app_record: Add option to prevent silence from being truncated
authorSean Bright <sean.bright@gmail.com>
Tue, 14 Feb 2017 14:12:31 +0000 (09:12 -0500)
committerSean Bright <sean.bright@gmail.com>
Tue, 14 Feb 2017 14:35:18 +0000 (09:35 -0500)
When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.

This patch adds the 'u' option to Record() to override that behavior.

ASTERISK-18286 #close
Reported by: var
Patches:
app_record-1.8.7.1.diff (license #6184) patch uploaded by var

Change-Id: Ia1cd163483235efe2db05e52f39054288553b957

CHANGES
apps/app_record.c

diff --git a/CHANGES b/CHANGES
index 1671d36..79a1840 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -80,6 +80,15 @@ app_queue
    using app_queue.
 
 ------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
+------------------------------------------------------------------------------
+
+app_record
+------------------
+ * Added new 'u' option to Record() application which prevents Asterisk from
+   truncating silence from the end of recorded files.
+
+------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 14.2.0 to Asterisk 14.3.0 ------------
 ------------------------------------------------------------------------------
 
index ede50be..0b85ff8 100644 (file)
@@ -79,6 +79,9 @@
                                        <option name="t">
                                                <para>use alternate '*' terminator key (DTMF) instead of default '#'</para>
                                        </option>
+                                       <option name="u">
+                                               <para>Don't truncate recorded silence.</para>
+                                       </option>
                                        <option name="x">
                                                <para>Ignore all terminator keys (DTMF) and keep recording until hangup.</para>
                                        </option>
@@ -131,6 +134,7 @@ enum {
        FLAG_HAS_PERCENT = (1 << 7),
        OPTION_ANY_TERMINATE = (1 << 8),
        OPTION_OPERATOR_EXIT = (1 << 9),
+       OPTION_NO_TRUNCATE = (1 << 10),
 };
 
 AST_APP_OPTIONS(app_opts,{
@@ -141,6 +145,7 @@ AST_APP_OPTIONS(app_opts,{
        AST_APP_OPTION('q', OPTION_QUIET),
        AST_APP_OPTION('s', OPTION_SKIP),
        AST_APP_OPTION('t', OPTION_STAR_TERMINATE),
+       AST_APP_OPTION('u', OPTION_NO_TRUNCATE),
        AST_APP_OPTION('y', OPTION_ANY_TERMINATE),
        AST_APP_OPTION('x', OPTION_IGNORE_TERMINATE),
 });
@@ -192,6 +197,7 @@ static int record_exec(struct ast_channel *chan, const char *data)
        int dspsilence = 0;
        int silence = 0;                /* amount of silence to allow */
        int gotsilence = 0;             /* did we timeout for silence? */
+       int truncate_silence = 1;       /* truncate on complete silence recording */
        int maxduration = 0;            /* max duration of recording in milliseconds */
        int gottimeout = 0;             /* did we timeout for maxduration exceeded? */
        int terminator = '#';
@@ -243,7 +249,10 @@ static int record_exec(struct ast_channel *chan, const char *data)
                        ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", args.silence);
                }
        }
-       
+
+       if (ast_test_flag(&flags, OPTION_NO_TRUNCATE))
+               truncate_silence = 0;
+
        if (args.maxduration) {
                if ((sscanf(args.maxduration, "%30d", &i) == 1) && (i > -1))
                        /* Convert duration to milliseconds */
@@ -443,7 +452,7 @@ static int record_exec(struct ast_channel *chan, const char *data)
                }
        }
 
-       if (gotsilence) {
+       if (gotsilence && truncate_silence) {
                ast_stream_rewind(s, silence - 1000);
                ast_truncstream(s);
        } else if (!gottimeout && f) {