From 1a7e1aee5eca1d78d7bc97ce38e63cd10c90f018 Mon Sep 17 00:00:00 2001 From: David Vossel Date: Mon, 21 Jun 2010 20:33:41 +0000 Subject: fixes logic error introduced by slin16 sip support git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- res/res_rtp_asterisk.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index edea6112c..0de03de28 100644 --- 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); -- cgit v1.2.3