fixes logic error introduced by slin16 sip support

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
15 years ago · 1a7e1aee5e
parent 63fd368411
commit 1a7e1aee5e
1 changed files with 2 additions and 1 deletions
--- 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);