Hello,
This media format related work seems really exciting. I had been thinking about making some changes on my own to avoid audio/video transcoding between two end points, when they share a common codec. I would like to know that the work which you are doing for this, how feasible it would be to back port your changes on asterisk 11. Regards, Nitesh Bansal On Tue, Jan 21, 2014 at 5:48 PM, Matthew Jordan <[email protected]> wrote: > On Tue, Jan 21, 2014 at 10:17 AM, Sean Darcy <[email protected]> wrote: > > On 01/21/2014 10:14 AM, Joshua Colp wrote: > >> > >> Greetings everyone, > >> > >> I thought I'd give an update on this. The wiki page[1] is now at a state > >> where I'm happy with the design aspect of it. Matt Jordan has also > >> contributed test cases to the wiki page which will be implemented so > >> that the existing code (and new code) will have tests that cover various > >> scenarios. If there are any critical scenarios that have been missed > >> don't hesitate to say something. I've also created a branch[2] where > >> implementation of the proposed design will occur. If there is an aspect > >> of the implementation (or tests) that you would like to help with don't > >> hesitate to respond to this email or join us on the #asterisk-dev IRC > >> channel. Have a spectacular day! > >> > >> [1] https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite > >> [2] http://svn.asterisk.org/svn/asterisk/team/group/media_formats/ > >> > >> Cheers, > >> > > > > Somewhat off topic, but is there a silk codec for 12? > > > > Not yet, but shortly. > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev >
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