[Speex-dev] Newbie: Get echo cancellation level

[Keith Kyzivat]

To cut down on jitter due to buggy software resampler code distributed by
Microsoft a number of years back that is used in many sound card drivers, it
is recommended that your input and output sample rates to/from the devices
be at the internal sample rate that the sound card operates at -- typically
48khz or 44.1khz.  In our application (sipXtapi - open source SIP) there was
much improvement in Jitter levels on Windows when we did this.  Once inside
our software (or outgoing from our software) we resampled using speexdsp
resampler, which is much better from a jitter perspective than relying on
the driver's resampler.  I believe we also saw some improvement on Linux,
however I think it was less of an improvement (jitter wasn't as much of a
problem on Linux).
If you're interested in ways of smoothing out jitter, you could take a look
at the sipXtapi dejitter code in our svn repository -- now over at
http://scm.sipfoundry.org/rep/sipX/main

I don't believe there is any dejitter tools in speexdsp, however I could be
wrong...

On Mon, Sep 22, 2008 at 8:55 AM, p_j_r_m <p_j_r_m at yahoo.com> wrote:

> Hi:
>
> I'm using speex to perform echo cancellation in Windows. I'm aware of the
> problem about out of sync clocks in record and play sample rates in usual
> sound cards . In order to have an idea of how good is my echo cancelation
> working I would like to know if there is any #define thing i can pass to
> speex_echo_ctl to get the actual level of echo cancellation. If not, how can
> i extract that value from the library?
> By the way, is any software solution for out of sync problem ? I'm trying
> to keep input and output buffers adjusted by resampling, but it does not
> work well.
>
> Thanks
>
>
> _______________________________________________
> Speex-dev mailing list
> Speex-dev at xiph.org
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>
>


-- 
Keith Kyzivat

SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000
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