[test-results] [Bamboo] Asterisk - Team Branches > Pimp My SIP > #136 was SUCCESSFUL (with 208 tests). Change made by root.

[Bamboo]

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Asterisk - Team Branches > Pimp My SIP > #136 was successful.
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Code has been updated by root.
All 2 jobs passed with 208 tests in total.

http://bamboo.asterisk.org/browse/ASTTEAM-PIMPMYSIP-136/


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Code Changes
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root (382237):

>Merged revisions 382232,382236 via svnmerge from 
>file:///srv/subversion/repos/asterisk/trunk
>
>................
>  r382232 | mjordan | 2013-02-28 10:56:20 -0600 (Thu, 28 Feb 2013) | 33 lines
>  
>  Let channels joining a MeetMe conference opt out of the denoiser
>  
>  For some channel drivers, specifically those that have a varying rate in the
>  number of audio samples, the audio quality for a MeetMe conference can be
>  exceedingly poor. This is due to a unilateral application of the DENOISE
>  function in func_speex to channels joining the conference.
>  
>  The denoiser function in the speex library is initialized with the number of
>  audio samples in each sample that will be provided to it. If the number of
>  audio samples changes, the denoiser has to be thrown away and re-initialized.
>  
>  While this could be worked around by removing func_speex, that doesn't help
>  if you actually use the denoiser with other channels on the system.
>  
>  This patches does the following:
>   * Checks for the presence of func_speex as opposed to codec_speex when
>     determining if the DENOISE function is present (which is where the function
>     is actually implemented)
>   * Adds an option to MeetMe 'n' that causes the denoiser to not be applied
>     to a channel when it joins. This keeps the current behavior the default, but
>     let's users disable the denoiser if it causes problems on their system.
>  
>  Review: https://reviewboard.asterisk.org/r/2358
>  
>  (closes issue AST-1062)
>  Reported by: Thomas Arimont
>  ........
>  
>  Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 382230 from http://svn.asterisk.org/svn/asterisk/branches/11
>................
>  r382236 | mjordan | 2013-02-28 11:17:35 -0600 (Thu, 28 Feb 2013) | 25 lines
>  
>  Prevent deadlock in chan_iax2 when attempting to set caller ID
>  
>  A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it
>  already holds the iax2 private lock and improperly fails to obtain the channel
>  lock before calling ast_set_callerid. By not safely obtaining the channel lock,
>  a locking inversion can take place, causing a deadlock.
>  
>  This patch solves this by calling the required deadlock avoidance functions
>  that obtain the channel lock before setting the caller ID.
>  
>  Thanks to Pavel for fixing my syntax errors and testing this patch out.
>  
>  (closes issue ASTERISK-21128)
>  Reported by: Pavel Troller
>  Tested by: Pavel Troller
>  patches:
>    ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
>    ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302)
>  ........
>  
>  Merged revisions 382233 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 382234 from http://svn.asterisk.org/svn/asterisk/branches/11
>................
>



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Tests
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Fixed Tests (1)
   - AsteriskUnitTests: /main/threadpool/auto increment

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