[asterisk-dev] 16khz/8khz Translation path

[asterisk at ntplx.net]

Still seeing the problem. I opened a new bug 18151:

https://issues.asterisk.org/view.php?id=18151

-- 

Andrew Lindh
<andrew at netplex.net>



Quoting David Vossel <dvossel at digium.com>:

> ----- Original Message -----
>> From: "Russell Bryant" <russell at digium.com>
>> To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>> Sent: Monday, September 27, 2010 9:24:37 AM
>> Subject: Re: [asterisk-dev] 16khz/8khz Translation path
>> On Mon, 2010-09-27 at 09:20 -0500, David Vossel wrote:
>> > I have noticed this as well and this needs to be investigated. It is
>> > possible that the second of audio we use to calculate the the least
>> > cost translation path is not accurate enough. If it turns out that
>> > G722 is actually somehow more cost efficient than using the
>> > libresample library then we need to come up with a way to force
>> > lossless codec translations when they are available.
>>
>> I think that would be a good thing to put in there regardless. It
>> shouldn't be too hard since you already put in some good hooks for
>> making the cost calculation take things other than time into account.
>>
>> --
>> Russell Bryant
>> Digium, Inc. | Engineering Manager, Open Source Software
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> jabber: rbryant at digium.com -=- skype: russell-bryant
>> www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org
>
>
> Looking back on this, I do not see how it is possible for the  
> translation path to go through g722 when libresample is installed.   
> I have made some changes in this area recently that should guarantee  
> slin->slin16 and slin16-slin translations unless the translation  
> costs are unreasonably inaccurate.  The costs would have to be  
> wildly inaccurate for this to occur, and this was not the case when  
> I just tested it.
>
> Since you are seeing this, we should verify a few things.  First,  
> are you using a 1.8 revision greater than 282047.  I made changes  
> during that revision that directly relate to what we are seeing  
> here.  If you are using a revision with the new codec translation  
> changes in it, please create a mantis issue so we can track this.   
> In the issue, please include your 'core show translation' output.
>
> Thanks,
>
> David Vossel
> Digium, Inc. | Software Developer, Open Source Software
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
> The_Boy_Wonder in #asterisk-dev
>
> --
> _____________________________________________________________________
> -- 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
>
>