An instructional video on minimum vs. linear phase

excellent.

i imagine a lot of folks that do funky things like copy tracks and slide them, get all kinds of funky artifacts that they do no understand.....

Yes, they do. And we all do parallel compression, parallel distortion, parallel whatever. Automatic PDC has made it easy for us to split and recombine tracks, rarely considering the possible ramifications. For example, a plugin that misreports its own latency (a common problem) can cause bizarre issues when used in a parallel signal path, and when that happens you'll have a devil of a time diagnosing the problem.


This got me thinking about the common scenario of a tempo-synced delay. The delays are fed back and recombined with the original signal, usually after passing through a LPF. I wonder now if this isn't why I've always preferred non-synced and gated delays. Or why simulated tape delays and modulated delays are so popular.

mike_mccue


The video suggests that linear phase pre ringing occurs because of look ahead and plug in delay compensation.

That explanation left me with a "wait a minute" moment.

Is that really why we get pre ringing? It seems like if that was the only reason that it would be predictable and that a methodology for negating the pre ringing could be applied.

Is it correct to think that pre ringing only occurs with dsp or does it also occur in analog circuits that are *almost* linear such as the Pultec design?

Is it really caused by look ahead processes?


best regards,
mike

The non-technical reason for why you get pre-ringing is because that's the way a linear phase filter works.
A linear phase filter is a symmetrical FIR filter with identical ringing on both sides of a signal. The delay compensation just shifts the whole thing over in time to get you back where your unfiltered signal started. 


If you try and get rid of the pre-ringing, you will end up with a very different filter. A quick example of what happens to a filter if I get rid of the pre-ringing:

"Analog FIR" is a bit of a misnomer here, as analog FIR's are impractical for audio applications.

But a filter is a filter and the math/theory works the exact same way in analog or digital. 

The advantage of digital here is the limitations come down to the math/theory itself and how much processing power and precision you want to throw at it. But analog also has physics to worry about.

bitflipper
Yes, they do. And we all do parallel compression, parallel distortion, parallel whatever. Automatic PDC has made it easy for us to split and recombine tracks, rarely considering the possible ramifications. For example, a plugin that misreports its own latency (a common problem) can cause bizarre issues when used in a parallel signal path, and when that happens you'll have a devil of a time diagnosing the problem.

Unless when you are using a parallel signal path/sends you are aware of possible downsides of using Automatic Processing Delay Compensation beforehand and cater for them accordingly of course.
 
I find an ADC disable switch along with a sample accurate line delay a great comfort and help.
 
Here's a good one at the right price.
 
http://www.voxengo.com/product/sounddelay/
 
and an even more comprehensive audio alignment suite again at the right price.
  
http://varietyofsound.wordpress.com/2011/05/02/prefix-final-teaser-and-release-info/