diff options
| author | dvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-06-21 20:33:41 +0000 |
|---|---|---|
| committer | dvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-06-21 20:33:41 +0000 |
| commit | c60ee4ec67c5339ef1451526d03ded8ec3949ab4 (patch) | |
| tree | 7aab3df1106468afc03e1e92150a8663f39a4cea /res | |
| parent | 4c84c7f85db1da7bc8f3a7bf4731592d7d2a5e64 (diff) | |
fixes logic error introduced by slin16 sip support
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271551 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'res')
| -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); |