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Summary:ASTERISK-10755: [patch] WaitForNoise added to accomplish WaitForSilence
Reporter:Philipp Skadorov (philipps)Labels:
Date Opened:2007-11-13 12:10:26.000-0600Date Closed:2008-03-05 18:13:59.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/NewFeature
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 20080218__bug11236.diff.txt
( 1) silencethreshold_waitforsilence_tabified.patch
( 2) waitfornoise_120108.patch
Description:Hi,
Here is the patch to add new app WaitForNoise to accompany WaitForSilence app.
WaitForNoise has been implemented in the same module as WaitForSilence (app_waitforsilence) to use common code and avoid copy-paste.
The patch itself incorporates the patch from issue # 0011235 so you can use just this one if both bugs are applicable :)

****** ADDITIONAL INFORMATION ******

The patch:

Index: apps/app_meetme.c
===================================================================
--- apps/app_meetme.c (revision 89245)
+++ apps/app_meetme.c (working copy)
@@ -1516,6 +1516,9 @@
  long time_left_ms = 0;
  struct timeval nexteventts = { 0, };
  int to;
+    int silencethreshold;
+
+    silencethreshold = ast_dsp_get_threshold_from_settings();
 
if (!(user = ast_calloc(1, sizeof(*user))))
return ret;
@@ -1709,7 +1712,7 @@
"%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, silencethreshold, 0, NULL);
else
res = ast_record_review(chan, "vm-rec-name", user->namerecloc, 10, "sln", &duration, NULL);
if (res == -1)
Index: apps/app_minivm.c
===================================================================
--- apps/app_minivm.c (revision 89245)
+++ apps/app_minivm.c (working copy)
@@ -2291,6 +2291,8 @@
{
int error = 0;

+    global_silencethreshold = ast_dsp_get_threshold_from_settings();
+
while (var) {
/* Mail command */
if (!strcmp(var->name, "mailcmd")) {
Index: apps/app_voicemail.c
===================================================================
--- apps/app_voicemail.c (revision 89245)
+++ apps/app_voicemail.c (working copy)
@@ -8156,7 +8156,7 @@
}

/* Silence treshold */
- silencethreshold = 256;
+ silencethreshold = ast_dsp_get_threshold_from_settings();
if ((val = ast_variable_retrieve(cfg, "general", "silencethreshold")))
silencethreshold = atoi(val);

Index: apps/app_dial.c
===================================================================
--- apps/app_dial.c (revision 89245)
+++ apps/app_dial.c (working copy)
@@ -1093,6 +1093,7 @@
char callerid[60];
int res;
char *l;
+    int silencethreshold;

if (!ast_strlen_zero(chan->cid.cid_num)) {
l = ast_strdupa(chan->cid.cid_num);
@@ -1169,8 +1170,9 @@
  "At the tone, please say your name:"

*/
+            silencethreshold = ast_dsp_get_threshold_from_settings();
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) {
Index: apps/app_waitforsilence.c
===================================================================
--- apps/app_waitforsilence.c (revision 89245)
+++ apps/app_waitforsilence.c (working copy)
@@ -29,6 +29,11 @@
 *
 * \author David C. Troy <dave@popvox.com>
 *
+ * \brief Wait For Noise
+ * The same as Wait For Silence but listenes noise on the chennel that is above the threshold specified\n
+ *
+ * \author Philipp Skadorov <skadorov@yahoo.com>
+ *
 * \ingroup applications
 */

@@ -49,10 +54,10 @@
#include "asterisk/module.h"
#include "asterisk/options.h"

-static char *app = "WaitForSilence";
-static char *synopsis = "Waits for a specified amount of silence";
-static char *descrip =
-"  WaitForSilence(silencerequired[,iterations][,timeout]):\n"
+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"
"An optional timeout specified the number of seconds to return\n"
@@ -67,23 +72,35 @@
"include two or more calls to WaitForSilence when dealing with an answering\n"
"machine; first waiting for the spiel to finish, then waiting for the beep, etc.\n\n"
  "Examples:\n"
-"  - WaitForSilence(500,2) will wait for 1/2 second of silence, twice\n"
+"  - WaitForSilence(500|2) will wait for 1/2 second of silence, twice\n"
"  - WaitForSilence(1000) will wait for 1 second of silence, once\n"
-"  - WaitForSilence(300,3,10) will wait for 300ms silence, 3 times,\n"
+"  - WaitForSilence(300|3|10) will wait for 300ms silence, 3 times,\n"
"     and returns after 10 sec, even if silence is not detected\n\n"
"Sets the channel variable WAITSTATUS with to one of these values:\n"
"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\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;
+    const char * format_string = "";

+    int silencethreshold = ast_dsp_get_threshold_from_settings();
+
+    /*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);
if (res < 0) {
@@ -102,9 +119,9 @@
f = NULL;
for(;;) {
/* Start with no silence received */
- dspsilence = 0;
+ dsptime = 0;

- res = ast_waitfor(chan, silencereqd);
+ res = ast_waitfor(chan, timereqd);

/* Must have gotten a hangup; let's exit */
if (res <= 0) {
@@ -114,32 +131,46 @@

/* 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 (option_verbose > 6) {
+            format_string = wait_for_silence?
+                VERBOSE_PREFIX_3 "Got %dms silence silence < %dms required\n":
+                VERBOSE_PREFIX_3 "Got %dms silence noise < %dms required\n";

- if (dspsilence >= silencereqd) {
- ast_verb(3, "Exiting with %dms silence >= %dms required\n", dspsilence, silencereqd);
+ ast_verbose(format_string, dsptime, timereqd);
+        }
+ if (dsptime >= timereqd) {
+ if (option_verbose > 2)
+                format_string = wait_for_silence?
+                    VERBOSE_PREFIX_3 "Exiting with %dms silence >= %dms required\n" :
+                    VERBOSE_PREFIX_3 "Exiting with %dms silence >= %dms required\n";
+ ast_verbose(format_string, 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");
+            if(wait_for_silence) {
+    ast_log(LOG_DEBUG, "WAITSTATUS was set to SILENCE\n");
+            }
+            else {
+    ast_log(LOG_DEBUG, "WAITSTATUS was set to NOISE\n");
+            }
break;
}

if ( timeout && (difftime(time(&now),waitstart) >= timeout) ) {
pbx_builtin_setvar_helper(chan, "WAITSTATUS", "TIMEOUT");
- ast_debug(1, "WAITSTATUS was set to TIMEOUT\n");
+ ast_log(LOG_DEBUG, "WAITSTATUS was set to TIMEOUT\n");
res = 0;
break;
}
@@ -153,43 +184,63 @@
return res;
}

-static int waitforsilence_exec(struct ast_channel *chan, void *data)
+/*Waits either for silence or for noise depending on wait_for_silence flag*/
+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);
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_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);
+
+ ast_module_user_hangup_all();
+
+ return res;
}

static int load_module(void)
{
- return ast_register_application(app, waitforsilence_exec, synopsis, descrip);
+ int 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");
Index: apps/app_amd.c
===================================================================
--- apps/app_amd.c (revision 89245)
+++ apps/app_amd.c (working copy)
@@ -49,7 +49,7 @@
static char *descrip =
"  AMD([initialSilence],[greeting],[afterGreetingSilence],[totalAnalysisTime]\n"
"      ,[minimumWordLength],[betweenWordsSilence],[maximumNumberOfWords]\n"
-"      ,[silenceThreshold])\n"
+"      ,[silencethreshold])\n"
"  This application attempts to detect answering machines at the beginning\n"
"  of outbound calls.  Simply call this application after the call\n"
"  has been answered (outbound only, of course).\n"
@@ -68,7 +68,7 @@
"   consider the audio that follows as a new word.\n"
"- 'maximumNumberOfWords'is the maximum number of words in the greeting. \n"
"   If exceeded then MACHINE.\n"
-"- 'silenceThreshold' is the silence threshold.\n"
+"- 'silencethreshold' is the silence threshold.\n"
"This application sets the following channel variables upon completion:\n"
"    AMDSTATUS - This is the status of the answering machine detection.\n"
"                Possible values are:\n"
@@ -93,7 +93,6 @@
static int dfltBetweenWordsSilence  = 50;
static int dfltMaximumNumberOfWords = 3;
static int dfltSilenceThreshold     = 256;
-
static void isAnsweringMachine(struct ast_channel *chan, void *data)
{
int res = 0;
@@ -123,7 +122,7 @@
int minimumWordLength    = dfltMinimumWordLength;
int betweenWordsSilence  = dfltBetweenWordsSilence;
int maximumNumberOfWords = dfltMaximumNumberOfWords;
- int silenceThreshold     = dfltSilenceThreshold;
+ int silencethreshold;

AST_DECLARE_APP_ARGS(args,
    AST_APP_ARG(argInitialSilence);
@@ -138,6 +137,7 @@

ast_verb(3, "AMD: %s %s %s (Fmt: %d)\n", chan->name ,chan->cid.cid_ani, chan->cid.cid_rdnis, chan->readformat);

+    silencethreshold = ast_dsp_get_threshold_from_settings();
/* Lets parse the arguments. */
if (!ast_strlen_zero(parse)) {
/* Some arguments have been passed. Lets parse them and overwrite the defaults. */
@@ -157,16 +157,16 @@
if (!ast_strlen_zero(args.argMaximumNumberOfWords))
maximumNumberOfWords = atoi(args.argMaximumNumberOfWords);
if (!ast_strlen_zero(args.argSilenceThreshold))
- silenceThreshold = atoi(args.argSilenceThreshold);
+ silencethreshold = atoi(args.argSilenceThreshold);
} else {
ast_debug(1, "AMD using the default parameters.\n");
}

/* Now we're ready to roll! */
ast_verb(3, "AMD: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
- "totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] \n",
+ "totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silencethreshold [%d] \n",
initialSilence, greeting, afterGreetingSilence, totalAnalysisTime,
- minimumWordLength, betweenWordsSilence, maximumNumberOfWords, silenceThreshold );
+ minimumWordLength, betweenWordsSilence, maximumNumberOfWords, silencethreshold );

/* Set read format to signed linear so we get signed linear frames in */
readFormat = chan->readformat;
@@ -186,7 +186,7 @@
}

/* Set silence threshold to specified value */
- ast_dsp_set_threshold(silenceDetector, silenceThreshold);
+ ast_dsp_set_threshold(silenceDetector, silencethreshold);

/* Now we go into a loop waiting for frames from the channel */
while ((res = ast_waitfor(chan, totalAnalysisTime)) > -1) {
@@ -365,7 +365,7 @@
ast_config_destroy(cfg);

ast_verb(3, "AMD defaults: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
- "totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] \n",
+ "totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silencethreshold [%d] \n",
dfltInitialSilence, dfltGreeting, dfltAfterGreetingSilence, dfltTotalAnalysisTime,
dfltMinimumWordLength, dfltBetweenWordsSilence, dfltMaximumNumberOfWords, dfltSilenceThreshold );

Index: apps/app_record.c
===================================================================
--- apps/app_record.c (revision 89245)
+++ apps/app_record.c (working copy)
@@ -116,6 +116,7 @@
int rfmt = 0;
int ioflags;
int waitres;
+    int silencethreshold;
struct ast_silence_generator *silgen = NULL;
struct ast_flags flags = { 0, };
AST_DECLARE_APP_ARGS(args,
@@ -252,7 +253,8 @@
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
- ast_dsp_set_threshold(sildet, 256);
+        silencethreshold = ast_dsp_get_threshold_from_settings();
+ ast_dsp_set_threshold(sildet, silencethreshold);
}

/* Create the directory if it does not exist. */
Index: apps/app_followme.c
===================================================================
--- apps/app_followme.c (revision 89245)
+++ apps/app_followme.c (working copy)
@@ -887,9 +887,13 @@
char *argstr;
char namerecloc[255];
int duration = 0;
+    int silencethreshold;
struct ast_channel *caller;
struct ast_channel *outbound;
static char toast[80];
+
+
+    silencethreshold = ast_dsp_get_threshold_from_settings();

AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(followmeid);
@@ -959,7 +963,7 @@
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, silencethreshold, 0, NULL) < 0)
goto outrun;

if (!ast_fileexists(namerecloc, NULL, chan->language))
Index: include/asterisk/dsp.h
===================================================================
--- include/asterisk/dsp.h (revision 89245)
+++ include/asterisk/dsp.h (working copy)
@@ -80,6 +80,10 @@
   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 */
int ast_dsp_busydetect(struct ast_dsp *dsp);
@@ -108,4 +112,7 @@
/*! \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(void);
+
#endif /* _ASTERISK_DSP_H */
Index: main/app.c
===================================================================
--- main/app.c (revision 89245)
+++ main/app.c (working copy)
@@ -1262,7 +1262,7 @@

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;
@@ -1280,6 +1280,8 @@

cmd = '3'; /* Want to start by recording */

+    silencethreshold = ast_dsp_get_threshold_from_settings();
+
while ((cmd >= 0) && (cmd != 't')) {
switch (cmd) {
case '1':
Index: main/dsp.c
===================================================================
--- main/dsp.c (revision 89245)
+++ main/dsp.c (working copy)
@@ -60,6 +60,7 @@
#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 {
@@ -181,6 +182,9 @@
#define BELL_MF_RELATIVE_PEAK 12.6    /* 11dB */
#endif

+/*config file for dsp settings*/
+#define CONFIG_FILE_NAME "dsp.conf"
+
#if !defined(BUSYDETECT_MARTIN) && !defined(BUSYDETECT) && !defined(BUSYDETECT_TONEONLY) && !defined(BUSYDETECT_COMPARE_TONE_AND_SILENCE)
#define BUSYDETECT_MARTIN
#endif
@@ -1197,7 +1201,7 @@
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;
@@ -1249,6 +1253,9 @@
}
if (totalsilence)
*totalsilence = dsp->totalsilence;
+    if (totalnoise)
+        *totalnoise = dsp->totalnoise;
+
return res;
}

@@ -1405,9 +1412,28 @@
}
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)
{
int silence;
@@ -1765,3 +1791,34 @@
{
return dsp->tcount;
}
+
+int ast_dsp_get_threshold_from_settings()
+{
+    static int threshold = -1;
+    const int default_threshold = 256;
+
+    struct ast_flags config_flags = { 0 };
+    struct ast_config *cfg;
+    struct ast_variable *var;
+
+    if(-1 == threshold) {
+        threshold = default_threshold;
+        cfg = ast_config_load(CONFIG_FILE_NAME, config_flags);
+        if (!cfg) {
+            ast_log(LOG_WARNING, "No %s file :(\n", CONFIG_FILE_NAME);
+        }
+        else {
+            for (var = ast_variable_browse(cfg, "default"); var; var = var->next) {
+                if (!strcasecmp(var->name, "silencethreshold")){
+                    if(sscanf(var->value, "%d", &threshold) != 1) {
+                        ast_log(LOG_WARNING, "Error processing file %s :(\n", CONFIG_FILE_NAME);
+                        threshold = default_threshold;
+                    }
+                    break;
+                }
+            }
+        }
+    }
+    return threshold;
+}
+
Index: res/res_agi.c
===================================================================
--- res/res_agi.c (revision 89245)
+++ res/res_agi.c (working copy)
@@ -892,7 +892,9 @@
char *silencestr=NULL;
int rfmt=0;

+    int silencethreshold;

+
/* XXX EAGI FIXME XXX */

if (argc < 6)
@@ -932,7 +934,8 @@
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
- ast_dsp_set_threshold(sildet, 256);
+        silencethreshold = ast_dsp_get_threshold_from_settings();
+ ast_dsp_set_threshold(sildet, silencethreshold);
}

/* backward compatibility, if no offset given, arg[6] would have been
Comments:By: Jason Parker (jparker) 2007-11-13 12:31:58.000-0600

Interesting.  I can see how this would be useful.

It does seem like it might be stepping on the toes of app_amd though.  Thoughts?
By: Philipp Skadorov (philipps) 2007-11-13 15:24:38.000-0600

qwell,
Yes, it overlaps with AMD, a bit different though.
My usecase is to wait for voice activity on the channel for a certain amount of time.
If there's nothing above silence threshold - hangup.
AMD is too excessive for this case.
By: Joshua C. Colp (jcolp) 2007-11-19 08:30:48.000-0600

Changed to feedback until license agreement issue is straightened out.
By: Philipp Skadorov (philipps) 2007-12-09 14:15:47.000-0600

Pls pickup silencethreshold_waitforsilence_tabified.patch with tabs put instead of spaces, where appropriate.
By: Tilghman Lesher (tilghman) 2007-12-28 09:28:17.000-0600

You've changed the help text to use "|" as the delimiter instead of ",".  That is no longer correct.  In trunk, the application delimiter is ",".
By: Philipp Skadorov (philipps) 2008-01-12 11:00:11.000-0600

Delimiter issue is fixed in waitfornoise_120108.patch
This patch correlates to silence_threshold_120108.patch from bugASTERISK-1117235
By: jmls (jmls) 2008-02-17 12:53:55.000-0600

Corydon76, can this now be applied to trunk ?
By: Tilghman Lesher (tilghman) 2008-02-17 23:38:03.000-0600

Not quite yet.  It's an entire API for a single setting.  If we're going to do this, then it needs to be generic for a whole host of possible settings, even if the others aren't yet implemented.
By: Tilghman Lesher (tilghman) 2008-02-18 00:30:01.000-0600

Here's my cleanup of the feature patch.  Needs testing.
By: Tilghman Lesher (tilghman) 2008-02-18 00:30:50.000-0600

Also, what this patch does NOT include are the addition of the dsp.conf.sample file and the additions to the CHANGES document.
By: Tilghman Lesher (tilghman) 2008-02-26 12:07:28.000-0600

philipps: ping
By: Philipp Skadorov (philipps) 2008-02-26 14:15:59.000-0600

Corydon76: ping reply. I'll put the sample config and notes till the end of the week.


By: Tilghman Lesher (tilghman) 2008-02-26 15:23:10.000-0600

philipps: Actually, what I'd like is some testing of the cleanup.
By: Philipp Skadorov (philipps) 2008-02-26 15:48:04.000-0600

Please specify the test cases then.
By: Tilghman Lesher (tilghman) 2008-02-26 17:22:05.000-0600

Um, test it to ensure that it behaves the way you wrote the original code to do?  This is just a cleanup of the code (that makes it respect our coding guidelines and code style), but I may have inadvertently introduced some bugs while cleaning up the code.
By: Philipp Skadorov (philipps) 2008-03-01 09:40:38.000-0600

Corydon76, it seems working.
By: Digium Subversion (svnbot) 2008-03-05 10:20:28.000-0600

Repository: asterisk
Revision: 106072

U   trunk/CHANGES
U   trunk/UPGRADE.txt
U   trunk/apps/app_amd.c
U   trunk/apps/app_dial.c
U   trunk/apps/app_followme.c
U   trunk/apps/app_meetme.c
U   trunk/apps/app_minivm.c
U   trunk/apps/app_record.c
U   trunk/apps/app_voicemail.c
U   trunk/apps/app_waitforsilence.c
A   trunk/configs/dsp.conf.sample
U   trunk/include/asterisk/dsp.h
U   trunk/main/app.c
U   trunk/main/asterisk.c
U   trunk/main/dsp.c
U   trunk/main/loader.c
U   trunk/res/res_agi.c

------------------------------------------------------------------------
r106072 | tilghman | 2008-03-05 10:20:23 -0600 (Wed, 05 Mar 2008) | 7 lines

Create a centralized configuration option for silencethreshold
(closes issue ASTERISK-10755)
Reported by: philipps
Patches:
      20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
Tested by: philipps

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=106072
By: Digium Subversion (svnbot) 2008-03-05 13:40:30.000-0600

Repository: asterisk
Revision: 106172

_U  team/murf/sched-rbtree/
U   team/murf/sched-rbtree/CHANGES
U   team/murf/sched-rbtree/UPGRADE.txt
U   team/murf/sched-rbtree/apps/app_amd.c
U   team/murf/sched-rbtree/apps/app_dial.c
U   team/murf/sched-rbtree/apps/app_followme.c
U   team/murf/sched-rbtree/apps/app_meetme.c
U   team/murf/sched-rbtree/apps/app_minivm.c
U   team/murf/sched-rbtree/apps/app_record.c
U   team/murf/sched-rbtree/apps/app_talkdetect.c
U   team/murf/sched-rbtree/apps/app_voicemail.c
U   team/murf/sched-rbtree/apps/app_waitforsilence.c
U   team/murf/sched-rbtree/channels/chan_sip.c
U   team/murf/sched-rbtree/channels/chan_zap.c
A   team/murf/sched-rbtree/configs/dsp.conf.sample
U   team/murf/sched-rbtree/include/asterisk/dsp.h
U   team/murf/sched-rbtree/include/asterisk/sched.h
U   team/murf/sched-rbtree/main/app.c
U   team/murf/sched-rbtree/main/asterisk.c
U   team/murf/sched-rbtree/main/dsp.c
U   team/murf/sched-rbtree/main/loader.c
U   team/murf/sched-rbtree/res/res_agi.c

------------------------------------------------------------------------
r106172 | murf | 2008-03-05 13:40:24 -0600 (Wed, 05 Mar 2008) | 57 lines

Merged revisions 106003-106170 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r106036 | tilghman | 2008-03-05 08:23:32 -0700 (Wed, 05 Mar 2008) | 15 lines
 
 Merged revisions 106015 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
 r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines
 
 Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
 (closes issue ASTERISK-11574)
  Reported by: slavon
  Patches:
        sch2.patch uploaded by slavon (license 288)
 (Patch slightly modified by me)
 
 ........
................
 r106040 | kpfleming | 2008-03-05 08:40:40 -0700 (Wed, 05 Mar 2008) | 15 lines
 
 Merged revisions 106038 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
 r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines
 
 when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
 
 (closes issue ASTERISK-11370)
 Reported by: mavetju
 Tested by: mavetju
 
 
 ........
................
 r106072 | tilghman | 2008-03-05 09:23:44 -0700 (Wed, 05 Mar 2008) | 7 lines
 
 Create a centralized configuration option for silencethreshold
 (closes issue ASTERISK-10755)
  Reported by: philipps
  Patches:
        20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
  Tested by: philipps
................
 r106110 | file | 2008-03-05 09:39:22 -0700 (Wed, 05 Mar 2008) | 2 lines
 
 Fix code up to what it was meant to be.
................
 r106139 | tilghman | 2008-03-05 10:40:42 -0700 (Wed, 05 Mar 2008) | 3 lines
 
 Should check these values for non-NULL before scanning.
 (Closes issue ASTERISK-11580)
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=106172
By: Digium Subversion (svnbot) 2008-03-05 14:41:39.000-0600

Repository: asterisk
Revision: 106177

_U  team/murf/bug11210/
U   team/murf/bug11210/CHANGES
U   team/murf/bug11210/UPGRADE.txt
U   team/murf/bug11210/apps/app_amd.c
U   team/murf/bug11210/apps/app_dial.c
U   team/murf/bug11210/apps/app_followme.c
U   team/murf/bug11210/apps/app_meetme.c
U   team/murf/bug11210/apps/app_minivm.c
U   team/murf/bug11210/apps/app_record.c
U   team/murf/bug11210/apps/app_talkdetect.c
U   team/murf/bug11210/apps/app_voicemail.c
U   team/murf/bug11210/apps/app_waitforsilence.c
U   team/murf/bug11210/channels/chan_sip.c
U   team/murf/bug11210/channels/chan_zap.c
A   team/murf/bug11210/configs/dsp.conf.sample
U   team/murf/bug11210/include/asterisk/dsp.h
U   team/murf/bug11210/include/asterisk/sched.h
U   team/murf/bug11210/main/app.c
U   team/murf/bug11210/main/asterisk.c
U   team/murf/bug11210/main/dsp.c
U   team/murf/bug11210/main/loader.c
U   team/murf/bug11210/res/res_agi.c

------------------------------------------------------------------------
r106177 | murf | 2008-03-05 14:41:31 -0600 (Wed, 05 Mar 2008) | 57 lines

Merged revisions 106001-106176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r106036 | tilghman | 2008-03-05 08:23:32 -0700 (Wed, 05 Mar 2008) | 15 lines
 
 Merged revisions 106015 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
 r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines
 
 Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
 (closes issue ASTERISK-11574)
  Reported by: slavon
  Patches:
        sch2.patch uploaded by slavon (license 288)
 (Patch slightly modified by me)
 
 ........
................
 r106040 | kpfleming | 2008-03-05 08:40:40 -0700 (Wed, 05 Mar 2008) | 15 lines
 
 Merged revisions 106038 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
 r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines
 
 when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
 
 (closes issue ASTERISK-11370)
 Reported by: mavetju
 Tested by: mavetju
 
 
 ........
................
 r106072 | tilghman | 2008-03-05 09:23:44 -0700 (Wed, 05 Mar 2008) | 7 lines
 
 Create a centralized configuration option for silencethreshold
 (closes issue ASTERISK-10755)
  Reported by: philipps
  Patches:
        20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
  Tested by: philipps
................
 r106110 | file | 2008-03-05 09:39:22 -0700 (Wed, 05 Mar 2008) | 2 lines
 
 Fix code up to what it was meant to be.
................
 r106139 | tilghman | 2008-03-05 10:40:42 -0700 (Wed, 05 Mar 2008) | 3 lines
 
 Should check these values for non-NULL before scanning.
 (Closes issue ASTERISK-11580)
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=106177
By: Digium Subversion (svnbot) 2008-03-05 18:13:58.000-0600

Repository: asterisk
Revision: 106312

_U  branches/1.6.0/

------------------------------------------------------------------------
r106312 | russell | 2008-03-05 18:13:51 -0600 (Wed, 05 Mar 2008) | 14 lines

Blocked revisions 106072 via svnmerge

........
r106072 | tilghman | 2008-03-05 10:23:44 -0600 (Wed, 05 Mar 2008) | 7 lines

Create a centralized configuration option for silencethreshold
(closes issue ASTERISK-10755)
Reported by: philipps
Patches:
      20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
Tested by: philipps

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=106312