[asterisk-dev] [Code Review] 3736: Media Formats: Fix crash bugs

[Corey Farrell]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3736/#review12540
-----------------------------------------------------------


Have you done leak testing?  I suspect the change to __frame_free will result in leaks that will be reported by REF_DEBUG.


team/group/media_formats-reviewed-trunk/main/frame.c
<https://reviewboard.asterisk.org/r/3736/#comment22836>

    We can just ao2_cleanup here since frames are zero'ed when retrieved from cached storage.



team/group/media_formats-reviewed-trunk/main/frame.c
<https://reviewboard.asterisk.org/r/3736/#comment22835>

    We can just ao2_cleanup here since fr will be freed.


- Corey Farrell


On July 10, 2014, 11:58 a.m., opticron wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3736/
> -----------------------------------------------------------
> 
> (Updated July 10, 2014, 11:58 a.m.)
> 
> 
> Review request for Asterisk Developers, Corey Farrell, Joshua Colp, and Matt Jordan.
> 
> 
> Bugs: ASTERISK-23960
>     https://issues.asterisk.org/jira/browse/ASTERISK-23960
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This contains the fixes necessary to get a PJSIP call up and playing files from the Playback() application with a translation path in the mix.
> 
> * channel.c: The set format helper functions were not actually setting the newly chosen formats resulting in no audio.
> * frame.c: The format on the frame was being overzealously unreffed resulting in a crash.
> * sorcery.c: The codec retrieval code was using the wrong level of indirection for ast_format_cap structures resuting in a crash for "pjsip show endpoint x"
> * translate.c: The chosen codecs were being set backward on set vs native for what was actually desired causing incorrect codecs to be chosen.
> * res_pjsip_sdp_rtp.c: An ast_rtp_codecs struct was not being initialized properly causing a crash.
> 
> 
> Diffs
> -----
> 
>   team/group/media_formats-reviewed-trunk/res/res_pjsip_sdp_rtp.c 418253 
>   team/group/media_formats-reviewed-trunk/main/translate.c 418253 
>   team/group/media_formats-reviewed-trunk/main/sorcery.c 418253 
>   team/group/media_formats-reviewed-trunk/main/frame.c 418253 
>   team/group/media_formats-reviewed-trunk/main/channel.c 418253 
> 
> Diff: https://reviewboard.asterisk.org/r/3736/diff/
> 
> 
> Testing
> -------
> 
> Call testing.
> 
> 
> Thanks,
> 
> opticron
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140710/65f77618/attachment.html>