[Ardour-Users] Automation from analog mix?

Arnold Krille arnold at arnoldarts.de
Sun Dec 19 10:00:13 PST 2010

On Sunday 19 December 2010 10:13:24 Giso Grimm wrote:
> Am 12/18/2010 02:56 PM, schrieb Jörn Nettingsmeier:
> > On 12/18/2010 10:35 AM, David Kastrup wrote:
> >> Arnold Krille <arnold at arnoldarts.de> writes:
> >>> No, the 16 channel converter is already there to record the main
> >>> signal directly after the pre-amps. You need another 16 channels for
> >>> the signal after the fader to get all the information you want for
> >>> automation of eq, gate, compressor and volume.
> >> 
> >> Nope.  Just the stereo mix at the end.  It has all the required info
> >> after decorrelation ("echo compensation techniques") if you take into
> >> account that changes are confined to small bursts of activity.
> > 
> > i don't quite grasp what decorrelation or echo compensation techniques
> > have to do with it.
> It is simply an adaptive filter technique, used in echo compensation (in
> nearly any modern phone or VoIP software), used for feedback
> cancellation in hearing aids or PA systems, used for noise cancellation
> in cars and active noise cancellation headphones for planes. Adaptive
> filters are even used to control heating systems and many more simple
> tasks - they are designed to estimate quasi-linear black box systems.
> They can involve estimation of time-depending complex filters (e.g.,
> feedback cancellation, echo compensation), but they also can be used to
> estimate a single gain only (the more constraints can be applied to the
> resulting filter, the more stable is the estimate, and it can be adapted
> faster and with less pre-conditions to the input signal (e.g., low
> auto-correlation). This is everyday technology. And the stereo mix would
> be sufficient (in combination with the inputs): If you see your mixing
> console as a time dependent matrix operation, X(t) is your input signal
> (with many channels, lets say N), H(t) is your time dependent mixing
> matrix with size Nx2, then your stereo mix Y is Y=H*X, and adaptive
> filtering is nothing more than estimating H. The method provides an
> error estimate. And if it does not have to be real-time, then the
> estimation definitely can be improved, especially to find the initial
> states.

You are over-simplifying things. For echo-cancellation its a time-dependent 
linear system (not even frequency dependent). Anything you want to extract as 
automation from a finished mix (and the raw signals) is non-linear. Automation 
in todays systems is more the just volume and panorama. And even if you are 
just interested in the volume, the non-linearity of eq, gate, compressor and 
other in-line effects can easily disturb your fine linear algorithm...

Have fun,

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