<div dir="ltr">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).<div>
<br></div><div>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 <a href="http://scm.sipfoundry.org/rep/sipX/main">http://scm.sipfoundry.org/rep/sipX/main</a></div>
<div><br></div><div>I don't believe there is any dejitter tools in speexdsp, however I could be wrong...</div><div><div><div><div><div><br><div class="gmail_quote">On Mon, Sep 22, 2008 at 8:55 AM, p_j_r_m <span dir="ltr"><<a href="mailto:p_j_r_m@yahoo.com">p_j_r_m@yahoo.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><table cellspacing="0" cellpadding="0" border="0"><tbody><tr><td valign="top" style="font:inherit">Hi:<br><br>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?<br>
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.<br><br>Thanks<br></td></tr></tbody></table><br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Keith Kyzivat<br><br>SIPez LLC.<br>SIP VoIP, IM and Presence Consulting<br><a href="http://www.SIPez.com">http://www.SIPez.com</a><br>tel: +1 (617) 273-4000<br>
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