[asterisk-dev] [Code Review] Fix a variety of memory leaks

Mark Michelson reviewboard at asterisk.org
Fri May 18 08:48:48 CDT 2012


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1922/#review6258
-----------------------------------------------------------

Ship it!


There are some leaks you're fixing where under current conditions, they cannot occur. However, the fixes you're putting in place are forward-looking and of course shut the static analyzer up. So go for it!

- Mark


On May 18, 2012, 8:41 a.m., Matt Jordan wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1922/
> -----------------------------------------------------------
> 
> (Updated May 18, 2012, 8:41 a.m.)
> 
> 
> Review request for Asterisk Developers, otherwiseguy, rmudgett, and opticron.
> 
> 
> Summary
> -------
> 
> This fixes a number of memory leaks in core modules (and a few modules that are extended support, but were easy to fix) that were found by a static analysis tool.  A brief summary of the changes:
> 
> * app_minivm: free ast_str objects on off nominal paths
> * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure
> * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked
> * app_read: dispose of the created silence detector in the presence of off nominal circumstances
> * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option
> * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process
> * chan_iax2: properly unref peer in CLI command "iax2 unregister"
> * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests
> * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read
> * func_odbc: free resultset in off nominal paths of odbc_read
> * cli: free match_list in off nominal paths of CLI match completion
> * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed
> * data: free XML nodes in various places
> * enum: free context buffer in off nominal paths
> * features: free ast_call_feature in off nominal paths of applicationmap config processing
> * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method.  Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated.  The method will now not allocate the ast_sockaddr struct if it will return failure.
> * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context
> * xmldoc: cleanup various nodes when we no longer need them
> * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths
> * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value
> * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately
> * res_jabber: free buffer in off nominal path
> * res_musiconhold: close the DIR* object in off nominal paths
> * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object
> * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object
> 
> 
> This addresses bug ASTERISK-19665.
>     https://issues.asterisk.org/jira/browse/ASTERISK-19665
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/apps/app_minivm.c 366879 
>   /branches/1.8/apps/app_page.c 366879 
>   /branches/1.8/apps/app_queue.c 366879 
>   /branches/1.8/apps/app_record.c 366879 
>   /branches/1.8/apps/app_voicemail.c 366879 
>   /branches/1.8/channels/chan_dahdi.c 366879 
>   /branches/1.8/channels/chan_iax2.c 366879 
>   /branches/1.8/channels/chan_sip.c 366879 
>   /branches/1.8/channels/sip/config_parser.c 366879 
>   /branches/1.8/funcs/func_dialgroup.c 366879 
>   /branches/1.8/funcs/func_odbc.c 366879 
>   /branches/1.8/main/cli.c 366879 
>   /branches/1.8/main/config.c 366879 
>   /branches/1.8/main/data.c 366879 
>   /branches/1.8/main/editline/readline.c 366879 
>   /branches/1.8/main/editline/term.c 366879 
>   /branches/1.8/main/editline/tokenizer.c 366879 
>   /branches/1.8/main/enum.c 366879 
>   /branches/1.8/main/features.c 366879 
>   /branches/1.8/main/netsock2.c 366879 
>   /branches/1.8/main/pbx.c 366879 
>   /branches/1.8/main/xmldoc.c 366879 
>   /branches/1.8/res/res_calendar.c 366879 
>   /branches/1.8/res/res_calendar_caldav.c 366879 
>   /branches/1.8/res/res_calendar_exchange.c 366879 
>   /branches/1.8/res/res_calendar_icalendar.c 366879 
>   /branches/1.8/res/res_jabber.c 366879 
>   /branches/1.8/res/res_musiconhold.c 366879 
>   /branches/1.8/res/res_rtp_asterisk.c 366879 
>   /branches/1.8/res/res_srtp.c 366879 
> 
> Diff: https://reviewboard.asterisk.org/r/1922/diff
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Matt
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120518/48395d88/attachment-0001.htm>


More information about the asterisk-dev mailing list