VOICE over UDP need some help here

[RandallSP]

I'd love to know more about how its been done before.  VoIP related stuff is cool 😄

On the other hand, Skype is really really good.  The voice quality is telephone quality or better with minimal bandwidth usage.

Take care,  and keep me posted on your results!

--Randall

[dirtyb15]

I could not get skype to work on my private work network,(not connected to internet).  Is there something i missed in the setup? I did not see anything for a local network.

On a side note, though i did get my vi working and it actually sounds pretty good.  Right now i have it running on 4 computers and i am not seeing any problems with running our disnet flight simulation software so i dont think it is hogging to much bandwidth.  It took me a while to figure out how to mix the audio signals, but i finally got around it by just having a seperate read loop for each ip address then just using labviews SO write vi.  So i have 4 SO writes going on and if 4 signals are being received, all are played simultaneusly.

Now that i have this working though i would like to try and optimize it.  Any ideas of doing squelch control or something so that if the mic level is not very high, no data packets are transmitted?  I also thought of trying some data compression but need to do a little more research. Thanks again for the help!!

[RandallSP]

Ahh, I think Skype requires an internet connection to function.  Teamspeak doesn't.

I presume for squelch control you could periodically monitor the intensity of the sound coming from the mic and only transmit if it goes over a threshold.  Can you do that?

--R  

[dirtyb15]

That is pretty much what i want to do, but i cant figure out how to measure the intesity. The labview mic read returns an 8 or 16 bit number.  I dont know what calculation to do on that number to get an intensitry or something i can work with.  Any ideas?

 I am really happy with this so far though, i am surpised how well it works.

[RandallSP]

Can you graph that number over time?  I imagine it is a waveform showing the sound the mic is picking up. 

So assuming you can get a sample of this waveform (you may need an input buffer), if you look at the signal's peak-to-peak intensity that should be a good indicator as your intensity.  Or maybe you could look at the signal in the frequency domain (take the fourier transform of the waveform--I think you need more then the base labview package to do this) -- that would give you a very clear transmit/don't transmit indicator (just measure the peak value!).

Then you can use a slider control on the front panel for your threshold to adjust it to your liking.

Message Edited by RandallSP on 07-26-2005 12:54 PM

[dirtyb15]

Thanks! Ill try the transform, i have the extra package for that among others to play with. Ill let you know how it goes,

Dave

[dirtyb15]

I ended up not going into the frequecy domain and just looked at max or min values.  I have pretty good control now with a slider to not send useless udp packets only containing background noise.  This cut down alot on network activity alone.  Now i am going to look into compression and see if i can reduce it even more.