R/E/P > Dan Lavry

Summing

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ruffrecords:
dobster wrote on Wed, 18 October 2006 03:29
i've been seeing some arguments that sending tracks directly to a/d outputs from within a DAW sounds/sums better than sending tracks to a bus or master bus and then to the a/d output.
isn't it all 1's and 0's? any validity to this?

Some. Going direct means essentially the bits are sent directly to the D/A. Going via the bus means the bits get multiplied by the fader value. Assuming this value is 1 then it depends on the DAW as to what happens. A smart DAW might realise x1 means do nothing and it would do nothing. Other DAWs might just do the x1 multiplication. Depending on the DAW this might or might not give exactly the same output as the direct connection. It depends on how math is handled in the DAW and whether or not it has any rounding errors. This means some samples may not come out at exactly the same value as they went in.

Ian

dobster:
hey Ian...thanks

so, how does one quantify whether or not a DAW possesses there rounding errors or not? and if that bus fader is at "unity" then is there much of a difference then going directly to the D/A directly?

Jon Hodgson:
dobster wrote on Mon, 23 October 2006 02:56
hey Ian...thanks

so, how does one quantify whether or not a DAW possesses there rounding errors or not? and if that bus fader is at "unity" then is there much of a difference then going directly to the D/A directly?


Any DAW will have rounding errors

HOWEVER, in setting the level and summing (as opposed to say, EQ) the process is so simple (one multiply and one add per channel) that the error is only going to be half a bit maximum. You are not going to hear a half bit error in a 24 bit system.

danlavry:
dobster wrote on Thu, 19 October 2006 21:16
Dan thanks for the response

I'd like to know then, how would you feel about the idea of recording the output of a digital track in the DAW with some type of non-linear processing plugin in realtime?
thanks


Aliasing occurs the instance one tries to include energy (signal) containing frequencies above half the sample rate (at a specific point in the circuit).
To avoid aliasing when converting with an AD, you must make sure that there is no energy there at frequencies above Nyquist (half the sample rate of the converter input circuitry - called the modulator). You can do it by pre filtering (analog filter) the high frequencies, by operating the modulator at higher rate (oversampling) to a high enough rate.... In practice you get to do both (filter and fast operating modulator).

You can do all sorts of processing in analog, and there will be no aliasing. Non linear process will generate high frequency, which when converting to analog can be dealt with as stated above by:
1. Pre filtering
2. Faster rate of conversion (oversampled modulator).

But when you are in the digital domain, as soon as you did a non linear operation, the aliasing is already there, and whatever falls on the audible range can not be removed, it is too late, because the non linear process and the aliasing happen simultaneously. You can help it some by doing the process at high oversampling rate, but whatever does alias to the audio, stays there. You can not filter it before the fact, you can not do it after the fact.

So you can do non linear processing in analog. If you want it in digital, make sure that the level of aliasing is acceptable.

Regards
Dan Lavry
http://www.lavryengineering.com

dobster:
thanks Jon and Dan...

jon, just to clarify, what if you bus @ unity then? meaning, no level adjustment at the bus?

dan, is it possible to create a plugin that upsamples momentarily locally on a track to process and then downsample again? or is that just a dumb idea?

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