[asterisk-commits] dvossel: trunk r271551 - /trunk/res/res_rtp_asterisk.c

[SVN commits to the Asterisk project]

Author: dvossel
Date: Mon Jun 21 15:33:41 2010
New Revision: 271551

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=271551
Log:
fixes logic error introduced by slin16 sip support

Modified:
    trunk/res/res_rtp_asterisk.c

Modified: trunk/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_rtp_asterisk.c?view=diff&rev=271551&r1=271550&r2=271551
==============================================================================
--- trunk/res/res_rtp_asterisk.c (original)
+++ trunk/res/res_rtp_asterisk.c Mon Jun 21 15:33:41 2010
@@ -2230,8 +2230,9 @@
 
 	if (rtp->f.subclass.codec & AST_FORMAT_AUDIO_MASK) {
 		rtp->f.samples = ast_codec_get_samples(&rtp->f);
-		if (rtp->f.subclass.codec == AST_FORMAT_SLINEAR || AST_FORMAT_SLINEAR16)
+		if ((rtp->f.subclass.codec == AST_FORMAT_SLINEAR) || (rtp->f.subclass.codec == AST_FORMAT_SLINEAR16)) {
 			ast_frame_byteswap_be(&rtp->f);
+		}
 		calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
 		/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
 		ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);