULL;
}
@@ -748,7 +745,7 @@
struct findme_user_listptr *findme_user_list;
int status;
- findme_user_list = ast_calloc(1, sizeof(*findme_user_list));
+ findme_user_list = ast_calloc(1, sizeof(*findme_user_list));
AST_LIST_HEAD_INIT_NOLOCK(findme_user_list);
/* We're going to figure out what the longest possible string of digits to collect is */
@@ -782,14 +779,14 @@
sprintf(dialarg, "%s", number);
else
sprintf(dialarg, "%s@%s", number, tpargs->context);
-
+
tmpuser = ast_calloc(1, sizeof(*tmpuser));
if (!tmpuser) {
ast_log(LOG_WARNING, "Out of memory!\n");
ast_free(findme_user_list);
return;
}
-
+
outbound = ast_request("Local", ast_best_codec(caller->nativeformats), dialarg, &dg);
if (outbound) {
ast_set_callerid(outbound, caller->cid.cid_num, caller->cid.cid_name, caller->cid.cid_num);
@@ -824,19 +821,17 @@
outbound = NULL;
}
}
-
}
} else
ast_log(LOG_WARNING, "Unable to allocate a channel for Local/%s cause: %s\n", dialarg, ast_cause2str(dg));
-
+
number = rest;
} while (number);
-
- status = 0;
+
+ status = 0;
if (!AST_LIST_EMPTY(findme_user_list))
winner = wait_for_winner(findme_user_list, nm, caller, tpargs->namerecloc, &status, tpargs);
-
-
+
while ((fmuser = AST_LIST_REMOVE_HEAD(findme_user_list, entry))) {
if (!fmuser->cleared && fmuser->ochan != winner)
clear_caller(fmuser);
@@ -845,21 +840,21 @@
fmuser = NULL;
tmpuser = NULL;
- headuser = NULL;
+ headuser = NULL;
if (winner)
break;
if (!caller) {
tpargs->status = 1;
ast_free(findme_user_list);
- return;
+ return;
}
idx++;
- AST_LIST_TRAVERSE(&tpargs->cnumbers, nm, entry)
+ AST_LIST_TRAVERSE(&tpargs->cnumbers, nm, entry) {
if (nm->order == idx)
break;
-
+ }
}
ast_free(findme_user_list);
if (!winner)
@@ -869,9 +864,7 @@
tpargs->outbound = winner;
}
-
return;
-
}
static int app_exec(struct ast_channel *chan, void *data)
@@ -887,7 +880,6 @@
struct ast_channel *caller;
struct ast_channel *outbound;
static char toast[80];
-
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(followmeid);
AST_APP_ARG(options);
@@ -897,7 +889,7 @@
ast_log(LOG_WARNING, "%s requires an argument (followmeid)\n", app);
return -1;
}
-
+
if (!(argstr = ast_strdupa((char *)data))) {
ast_log(LOG_ERROR, "Out of memory!\n");
return -1;
@@ -916,7 +908,7 @@
break;
}
AST_RWLIST_UNLOCK(&followmes);
-
+
ast_debug(1, "New profile %s.\n", args.followmeid);
if (!f) {
@@ -927,7 +919,7 @@
/* XXX TODO: Reinsert the db check value to see whether or not follow-me is on or off */
if (args.options)
ast_app_parse_options(followme_opts, &targs.followmeflags, NULL, args.options);
-
+
/* Lock the profile lock and copy out everything we need to run with before unlocking it again */
ast_mutex_lock(&f->lock);
targs.mohclass = ast_strdupa(f->moh);
@@ -948,38 +940,38 @@
AST_LIST_INSERT_TAIL(&targs.cnumbers, newnm, entry);
}
ast_mutex_unlock(&f->lock);
-
+
if (ast_test_flag(&targs.followmeflags, FOLLOWMEFLAG_STATUSMSG))
ast_stream_and_wait(chan, targs.statusprompt, "");
-
+
snprintf(namerecloc,sizeof(namerecloc),"%s/followme.%s",ast_config_AST_SPOOL_DIR,chan->uniqueid);
duration = 5;
-
+
if (ast_test_flag(&targs.followmeflags, FOLLOWMEFLAG_RECORDNAME))
- if (ast_play_and_record(chan, "vm-rec-name", namerecloc, 5, "sln", &duration, 128, 0, NULL) < 0)
+ if (ast_play_and_record(chan, "vm-rec-name", namerecloc, 5, "sln", &duration, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE), 0, NULL) < 0)
goto outrun;
-
+
if (!ast_fileexists(namerecloc, NULL, chan->language))
- ast_copy_string(namerecloc, "", sizeof(namerecloc));
-
+ ast_copy_string(namerecloc, "", sizeof(namerecloc));
+
if (ast_streamfile(chan, targs.plsholdprompt, chan->language))
goto outrun;
if (ast_waitstream(chan, "") < 0)
goto outrun;
ast_moh_start(chan, S_OR(targs.mohclass, NULL), NULL);
-
+
targs.status = 0;
targs.chan = chan;
ast_copy_string(targs.namerecloc, namerecloc, sizeof(targs.namerecloc));
-
- findmeexec(&targs);
-
+
+ findmeexec(&targs);
+
while ((nm = AST_LIST_REMOVE_HEAD(&targs.cnumbers, entry)))
ast_free(nm);
-
+
if (!ast_strlen_zero(namerecloc))
- unlink(namerecloc);
-
+ unlink(namerecloc);
+
if (targs.status != 100) {
ast_moh_stop(chan);
if (ast_test_flag(&targs.followmeflags, FOLLOWMEFLAG_UNREACHABLEMSG))
@@ -989,12 +981,12 @@
caller = chan;
outbound = targs.outbound;
/* Bridge the two channels. */
-
- memset(&config,0,sizeof(struct ast_bridge_config));
+
+ memset(&config, 0, sizeof(config));
ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
-
+
ast_moh_stop(caller);
/* Be sure no generators are left on it */
ast_deactivate_generator(caller);
@@ -1006,7 +998,7 @@
goto outrun;
}
time(&answer_time);
- res = ast_bridge_call(caller,outbound,&config);
+ res = ast_bridge_call(caller, outbound, &config);
time(&end_time);
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
pbx_builtin_setvar_helper(caller, "DIALEDTIME", toast);
@@ -1017,7 +1009,7 @@
}
outrun:
-
+
return res;
}
@@ -1051,7 +1043,7 @@
{
reload_followme(1);
- return 0;
+ return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Find-Me/Follow-Me Application",
Modified: trunk/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_meetme.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/apps/app_meetme.c (original)
+++ trunk/apps/app_meetme.c Wed Mar 5 10:23:44 2008
@@ -1699,7 +1699,7 @@
"%s/meetme/meetme-username-%s-%d", ast_config_AST_SPOOL_DIR,
conf->confno, user->user_no);
if (confflags & CONFFLAG_INTROUSERNOREVIEW)
- res = ast_play_and_record(chan, "vm-rec-name", user->namerecloc, 10, "sln", &duration, 128, 0, NULL);
+ res = ast_play_and_record(chan, "vm-rec-name", user->namerecloc, 10, "sln", &duration, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE), 0, NULL);
else
res = ast_record_review(chan, "vm-rec-name", user->namerecloc, 10, "sln", &duration, NULL);
if (res == -1)
Modified: trunk/apps/app_minivm.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_minivm.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/apps/app_minivm.c (original)
+++ trunk/apps/app_minivm.c Wed Mar 5 10:23:44 2008
@@ -2360,7 +2360,6 @@
/* First, set some default settings */
global_externnotify[0] = '\0';
global_logfile[0] = '\0';
- global_silencethreshold = 256;
global_vmmaxmessage = 2000;
global_maxgreet = 2000;
global_vmminmessage = 0;
@@ -2374,6 +2373,8 @@
/* Reset statistics */
memset(&global_stats, 0, sizeof(global_stats));
global_stats.reset = ast_tvnow();
+
+ global_silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
/* Make sure we could load configuration file */
if (!cfg) {
@@ -2640,7 +2641,7 @@
ast_cli(a->fd, "\n");
ast_cli(a->fd, " Mail command (shell): %s\n", global_mailcmd);
ast_cli(a->fd, " Max silence: %d\n", global_maxsilence);
- ast_cli(a->fd, " Silence treshold: %d\n", global_silencethreshold);
+ ast_cli(a->fd, " Silence threshold: %d\n", global_silencethreshold);
ast_cli(a->fd, " Max message length (secs): %d\n", global_vmmaxmessage);
ast_cli(a->fd, " Min message length (secs): %d\n", global_vmminmessage);
ast_cli(a->fd, " Default format: %s\n", default_vmformat);
Modified: trunk/apps/app_record.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_record.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/apps/app_record.c (original)
+++ trunk/apps/app_record.c Wed Mar 5 10:23:44 2008
@@ -243,7 +243,7 @@
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
- ast_dsp_set_threshold(sildet, 256);
+ ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
}
/* Create the directory if it does not exist. */
Modified: trunk/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_voicemail.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/apps/app_voicemail.c (original)
+++ trunk/apps/app_voicemail.c Wed Mar 5 10:23:44 2008
@@ -8399,7 +8399,7 @@
}
/* Silence treshold */
- silencethreshold = 256;
+ silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
if ((val = ast_variable_retrieve(cfg, "general", "silencethreshold")))
silencethreshold = atoi(val);
Modified: trunk/apps/app_waitforsilence.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_waitforsilence.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/apps/app_waitforsilence.c (original)
+++ trunk/apps/app_waitforsilence.c Wed Mar 5 10:23:44 2008
@@ -29,6 +29,12 @@
*
* \author David C. Troy <dave at popvox.com>
*
+ * \brief Wait For Noise
+ * The same as Wait For Silence but listenes noise on the chennel that is above \n
+ * the pre-configured silence threshold from dsp.conf
+ *
+ * \author Philipp Skadorov <skadorov at yahoo.com>
+ *
* \ingroup applications
*/
@@ -42,9 +48,9 @@
#include "asterisk/dsp.h"
#include "asterisk/module.h"
-static char *app = "WaitForSilence";
-static char *synopsis = "Waits for a specified amount of silence";
-static char *descrip =
+static char *app_silence = "WaitForSilence";
+static char *synopsis_silence = "Waits for a specified amount of silence";
+static char *descrip_silence =
" WaitForSilence(silencerequired[,iterations][,timeout]):\n"
"Wait for Silence: Waits for up to 'silencerequired' \n"
"milliseconds of silence, 'iterations' times or once if omitted.\n"
@@ -68,14 +74,23 @@
"SILENCE - if exited with silence detected\n"
"TIMEOUT - if exited without silence detected after timeout\n";
-static int do_waiting(struct ast_channel *chan, int silencereqd, time_t waitstart, int timeout) {
+static char *app_noise = "WaitForNoise";
+static char *synopsis_noise = "Waits for a specified amount of noise";
+static char *descrip_noise =
+"WaitForNoise(noiserequired[,iterations][,timeout]) \n"
+"Wait for Noise: The same as Wait for Silance but waits for noise that is above the threshold specified\n";
+
+static int do_waiting(struct ast_channel *chan, int timereqd, time_t waitstart, int timeout, int wait_for_silence) {
struct ast_frame *f;
- int dspsilence = 0;
- static int silencethreshold = 128;
+ int dsptime = 0;
int rfmt = 0;
int res = 0;
struct ast_dsp *sildet; /* silence detector dsp */
time_t now;
+
+ /*Either silence or noise calc depending on wait_for_silence flag*/
+ int (*ast_dsp_func)(struct ast_dsp*, struct ast_frame*, int*) =
+ wait_for_silence ? ast_dsp_silence : ast_dsp_noise;
rfmt = chan->readformat; /* Set to linear mode */
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
@@ -89,15 +104,15 @@
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
- ast_dsp_set_threshold(sildet, silencethreshold);
+ ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
/* Await silence... */
f = NULL;
for(;;) {
/* Start with no silence received */
- dspsilence = 0;
-
- res = ast_waitfor(chan, silencereqd);
+ dsptime = 0;
+
+ res = ast_waitfor(chan, timereqd);
/* Must have gotten a hangup; let's exit */
if (res <= 0) {
@@ -107,30 +122,36 @@
/* We waited and got no frame; sounds like digital silence or a muted digital channel */
if (!res) {
- dspsilence = silencereqd;
+ dsptime = timereqd;
} else {
/* Looks like we did get a frame, so let's check it out */
f = ast_read(chan);
if (!f)
break;
if (f && f->frametype == AST_FRAME_VOICE) {
- ast_dsp_silence(sildet, f, &dspsilence);
+ ast_dsp_func(sildet, f, &dsptime);
ast_frfree(f);
}
}
- ast_verb(3, "Got %dms silence< %dms required\n", dspsilence, silencereqd);
-
- if (dspsilence >= silencereqd) {
- ast_verb(3, "Exiting with %dms silence >= %dms required\n", dspsilence, silencereqd);
+ if (wait_for_silence)
+ ast_verb(6, "Got %dms silence < %dms required\n", dsptime, timereqd);
+ else
+ ast_verb(6, "Got %dms noise < %dms required\n", dsptime, timereqd);
+
+ if (dsptime >= timereqd) {
+ if (wait_for_silence)
+ ast_verb(3, "Exiting with %dms silence >= %dms required\n", dsptime, timereqd);
+ else
+ ast_verb(3, "Exiting with %dms noise >= %dms required\n", dsptime, timereqd);
/* Ended happily with silence */
res = 1;
- pbx_builtin_setvar_helper(chan, "WAITSTATUS", "SILENCE");
- ast_debug(1, "WAITSTATUS was set to SILENCE\n");
+ pbx_builtin_setvar_helper(chan, "WAITSTATUS", wait_for_silence ? "SILENCE" : "NOISE");
+ ast_debug(1, "WAITSTATUS was set to %s\n", wait_for_silence ? "SILENCE" : "NOISE");
break;
}
- if ( timeout && (difftime(time(&now),waitstart) >= timeout) ) {
+ if (timeout && (difftime(time(&now), waitstart) >= timeout)) {
pbx_builtin_setvar_helper(chan, "WAITSTATUS", "TIMEOUT");
ast_debug(1, "WAITSTATUS was set to TIMEOUT\n");
res = 0;
@@ -146,43 +167,60 @@
return res;
}
-static int waitforsilence_exec(struct ast_channel *chan, void *data)
+static int waitfor_exec(struct ast_channel *chan, void *data, int wait_for_silence)
{
int res = 1;
- int silencereqd = 1000;
+ int timereqd = 1000;
int timeout = 0;
int iterations = 1, i;
time_t waitstart;
res = ast_answer(chan); /* Answer the channel */
- if (!data || ( (sscanf(data, "%d,%d,%d", &silencereqd, &iterations, &timeout) != 3) &&
- (sscanf(data, "%d|%d", &silencereqd, &iterations) != 2) &&
- (sscanf(data, "%d", &silencereqd) != 1) ) ) {
+ if (!data || ( (sscanf(data, "%d,%d,%d", &timereqd, &iterations, &timeout) != 3) &&
+ (sscanf(data, "%d,%d", &timereqd, &iterations) != 2) &&
+ (sscanf(data, "%d", &timereqd) != 1) ) ) {
ast_log(LOG_WARNING, "Using default value of 1000ms, 1 iteration, no timeout\n");
}
- ast_verb(3, "Waiting %d time(s) for %d ms silence with %d timeout\n", iterations, silencereqd, timeout);
+ ast_verb(3, "Waiting %d time(s) for %d ms silence with %d timeout\n", iterations, timereqd, timeout);
time(&waitstart);
res = 1;
for (i=0; (i<iterations) && (res == 1); i++) {
- res = do_waiting(chan, silencereqd, waitstart, timeout);
+ res = do_waiting(chan, timereqd, waitstart, timeout, wait_for_silence);
}
if (res > 0)
res = 0;
return res;
}
+static int waitforsilence_exec(struct ast_channel *chan, void *data)
+{
+ return waitfor_exec(chan, data, 1);
+}
+
+static int waitfornoise_exec(struct ast_channel *chan, void *data)
+{
+ return waitfor_exec(chan, data, 0);
+}
static int unload_module(void)
{
- return ast_unregister_application(app);
+ int res;
+ res = ast_unregister_application(app_silence);
+ res |= ast_unregister_application(app_noise);
+
+ return res;
}
static int load_module(void)
{
- return ast_register_application(app, waitforsilence_exec, synopsis, descrip);
+ int res;
+
+ res = ast_register_application(app_silence, waitforsilence_exec, synopsis_silence, descrip_silence);
+ res |= ast_register_application(app_noise, waitfornoise_exec, synopsis_noise, descrip_noise);
+ return res;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Wait For Silence");
Added: trunk/configs/dsp.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/dsp.conf.sample?view=auto&rev=106072
==============================================================================
--- trunk/configs/dsp.conf.sample (added)
+++ trunk/configs/dsp.conf.sample Wed Mar 5 10:23:44 2008
@@ -1,0 +1,7 @@
+[default]
+;
+; Length of sound (in milliseconds) before a period of silence is considered
+; to be a change from talking to silence or a period of noise converts silence
+; to talking. [default=256]
+;
+;silencethreshold=256
Propchange: trunk/configs/dsp.conf.sample
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/configs/dsp.conf.sample
------------------------------------------------------------------------------
svn:keywords = Id Date Revision Author
Propchange: trunk/configs/dsp.conf.sample
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: trunk/include/asterisk/dsp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/dsp.h?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/include/asterisk/dsp.h (original)
+++ trunk/include/asterisk/dsp.h Wed Mar 5 10:23:44 2008
@@ -58,6 +58,13 @@
struct ast_dsp;
+enum threshold {
+ /* Array offsets */
+ THRESHOLD_SILENCE = 0,
+ /* Always the last */
+ THRESHOLD_MAX = 1,
+};
+
struct ast_dsp *ast_dsp_new(void);
void ast_dsp_free(struct ast_dsp *dsp);
@@ -83,6 +90,10 @@
/*! \brief Return non-zero if this is silence. Updates "totalsilence" with the total
number of seconds of silence */
int ast_dsp_silence(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence);
+
+/*! \brief Return non-zero if this is noise. Updates "totalnoise" with the total
+ number of seconds of noise */
+int ast_dsp_noise(struct ast_dsp *dsp, struct ast_frame *f, int *totalnoise);
/*! \brief Return non-zero if historically this should be a busy, request that
ast_dsp_silence has already been called */
@@ -115,4 +126,12 @@
/*! \brief Get tcount (Threshold counter) */
int ast_dsp_get_tcount(struct ast_dsp *dsp);
+/*! \brief Get silence threshold from dsp.conf*/
+int ast_dsp_get_threshold_from_settings(enum threshold which);
+
+/* \brief Reloads dsp settings from dsp.conf*/
+int ast_dsp_reload(void);
+
+int ast_dsp_init(void);
+
#endif /* _ASTERISK_DSP_H */
Modified: trunk/main/app.c
URL: http://svn.digium.com/view/asterisk/trunk/main/app.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/main/app.c (original)
+++ trunk/main/app.c Wed Mar 5 10:23:44 2008
@@ -1268,7 +1268,7 @@
int ast_record_review(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, const char *path)
{
- int silencethreshold = 128;
+ int silencethreshold;
int maxsilence = 0;
int res = 0;
int cmd = 0;
@@ -1285,6 +1285,8 @@
}
cmd = '3'; /* Want to start by recording */
+
+ silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
while ((cmd >= 0) && (cmd != 't')) {
switch (cmd) {
Modified: trunk/main/asterisk.c
URL: http://svn.digium.com/view/asterisk/trunk/main/asterisk.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/main/asterisk.c (original)
+++ trunk/main/asterisk.c Wed Mar 5 10:23:44 2008
@@ -121,6 +121,7 @@
#include "asterisk/linkedlists.h"
#include "asterisk/devicestate.h"
#include "asterisk/module.h"
+#include "asterisk/dsp.h"
#include "asterisk/doxyref.h" /* Doxygen documentation */
@@ -3218,7 +3219,7 @@
}
ast_rtp_init();
-
+ ast_dsp_init();
ast_udptl_init();
if (ast_image_init()) {
Modified: trunk/main/dsp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/dsp.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/main/dsp.c (original)
+++ trunk/main/dsp.c Wed Mar 5 10:23:44 2008
@@ -53,6 +53,7 @@
#include "asterisk/alaw.h"
#include "asterisk/utils.h"
#include "asterisk/options.h"
+#include "asterisk/config.h"
/*! Number of goertzels for progress detect */
enum gsamp_size {
@@ -193,6 +194,8 @@
#define SAMPLES_IN_FRAME 160
+#define CONFIG_FILE_NAME "dsp.conf"
+
typedef struct {
int v2;
int v3;
@@ -271,6 +274,8 @@
static char dtmf_positions[] = "123A" "456B" "789C" "*0#D";
static char bell_mf_positions[] = "1247C-358A--69*---0B----#";
+
+static int thresholds[THRESHOLD_MAX];
static inline void goertzel_sample(goertzel_state_t *s, short sample)
{
@@ -1021,7 +1026,7 @@
return __ast_dsp_call_progress(dsp, inf->data, inf->datalen / 2);
}
-static int __ast_dsp_silence(struct ast_dsp *dsp, short *s, int len, int *totalsilence)
+static int __ast_dsp_silence_noise(struct ast_dsp *dsp, short *s, int len, int *totalsilence, int *totalnoise)
{
int accum;
int x;
@@ -1073,6 +1078,8 @@
}
if (totalsilence)
*totalsilence = dsp->totalsilence;
+ if (totalnoise)
+ *totalnoise = dsp->totalnoise;
return res;
}
@@ -1179,8 +1186,27 @@
}
s = f->data;
len = f->datalen/2;
- return __ast_dsp_silence(dsp, s, len, totalsilence);
-}
+ return __ast_dsp_silence_noise(dsp, s, len, totalsilence, NULL);
+}
+
+int ast_dsp_noise(struct ast_dsp *dsp, struct ast_frame *f, int *totalnoise)
+{
+ short *s;
+ int len;
+
+ if (f->frametype != AST_FRAME_VOICE) {
+ ast_log(LOG_WARNING, "Can't calculate noise on a non-voice frame\n");
+ return 0;
+ }
+ if (f->subclass != AST_FORMAT_SLINEAR) {
+ ast_log(LOG_WARNING, "Can only calculate noise on signed-linear frames :(\n");
+ return 0;
+ }
+ s = f->data;
+ len = f->datalen/2;
+ return __ast_dsp_silence_noise(dsp, s, len, NULL, totalnoise);
+}
+
struct ast_frame *ast_dsp_process(struct ast_channel *chan, struct ast_dsp *dsp, struct ast_frame *af)
{
@@ -1236,7 +1262,7 @@
ast_log(LOG_WARNING, "Inband DTMF is not supported on codec %s. Use RFC2833\n", ast_getformatname(af->subclass));
return af;
}
- silence = __ast_dsp_silence(dsp, shortdata, len, NULL);
+ res = __ast_dsp_silence_noise(dsp, shortdata, len, &silence, NULL);
if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) && silence) {
memset(&dsp->f, 0, sizeof(dsp->f));
dsp->f.frametype = AST_FRAME_NULL;
@@ -1516,3 +1542,42 @@
{
return dsp->tcount;
}
+
+static int _dsp_init(int reload)
+{
+ struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
+ struct ast_config *cfg;
+ struct ast_variable *var;
+
+ cfg = ast_config_load(CONFIG_FILE_NAME, config_flags);
+
+ if (cfg && cfg != CONFIG_STATUS_FILEUNCHANGED) {
+ const char *value;
+
+ value = ast_variable_retrieve(cfg, "default", "silencethreshold");
+ if (value && sscanf(value, "%d", &thresholds[THRESHOLD_SILENCE]) != 1) {
+ ast_log(LOG_WARNING, "%s: '%s' is not a valid silencethreshold value\n", CONFIG_FILE_NAME, var->value);
+ thresholds[THRESHOLD_SILENCE] = 256;
+ } else if (!value)
+ thresholds[THRESHOLD_SILENCE] = 256;
+
+ ast_config_destroy(cfg);
+ }
+ return 0;
+}
+
+int ast_dsp_get_threshold_from_settings(enum threshold which)
+{
+ return thresholds[which];
+}
+
+int ast_dsp_init(void)
+{
+ return _dsp_init(0);
+}
+
+int ast_dsp_reload(void)
+{
+ return _dsp_init(1);
+}
+
Modified: trunk/main/loader.c
URL: http://svn.digium.com/view/asterisk/trunk/main/loader.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/main/loader.c (original)
+++ trunk/main/loader.c Wed Mar 5 10:23:44 2008
@@ -47,6 +47,7 @@
#include "asterisk/http.h"
#include "asterisk/lock.h"
#include "asterisk/features.h"
+#include "asterisk/dsp.h"
#ifdef DLFCNCOMPAT
#include "asterisk/dlfcn-compat.h"
@@ -249,6 +250,7 @@
{ "http", ast_http_reload },
{ "logger", logger_reload },
{ "features", ast_features_reload },
+ { "dsp", ast_dsp_reload},
{ NULL, NULL }
};
Modified: trunk/res/res_agi.c
URL: http://svn.digium.com/view/asterisk/trunk/res/res_agi.c?view=diff&rev=106072&r1=106071&r2=106072
==============================================================================
--- trunk/res/res_agi.c (original)
+++ trunk/res/res_agi.c Wed Mar 5 10:23:44 2008
@@ -1296,7 +1296,7 @@
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
- ast_dsp_set_threshold(sildet, 256);
+ ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
}
/* backward compatibility, if no offset given, arg[6] would have been
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