[asterisk-commits] file: branch 1.4 r90588 - /branches/1.4/main/rtp.c

[SVN commits to the Asterisk project]

Author: file
Date: Mon Dec  3 14:05:42 2007
New Revision: 90588

URL: http://svn.digium.com/view/asterisk?view=rev&rev=90588
Log:
Do not create a smoother for G723.1 frames, they need to be left alone to their native 20/24 byte size.

Modified:
    branches/1.4/main/rtp.c

Modified: branches/1.4/main/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/rtp.c?view=diff&rev=90588&r1=90587&r2=90588
==============================================================================
--- branches/1.4/main/rtp.c (original)
+++ branches/1.4/main/rtp.c Mon Dec  3 14:05:42 2007
@@ -2748,7 +2748,7 @@
 		rtp->smoother = NULL;
 	}
 
-	if (!rtp->smoother && subclass != AST_FORMAT_SPEEX) {
+	if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) {
 		struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
 		if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
 			if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {