ADC to IP and IP to DAC

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They are not my "unreasonable" prerequisites.....They're theirs.
As for me, I'm always trying to figure out a way to do exactly as told and if it's impossible, I move to a "layer down" by giving my customers another "close to this" solution. :)

Right. In this case, the answer is to accept higher latency (at lest the time for packet to time out and be retransmitted) or accept occasional dropouts.
 
b. By the "occasional dropouts" what exactly do you mean?.......A note of an instrament (ex. saxophone) of theirs perhaps - I can't understant it. :eek:

What happens if the network drops a packet? There are two things you can do. One is to wait for the sender to retransmit the packet (with an adaptive timeout as per RFC793) causing latency, the other is to "fake it" by not producing any sound for the length of the packet. The shorter you make the packet (a 1024 byte packet corresponds to 1 ms at 24 bit 192 KHz stereo), the shorter the possible dropout, but the higher the overhead and the risk of packet loss.

I don't think there is any way to guarantee a 500 ms timeout/retransmit using TCP, so you would have to implement your own acknowledgement/timeout/retransmit protocol on top of UDP.
 
.....yes, I'm a Greek guy and my English are good (I think) but not THAT good!... :D

For example:

" think the OP needs to go back and tell his musician friends that unless they can find a way to avoid causality and the Nyquist sampling theorem......"

I didn't got it..... I can't "fully" understand with these strange words involved.:D
Sorry. :)
 
With a first glance for "AES-X192 / RAVENNA" on google, I didn't found any device as I describe. :(

The standard isn't finalized yet, but it is something to use as a basis for implementing what you want - but you would probably have to do it yourself.

There are no devices currently available, because nobody really needs low latency at that kind of frequency range.
 
actually Julf, looks like Hans Zimmer has it under control (see insane 'home' studio below), but they wrote their own software and have unlimited budget with a coder on staff... as you can see.

they used it not only for control surfaces/midi/audio, but for managing massive hi-rez multitimbral sample based instruments for soundtrack work. (The Dark Knight, A league of their own, Inception etc.)

how they go with latency on live performance i'm not sure, perhaps they work with low res proxies and then sub in the hi-res when they are nearing completion/mixdown. which would be an option for you too hipro5, but does add another layer of complexity on the programming


its starting to sound like a commercial project but you dont wish to admit it...
 

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OK.

Can someone do this for us? :)

but no, nobody can break the laws of physics for you ;) there are some elements here that are simply not possible and you'll forgive me for thinking these 'customers' , their requirements, the project and the preconceptions of 'needs' may be a complete PITA for anyone that took it on.

it would be big money, I dont have the skills, few people do and some desires it wont matter how much money you throw at it.
 
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