diff options
| author | David Vossel <dvossel@digium.com> | 2010-06-21 20:33:41 +0000 |
|---|---|---|
| committer | David Vossel <dvossel@digium.com> | 2010-06-21 20:33:41 +0000 |
| commit | 1a7e1aee5eca1d78d7bc97ce38e63cd10c90f018 (patch) | |
| tree | 7aab3df1106468afc03e1e92150a8663f39a4cea /res/res_rtp_asterisk.c | |
| parent | 63fd36841188a083df8586760c63e6e9df819488 (diff) | |
fixes logic error introduced by slin16 sip support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_rtp_asterisk.c')
| -rw-r--r-- | res/res_rtp_asterisk.c | 3 |
1 files changed, 2 insertions, 1 deletions
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); |