fixes logic error introduced by slin16 sip support
authorDavid Vossel <dvossel@digium.com>
Mon, 21 Jun 2010 20:33:41 +0000 (20:33 +0000)
committerDavid Vossel <dvossel@digium.com>
Mon, 21 Jun 2010 20:33:41 +0000 (20:33 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res/res_rtp_asterisk.c

index edea611..0de03de 100644 (file)
@@ -2230,8 +2230,9 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 
        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);