Re: [Linphone-developers] G722

[Simon Morlat]

Hi Simon,

Interesting work !
Thank you very much !
I integrated the oRTP fix.
Are you still developing into your repository ? I could add it to
git.linphone.org with other plugins, what do you think ?
Also, if nobody has objections to it (Vadim and you), I would like his
license to be changed to LGPL. Indeed it relies on LGPL source code
taken from spandsp. Agreed ?

Simon


Le mardi 20 juillet 2010 à 17:19 +0200, Simon Brenner a écrit :
> Hi Simon & everyone,
> 
> I found this post on a mailing list archive and I wanted to ask you whether 
> you've tried out my
> plugin code. It was once your suggestion to realize a G.722 en-/decoder as a 
> plugin and that is what
> I've tried to do.
> 
> I would be glad about probably some final hints to get the plugin running. 
> There were quite some
> requests related to G.722 in Linphone.
> 
> Here's the repository:
> 
> https://bitbucket.org/herrpi/msg722
> 
> Greetings!
> 
> -Simon.
> 
> 
> > Hello,
> >
> > First of all thanks for G722 and G726 contributions. Actually instead of
> > integrating them directly in mediastreamer2, I would like to make them
> > plugins, such as msilbc and msx264.
> >
> > I've applied your portaudio fix.
> >
> > Concerning the ugly IETF mistake concerning the rate of G722, my preferrence
> > would be to define the PayloadType with a 8000 Hz clockrate, but put a
> > limited hack within audiostream.c so that the soundcards are open at 16000 
> > Hz
> > instead of 8000 as written in the payloadtype.
> > Or do the reverse thing (perhaps it's better):
> > declare the PayloadType as 16000 Hz but msrtp will "read" 8000 in case
> > mime_type=="g722" .
> >
> > I think it should work.
> >
> > The reason why I dislike the rtp_rate addon is that it modifies the ABI the
> > PayloadType struct and brings confustion just to workaround an IETF mistake
> > for a single payload type. Not sure it is worth to do that.
> >
> > What do you think ?
> >
> > Simon
> >
> > Le Wednesday 28 January 2009 19:30:56 Vadim Lebedev, vous avez écrit :
> >> Hello,
> >>
> >> We've a following interop problem with G722 codec:
> >>
> >> Because of historical reasons the relevant RTP RFC   speicifies that
> >> when using G722 payload
> >> RTP TIMESTAMP should be incremented with 8KHZ frequency  even if the
> >> REAL sampling rate
> >> is 16KHZ.
> >>
> >> As you understand msrtp.c filter is unable to handle this situation,
> >> so we've been thinking about possible enchancements.
> >>
> >> One idea that comes to mind is to add a 'rtp_rate'  field to Payloadtype
> >> structure , and if it is non zero and different from sampling rate to
> >> compute adjustement (divider or multiplier) to rtp time stamp.
> >>
> >> Any comments on this approach?
> >>
> >>
> >> Thanks
> >> Vadim
> 
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