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Merged revisions 105933 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/trunk

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r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008) | 13 lines

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/branches/1.6.0@106310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2008-03-06 00:16:30 +00:00
parent 51f5f4470a
commit 78393c6698
3 changed files with 15 additions and 12 deletions
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11
include/asterisk/frame.h
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@@ -611,6 +611,17 @@ int ast_frame_adjust_volume(struct ast_frame *f, int adjustment);
*/
int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2);
/*!
* \brief Get the sample rate for a given format.
*/
static force_inline int ast_format_rate(int format)
{
if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16)
return 16000;
return 8000;
}
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif