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| author | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-03-05 18:29:38 +0000 |
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| committer | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-03-05 18:29:38 +0000 |
| commit | 5d5eb54ba76e257b12f5e74bfdfc925e75518a40 (patch) | |
| tree | 83571b4f0d3351db8bf0aff77046f624f430d382 /res | |
| parent | 78d18e52a495a2b1f964a2dceeb7fcc705a79d73 (diff) | |
Merged revisions 180372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Review: http://reviewboard.digium.com/r/184/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180373 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'res')
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