fixes logic error introduced by slin16 sip support

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index edea6112c029c5f70c030382c16869d252fdc291..0de03de2831007f3507efb7e4f17951c862855d1 100644 (file)
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -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);