]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Merged revisions 177101 via svnmerge from
authorRussell Bryant <russell@russellbryant.com>
Wed, 18 Feb 2009 19:30:18 +0000 (19:30 +0000)
committerRussell Bryant <russell@russellbryant.com>
Wed, 18 Feb 2009 19:30:18 +0000 (19:30 +0000)
https://origsvn.digium.com/svn/asterisk/trunk

........
r177101 | russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines

Re-add 'o' option to MeetMe, reverting rev 62297.

Enabling this option by default proved to be a bad idea, as the talker detection
is not very reliable.  So, make it optional again, and off by default.

(issue #13801)
Reported by: justdave

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@177158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_meetme.c

index f1406d49a0f5b01bd0ccd224618fbabf727f5a77..0e5ea3e35c2eff959c6f660d3e1f73245c6e8887 100644 (file)
@@ -143,6 +143,8 @@ enum {
        CONFFLAG_EMPTY = (1 << 18),
        CONFFLAG_EMPTYNOPIN = (1 << 19),
        CONFFLAG_ALWAYSPROMPT = (1 << 20),
+       /*! If set, treat talking users as muted users */
+       CONFFLAG_OPTIMIZETALKER = (1 << 21),
        /*! If set, won't speak the extra prompt when the first person 
         *  enters the conference */
        CONFFLAG_NOONLYPERSON = (1 << 22),
@@ -187,6 +189,7 @@ AST_APP_OPTIONS(meetme_opts, BEGIN_OPTIONS
        AST_APP_OPTION('I', CONFFLAG_INTROUSERNOREVIEW ),
        AST_APP_OPTION_ARG('M', CONFFLAG_MOH, OPT_ARG_MOH_CLASS ),
        AST_APP_OPTION('m', CONFFLAG_STARTMUTED ),
+       AST_APP_OPTION('o', CONFFLAG_OPTIMIZETALKER ),
        AST_APP_OPTION('P', CONFFLAG_ALWAYSPROMPT ),
        AST_APP_OPTION_ARG('p', CONFFLAG_KEYEXIT, OPT_ARG_EXITKEYS ),
        AST_APP_OPTION('q', CONFFLAG_QUIET ),
@@ -2147,7 +2150,7 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, int c
                        x = 1;
                        ast_channel_setoption(chan, AST_OPTION_TONE_VERIFY, &x, sizeof(char), 0);
                }       
-               if (!(confflags & CONFFLAG_MONITOR) && !(dsp = ast_dsp_new())) {
+               if ((confflags & CONFFLAG_OPTIMIZETALKER) && !(confflags & CONFFLAG_MONITOR) && !(dsp = ast_dsp_new())) {
                        ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
                        res = -1;
                }
@@ -2463,7 +2466,7 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, int c
                                        if (user->talk.actual)
                                                ast_frame_adjust_volume(f, user->talk.actual);
 
-                                       if (!(confflags & CONFFLAG_MONITOR)) {
+                                       if ((confflags & CONFFLAG_OPTIMIZETALKER) && !(confflags & CONFFLAG_MONITOR)) {
                                                int totalsilence;
 
                                                if (user->talking == -1)
@@ -2506,8 +2509,9 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, int c
                                                   don't want to block, but we do want to at least *try*
                                                   to write out all the samples.
                                                 */
-                                               if (user->talking)
+                                               if (user->talking && !(confflags & CONFFLAG_OPTIMIZETALKER)) {
                                                        careful_write(fd, f->data.ptr, f->datalen, 0);
+                                               }
                                        }
                                } else if (((f->frametype == AST_FRAME_DTMF) && (f->subclass == '*') && (confflags & CONFFLAG_STARMENU)) || ((f->frametype == AST_FRAME_DTMF) && menu_active)) {
                                        if (confflags & CONFFLAG_PASS_DTMF)
@@ -2725,10 +2729,11 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, int c
                                        fr.samples = res / 2;
                                        fr.data.ptr = buf;
                                        fr.offset = AST_FRIENDLY_OFFSET;
-                                       if (!user->listen.actual && 
-                                               ((confflags & CONFFLAG_MONITOR) || 
+                                       if (!user->listen.actual &&
+                                               ((confflags & CONFFLAG_MONITOR) ||
                                                 (user->adminflags & (ADMINFLAG_MUTED | ADMINFLAG_SELFMUTED)) ||
-                                                (!user->talking)) ) {
+                                                (!user->talking && (confflags & CONFFLAG_OPTIMIZETALKER))
+                                                )) {
                                                int idx;
                                                for (idx = 0; idx < AST_FRAME_BITS; idx++)
                                                        if (chan->rawwriteformat & (1 << idx))