Merged revisions 105932 via svnmerge from
authorRussell Bryant <russell@russellbryant.com>
Wed, 5 Mar 2008 01:54:16 +0000 (01:54 +0000)
committerRussell Bryant <russell@russellbryant.com>
Wed, 5 Mar 2008 01:54:16 +0000 (01:54 +0000)
commita760a033e93f44d84c75c1c672ac70c0dd42d054
tree210032a142b3c0eb0ce597ad9a35516cd3754139
parent39b9f810163ea0c25e78f98907f28563ccbae277
Merged revisions 105932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines

Fix a bug that I just noticed in the RTP code.  The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use.  The len field
represents the number of ms of audio that the frame contains.  It would have
set the value to be twice what it should be.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
include/asterisk/frame.h
main/rtp.c
main/translate.c