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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Hi,<o:p></o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p> </o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>I
asked this on asterisk-users list but did not get a definitive answer. On 1.6.2,
does Meet-me bridge support native WB mixing (meaning no downscaling to SLIN)
for G722 or any other WB codec ? I tried 1.6.2 Beta3 but it sounded like it was
still doing a downgrade to SLIN first.<o:p></o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p> </o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p> </o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Thanks,<o:p></o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Serhad
Doken<o:p></o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p> </o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>--------</span></b><b><span
style='font-size:10.0pt;font-family:Wingdings'>à</span></b><b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p> </o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Michael
Graves<br>
<b>Sent:</b> Thursday, June 18, 2009 4:06 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Wideband (G722) MeetMe<o:p></o:p></span></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt'>I'm
told that Asterisks wideband capability is exclusively based upon 16 khz
sampling. Higher sampling rates, like you might find with CELT, are downsampled
for mixing at 16 khz.<br>
<br>
My guess is that not everyone will be happy with this, especially vendors
trumpeting codecs with higher sampling rates. To handle their output will be more
CPU intensive.<br>
<br>
I'm not an Asterisk insider. I just listen closely when the real insiders speak
up ;-)<br>
<br>
Michael<br>
<br>
--Original Message Text---<br>
<b>From:</b> Doken, Serhad<br>
<b>Date:</b> Thu, 18 Jun 2009 14:15:42 -0700<br>
<br>
<br>
<br>
Thanks Michael. I guess prior to 1.6.2, Asterisk was downgrading streams to
SLIN before mixing and then mixed stream got upgraded to WB. <br>
<br>
<br>
<br>
My question is, with this release, is Asterisk converting WB codecs to SLIN16
and mix them that way ? That seems to be the logical way to me just wanted an
insider expert to confirm/deny that. <br>
<br>
<br>
<br>
Is this the right list to ask that question/find the right contact before I
delve into the code knee deep ? <br>
<br>
<br>
<br>
Serhad Doken <br>
<br>
<br>
<br>
<b>From:</b> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Michael
Graves<br>
<b>Sent:</b> Thursday, June 18, 2009 6:14 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Wideband (G722) MeetMe <br>
<br>
<br>
<br>
<br>
<br>
--Original Message Text---<br>
<b>From:</b> Doken, Serhad<br>
<b>Date:</b> Wed, 17 Jun 2009 16:07:12 -0700<br>
<br>
<br>
<br>
Hi, <br>
<br>
I wanted to follow up on this thread about WB support on the MeetMe bridge that
is in 1.6.2. Does it only work for G722 or any WB codec ? <br>
<br>
I am working with another 16k WB codec that I can transcode to 722 and vice
versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any other
WB codec natively(without downscaling). <br>
<br>
Thanks, <br>
<br>
Serhad Doken <br>
<br>
While not an expert in Asterisk internal, it seems unlikely that Asterisk is
mixing signals in encoded space. It's most likely converting the stream to slin
for mixing then encoding back into whatever is most appropriate for each
end-point.<br>
<br>
Michael <br>
<br>
--<br>
Michael Graves<br>
mgraves<at>mstvp.com<br>
http://blog.mgraves.org<br>
o713-861-4005<br>
c713-201-1262<br>
sip:mgraves@mstvp.onsip.com<br>
skype mjgraves<br>
fwd 54245 <br>
<br>
</span><o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:#000001'>--<br>
Michael Graves<br>
mgraves<at>mstvp.com<br>
http://blog.mgraves.org<br>
o713-861-4005<br>
c713-201-1262<br>
sip:mgraves@mstvp.onsip.com<br>
skype mjgraves<br>
fwd 54245<o:p></o:p></span></p>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Doken, Serhad <br>
<b>Sent:</b> Wednesday, June 17, 2009 4:07 PM<br>
<b>To:</b> 'asterisk-users@lists.digium.com'<br>
<b>Subject:</b> Re: [asterisk-users] Wideband (G722) MeetMe<o:p></o:p></span></p>
<p class=MsoNormal><o:p> </o:p></p>
<p>Hi,<o:p></o:p></p>
<p>I wanted to follow up on this thread about WB support on the MeetMe bridge
that is in 1.6.2. Does it only work for G722 or any WB codec ? <o:p></o:p></p>
<p>I am working with another 16k WB codec that I can transcode to 722 and vice versa.
I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any other WB
codec natively(without downscaling).<o:p></o:p></p>
<p>Thanks,<o:p></o:p></p>
<p>Serhad Doken<o:p></o:p></p>
<p>----------------<span style='font-family:Wingdings'>à</span><o:p></o:p></p>
<p>Razza wrote: <o:p></o:p></p>
<p>If so, i’ve also read that Joshua Colp was/is working on a replacement
(conf_bridge?) that works with G722. If this is this still in active
development are there any planned timelines? If it’s in 1.6.0.6, and
i’ve just missed it or it’s been renamed please be nice in your
responses! :o) <o:p></o:p></p>
<p>It was just merged into Asterisk SVN trunk, on its way to becoming part of
Asterisk 1.6.2. It's new and still has some limitations, but progress is being
made and we welcome additional testing! <o:p></o:p></p>
<p class=MsoNormal>-- Kevin P. Fleming Digium, Inc. | Director of Software
Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype:
kpfleming | jabber: <a href="mailto:kpfl...@digium.com">kpfl...@digium.com</a>
Check us out at <a href="http://www.digium.com">www.digium.com</a> & <a
href="http://www.asterisk.org">www.asterisk.org</a><o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p>
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