diff options
| author | dvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-06-17 18:36:06 +0000 |
|---|---|---|
| committer | dvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-06-17 18:36:06 +0000 |
| commit | 6fbec565b2d014b8dd032f3dcc111ef746454f4b (patch) | |
| tree | c62961225d6079081762d5ed7e6a1ca96472b069 /res | |
| parent | 637447be7d05aaac944171053df4f82e2cb8ab07 (diff) | |
adds support for slin16 in sip
(closes issue #16153)
Reported by: kfister
Patches:
16153-1.6.2.0-rc5.patch uploaded by kfister (license 912)
slin16.sip.patch.1 uploaded by malcolmd (license 924)
Tested by: kfister, malcolmd
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271261 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'res')
| -rw-r--r-- | res/res_rtp_asterisk.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index 7fe8ff9ba..edea6112c 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -2230,7 +2230,7 @@ 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) + if (rtp->f.subclass.codec == AST_FORMAT_SLINEAR || 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 */ |