[asterisk-dev] [Code Review] 3671: media_formats: res_rtp_asterisk and other fixes

[Joshua Colp]

> On June 25, 2014, 11:45 a.m., Joshua Colp wrote:
> > /team/group/media_formats-reviewed-trunk/main/format.c, lines 200-202
> > <https://reviewboard.asterisk.org/r/3671/diff/1/?file=60620#file60620line200>
> >
> >     Just curious - does this happen?
> 
> Corey Farrell wrote:
>     This could happen if someone configures a blank format_cap (disallow=all) - the first format would be NULL (ast_best_codec).
>     
>     Maybe an assert should go here until we get a better handle on things?  If NULL's to this procedure are possible we probably want to check this second (if both are NULL then they are equal).

My question becomes: Should this accept NULLs or should the caller see that something they expected to be non-NULL is NULL and thus should print out an error/warning/something.


On June 25, 2014, 11:45 a.m., Corey Farrell wrote:
> > For cases where is it doing a cleanup/copy it might be useful to add a BUGBUG for when replace exists.
> 
> Corey Farrell wrote:
>     ao2_replace already exists.  Do we want to just directly use that or create a #define ast_format_replace(dst,src) ao2_replace(dst,src) ?
>     
>     I prefer we just use ao2_replace directly, I think this makes the code clearer.  I think of the #define ast_format_copy(format) as a compatibility macro that should go away during Matt's API rename.

I'm down with that.


- Joshua


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On June 25, 2014, 12:14 p.m., Corey Farrell wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3671/
> -----------------------------------------------------------
> 
> (Updated June 25, 2014, 12:14 p.m.)
> 
> 
> Review request for Asterisk Developers, Joshua Colp and Matt Jordan.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This update gives media_formats the ability to receive a call using chan_sip.  Possibly other channel drivers might work, I haven't tried them.
> 
> * ast_format_cap_is_compatible_format needs to be checked against AST_FORMAT_CMP_NOT_EQUAL, not zero/non-zero.  All calls to ast_format_cap_is_compatible_format were fixed.
> * res_rtp_asterisk was updated by Matt Jordan, along with related changes to codec.c, codec.h, format.c, format.c and codec_builtin.c.
> * Switch ast_format_copy from function to macro to ao2_bump.  This allows REF_DEBUG to give better results.
> 
> 
> Diffs
> -----
> 
>   /team/group/media_formats-reviewed-trunk/res/res_speech.c 417190 
>   /team/group/media_formats-reviewed-trunk/res/res_rtp_asterisk.c 417190 
>   /team/group/media_formats-reviewed-trunk/main/translate.c 417190 
>   /team/group/media_formats-reviewed-trunk/main/frame.c 417190 
>   /team/group/media_formats-reviewed-trunk/main/format.c 417190 
>   /team/group/media_formats-reviewed-trunk/main/codec_builtin.c 417190 
>   /team/group/media_formats-reviewed-trunk/main/codec.c 417190 
>   /team/group/media_formats-reviewed-trunk/main/channel.c 417190 
>   /team/group/media_formats-reviewed-trunk/main/bridge.c 417190 
>   /team/group/media_formats-reviewed-trunk/include/asterisk/format.h 417190 
>   /team/group/media_formats-reviewed-trunk/include/asterisk/codec.h 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_unistim.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_skinny.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_sip.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_pjsip.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_oss.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_nbs.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_motif.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_mgcp.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_jingle.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_h323.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_gtalk.c 417190 
>   /team/group/media_formats-reviewed-trunk/channels/chan_alsa.c 417190 
>   /team/group/media_formats-reviewed-trunk/addons/chan_ooh323.c 417190 
>   /team/group/media_formats-reviewed-trunk/addons/chan_mobile.c 417190 
> 
> Diff: https://reviewboard.asterisk.org/r/3671/diff/
> 
> 
> Testing
> -------
> 
> Called from Asterisk 11 to a test server with this code, I was able to hear the 'invalid' message, everything seemed during the call.  I received TONS of ao2 frack's when stopping Asterisk.  The sip.conf peer on both Asterisk servers was setup for disallow=all / allow=ulaw.
> 
> 
> Thanks,
> 
> Corey Farrell
> 
>

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