Sorry, I assumed someone trying to do the impossible (or nearly so) will have explored what makes it impossible - or would do so when it was drawn to his attention. Especially if someone is paying. FWIW Nyquist is not an English name!hipro5 said:
The nice thing about having a simple view of the world is that all problems (including impossible ones) look simple.
You still haven't told us what the application is, and how your friends found fans with ultrasonic hearing.
Maybe with my English I didn't explained it well.
a. They want to pass frequencies (Audio) from 0 to 20KHz FLAT.
b. They have devices (transmiter/reveiver - upconverter/downconverter) that you give them an input of 0 to 20KHz and the "transmiter" with its local oscilator (45KHz), upconverts them at 45KHz + 0 to 20KHz - thus MAX 65KHz (Bandwidth from 45KHz to 65KHz).
The receiver gets this - from 45KHz to 65KHz bandwidth signal - and with its local oscilator (45KHz), downconverts it to AGAIN 0 to 20KHz.
By doing so, you pass audio frequencies from somewere to somewere else, without hearing them in the midle (of the transmission).
So by doing so, you could pass through an audio signal and ANOTHER audio siglal at the same time.
Can you understand it now somehow?
That is the reason that they want something with FLAT bandwidth from 0 - 65KHz.
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very strange, surely the use of another channel for that extra embedded 20khz would be easier... how does that even work? how do you demodulate 20hz from 45.020Hz doesnt seem to make sense, but i'm no expert here and I cant say i've though about frequncy like that before, so maybe it just seems odd
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If I knew what they actualy what these guys to do, I could explain it better.
Problem is that my "brother" (best friend of mine) is involved with this and I'm trying to help him as I can.
I've already asked him why don't they just pass 0 to 20KHz from one AUDIO channel and the other 0 to 20KHz from another AUDIO channel and they will have the same resault.
He said that they don't want that. They want as they describe.
Problem is that my "brother" (best friend of mine) is involved with this and I'm trying to help him as I can.
I've already asked him why don't they just pass 0 to 20KHz from one AUDIO channel and the other 0 to 20KHz from another AUDIO channel and they will have the same resault.
He said that they don't want that. They want as they describe.
Why upconvert? It achieves nothing for a digital transmission, except to make the problem harder by requiring a faster sampling rate. If you want to send several audio streams over the same channel then there are digital techniques for that. Two streams are routine - we call it stereo.
Something your friends need to learn is that when you need a solution you first have to define the problem, not insist on a half-baked solution and call that the problem.
Something your friends need to learn is that when you need a solution you first have to define the problem, not insist on a half-baked solution and call that the problem.
192k "audio" ADC is no problem, should be flat to ~80 kHz
popular Delta-Sigma types do add a few dozens to maybe ~100 sample times latency in their digital filters
but the big variable is the network - are you talking on a company or Uni intranet? - could be very low latency, high "QoS"
or do you mean any 2 network connections? - say one provider's DSL with another's cable service at the other end - provided both even have enough bandwith to your machines the packets may go 100s of miles before "connecting"
no one wants to publish QoS limits in the general case - you need to be specific about the connections
bidirectional delays are critical for collaborative performance - 10-20 ms seems to be recommended as maximum delays for the musicians to play together
popular Delta-Sigma types do add a few dozens to maybe ~100 sample times latency in their digital filters
but the big variable is the network - are you talking on a company or Uni intranet? - could be very low latency, high "QoS"
or do you mean any 2 network connections? - say one provider's DSL with another's cable service at the other end - provided both even have enough bandwith to your machines the packets may go 100s of miles before "connecting"
no one wants to publish QoS limits in the general case - you need to be specific about the connections
bidirectional delays are critical for collaborative performance - 10-20 ms seems to be recommended as maximum delays for the musicians to play together
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but that doesnt take you any closer to the solution lol we all knew thats what you meant for the demands, we just didnt know why ... I still dont, very wierd way to go about it. there are more solutions for lower bandwidth multichannel networks for audio that high res.
so they actually want to play together live? good luck with that.
so they actually want to play together live? good luck with that.
Right. They want to play live, we got that. What we don't get is the need for the silly HF modulation stuff - it sounds like they are trying to do an old-fashioned analog transmission system on top of a digital system, and that is just a bad idea.
Please, go back to your friend/brother/whatever, and ask what the musicians *really* want to do (the HF modulation stuff is part of the *how*, not the *what*), and we might be able to come up with a solution that is workable.
Please, go back to your friend/brother/whatever, and ask what the musicians *really* want to do (the HF modulation stuff is part of the *how*, not the *what*), and we might be able to come up with a solution that is workable.
Some news.
As much as I could understand out of it.
They want to be three Groups.
a. Group A and Group B will communicate with the 0 to 20KHz bandwidth (Pure audio).
b. Group A and Group C will communicate with the 45KHz to 65KHz bandwidth (via external upcoverter/downconverter boxes).
When Group A and Group B play together, Group B, will not able to hear what Group C plays and Group C ALSO will not be able to hear what Group B plays.
Group A will be the "center composer" which will hear both Group B and Group C.
So - I guess - There will be 2 musical compossitions at the SAME TIME with a center music of the Group A.
What can I say????....
As much as I could understand out of it.
They want to be three Groups.
a. Group A and Group B will communicate with the 0 to 20KHz bandwidth (Pure audio).
b. Group A and Group C will communicate with the 45KHz to 65KHz bandwidth (via external upcoverter/downconverter boxes).
When Group A and Group B play together, Group B, will not able to hear what Group C plays and Group C ALSO will not be able to hear what Group B plays.
Group A will be the "center composer" which will hear both Group B and Group C.
So - I guess - There will be 2 musical compossitions at the SAME TIME with a center music of the Group A.
What can I say????....
Unfortunately you need to go back to them and tell them that the 45 to 65 KHz part is not part of a requirement specification, but one suggested (but bad) attempt at a solution.
For their 3 groups they need:
- one (isolated) audio channel from group A to group B
- one (isolated) audio channel from group A to group C
- one (isolated) audio channel from group B to group A
- one (isolated) audio channel from group B to group A
Right? No need for any silly carrier frequencies.
Are they playing acoustic or electric instruments? Will they be using headphones or monitor speakers? If using acoustic instruments and/or voice, and monitor speakers, the real challenge will be preventing bleed from the monitor speakers of group A to the microphones leading to groups B and C.
The only reason would be to go long distances over the Internet, but then the latency and packet loss issues take on a completely different dimension...
Oppppsssssssss......I found something simillar for our needs in NET.
THOUGH a LOT of $$$$$ involved. 
FMC01 - FM MPX Codec*:: 2wcom - Audio & IP Sat Systems, FM/RDS Solutions, DVB-T Monitoring
http://www.2wcom.com/fileadmin/redaktion/dokumente/Produkte/FMC01_FM_MPX_Codec_datasheet.pdf
Problem is that we want AT LEAST 65KHz.
How hard is it to be?
We don't want ALL these "funcy" things on.
THOUGH a LOT of $$$$$ involved.
FMC01 - FM MPX Codec*:: 2wcom - Audio & IP Sat Systems, FM/RDS Solutions, DVB-T Monitoring
http://www.2wcom.com/fileadmin/redaktion/dokumente/Produkte/FMC01_FM_MPX_Codec_datasheet.pdf
Problem is that we want AT LEAST 65KHz.
How hard is it to be?
We don't want ALL these "funcy" things on.
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