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adds support for slin16 in sip

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(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: https://origsvn.digium.com/svn/asterisk/trunk@271261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
David Vossel
2010-06-17 18:36:06 +00:00
parent b00f58da25
commit ba3d1ad680
3 changed files with 7 additions and 4 deletions
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2
res/res_rtp_asterisk.c
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@@ -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 */