Xref: utzoo rec.audio:7298 sci.electronics:3333
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From: phd@speech1.cs.cmu.edu (Paul Dietz)
Newsgroups: rec.audio,sci.electronics
Subject: Re: Restoration
Keywords: DTW, Cockpit noise...
Message-ID: <2291@pt.cs.cmu.edu>
Date: 16 Jul 88 10:12:39 GMT
References: <4944@husc6.harvard.edu> <2266@pt.cs.cmu.edu> <4493@pasteur.Berkeley.Edu>
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Max Hauser writes:
>In article <2266@pt.cs.cmu.edu>, Paul Dietz wrote:
>| In article <4944@husc6.harvard.edu> ... (Paul Gallagher) writes:
>| >Why isn't it possible to completely restore a recording: for example, to remove
>| >all extraneous noise (hiss, clicks, coughs), even to make a reasonable guess
>| >about information not in the original recording (for example, given a score
>| >and a knowledge of the harmonics of a voice or an instrument, to recreate
>| >something close to the sound of the original performance)?

>| Actually, this sort of thing is commonly done. 

>I'm not sure that I would concur with this.  People frequently suggest 
>strategies similar to Paul Gallagher's, above; but the problem arises in
>translating ideas like "knowledge of harmonics of a voice or an instrument"
>into hard specifics, algorithms that act on the recorded information,
>to do the job.  The obstacles are not in the broad concept but in the
>nitty gritty.
> Unfortunately, classical (Widrow-type LMS-linear)
>"adaptive" filtering, while it is another very useful tool with a lot of 
>applications, really addresses a different class of problems from what 
>Paul Gallagher proposed.

I guess I jumped a head a bit on this on. I was thinking more in terms
of devices like noise cancelation systems used in cockpits where
LMS and RLS type approaches work quite well. However, I could imagine
applying very similar ideas to exactly what Paul Gallagher suggested.
For instance, let's say you had a noisey recording of a single
trumpet that you wanted to clean up. One approach might be to create
a transmission line-like filter (with harmonically related pass areas)
that could be adaptively tuned (both in frequency, and peak/valley
ratio) to obtain maximal response. Seems to me that this could work
pretty well. Of course, trying to do this for more than one
instrument at a time instantly becomes a nightmare... (Essentially,
this is equivalent to the "narrow band" problem you mentioned LMS
was good for...)

>Another basic tool that might have relevance here is dynamic time
>warping (DTW), used routinely in applications like pattern recognition
>where inputs (like speech) are subject to uncontrollable time-scale
>expansions and compressions.  Unlike the classes of algorithms mentioned
>so far, DTW doesn't assume rigid time alignment among the different
>signals being manipulated and compared. 

Sounds interesting! Do you have any references? (Probably one of
those books on multirate DSP that I always ignore...)

Paul H. Dietz                                        ____          ____
Dept. of Electrical and Computer Engineering        / oo \        <_<\\\
Carnegie Mellon University                        /|  \/  |\        \\ \\
--------------------------------------------     | | (  ) | |       | ||\\
"If God had meant for penguins to fly,             -->--<--        / / |\\\  /
he would have given them wings."            _________^__^_________/ / / \\\\-