Xref: utzoo rec.audio:7298 sci.electronics:3333 Path: utzoo!attcan!uunet!lll-winken!lll-lcc!ames!ncar!gatech!udel!rochester!pt.cs.cmu.edu!speech1.cs.cmu.edu!phd 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> Sender: netnews@pt.cs.cmu.edu Followup-To: rec.audio Organization: Carnegie-Mellon University, CS/RI Lines: 51 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." _________^__^_________/ / / \\\\-