From 6fbec565b2d014b8dd032f3dcc111ef746454f4b Mon Sep 17 00:00:00 2001 From: dvossel Date: Thu, 17 Jun 2010 18:36:06 +0000 Subject: 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 --- res/res_rtp_asterisk.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'res') 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 */ -- cgit v1.2.3