Hi! I am still trying to clean the voice after procesing it with speex echo cancellation filter. I get some artifacts, the echo is removed, but you can here some metallic voices in background. Jean, in your test client you are using UDP connection and SpeexJitter to make sure UDP packets are in order. What if I use TCP/IP, do I still have to use SpeexJitter when using speex_echo_cancel ? regards hs
> I am still trying to clean the voice after procesing it with speex echo > cancellation filter. > I get some artifacts, the echo is removed, but you can here some metallic > voices in background.That's a known problem with the method I'm using. It's not easy to improve (it's still an active research topic).> Jean, in your test client you are using UDP connection and SpeexJitter to > make sure UDP packets are in order. What if I use TCP/IP, do I still have > to use SpeexJitter when using speex_echo_cancel ?Yes and no. You don't need the jitter buffer to compensate for packet losses or out-of-order. However, you'll have to compensate for the fact that both ends probably don't sample at the same time (the jitter buffer could do it for you). That being said, TCP for VoIP is a **BAD** idea. Jean-Marc
On Tue, 25 Oct 2005 14:05:29 +0200, Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote:>> I am still trying to clean the voice after procesing it with speex echo >> cancellation filter. >> I get some artifacts, the echo is removed, but you can here some >> metallic >> voices in background. > > That's a known problem with the method I'm using. It's not easy to > improve (it's still an active research topic).is there DTD implemented in speex_echo ?> >> Jean, in your test client you are using UDP connection and SpeexJitter >> to >> make sure UDP packets are in order. What if I use TCP/IP, do I still >> have >> to use SpeexJitter when using speex_echo_cancel ? > > Yes and no. You don't need the jitter buffer to compensate for packet > losses or out-of-order. However, you'll have to compensate for the fact > that both ends probably don't sample at the same time (the jitter buffer > could do it for you). That being said, TCP for VoIP is a **BAD** idea.sorry, I am very new in this topic and still learing :).