"Correcting" output of a filter
in [DSP]
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From: Jerry Avins on 26 Apr 2010 10:25 On 4/26/2010 9:37 AM, Greg Berchin wrote: ... > Ah, the perils of designing in the frequency domain. It's so much easier to get > that frequency response "just right" when you don't have to worry about the time > domain. > > Consider that frequency response is a steady-state concept. That is to say, > when you specify the response of a filter at a certain frequency, you tacitly > assume that a signal of constant amplitude at that frequency was applied an > infinitely long time ago, and will continue for an infinite time to come. > > But we live in the time domain. Signals start, stop, and change. Those starts, > stops, and changes trigger time-domain responses from filters. And the > time-domain responses can be very unattractive when the filters are designed in > the frequency domain. It is a fundamental give-and-take; to optimize something > in the frequency domain almost always means to "un-optimize" something else in > the time domain, and vice-versa. > > So the answer to your question on how to "correct" the filter is to design the > filter for better time domain response in the first place. And, of course, that > means giving up something in the frequency domain. What a wonderfully concise way to emphasize a simple truth! Those paragraphs should be in the preface to every writing on filter design. Jerry -- "I view the progress of science as ... the slow erosion of the tendency to dichotomize." --Barbara Smuts, U. Mich. �����������������������������������������������������������������������
From: Vladimir Vassilevsky on 26 Apr 2010 10:30 Ross wrote: > "Correcting" is not going to be the right word, but here's my problem. Could you tell what is the real problem. I.e. what are you trying to accomplish as the final result. > I would like to have a brick wall filter with no phase delay and > moderate computational complexity. (Yes, I know, who wouldn't). There is no free cheese. A linear filter with a bandwidth B will have a delay no less then about 1/B. > Are there any techniques or research on how to "correct" early output > from a filter after there has been a sudden change from (relative) > silence to signal? Yes, there are. If you have a particular model of the signal, and a particular goal to accomplish, you can make a nonlinear adaptive filter. > I will not be using the output of the filter in any > audio that is actually listenened to. If this is about audio, why can't you delay the rest of the signal by the same amount as the delay in the filter? > I will only be tracking the zero > crossings and peaks of the waveform. What is the goal of this? Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
From: Rune Allnor on 26 Apr 2010 10:40 On 26 apr, 15:36, Ross <rossclem...(a)gmail.com> wrote: > On Apr 26, 12:30 pm, Rune Allnor <all...(a)tele.ntnu.no> wrote: > > > On 26 apr, 12:20, Ross <rossclem...(a)gmail.com> wrote: > > > > "Correcting" is not going to be the right word, but here's my problem. > > > > I would like to have a brick wall filter with no phase delay and > > > moderate computational complexity. (Yes, I know, who wouldn't). > > > ...so you don't know much 'bout filters, do you? > > Can you please explain what you mean by this? To establish a baseline from which to start. Somebody who knows the basics of filter design would never mention the ideal filter in a design setting. > I know that there are no > brick wall filters and I believe my message made it clear that I > wasn't expecting to be able to completely repair the signal ....and right there you revealed your ignorance again. There is nothing to repair, as the effect you see is impossible to avoid. The best one can do is control it. > from my > significantly not "brick wall" filter to make it a higher performing > *all purpose* filter. There is no such thing as an 'all purpose' filter. > My comment about wanting a brick wall filter > without significant delay nor excessive computational requirements was > a tongue in check comment intended to be read as "ideally what I'd > like is this, but since I know I won't get it, ....." Yet again you reveal a total ignorance about the subject. .... > Is this clearer? What is clear is that you have no idea what a filter is, how it works, or how to design one. Why not describe what you attempt to achieve with your filter. Rune
From: Vladimir Vassilevsky on 26 Apr 2010 11:08 Rune Allnor wrote: > Why not describe what you attempt to achieve > with your filter. Hm, it is interesting to guess. The OP is obviously not a pro although he tries to accomplish some kind of sub audio signaling; so it should be something very common. My first guess would be amateurish EEG or ECG. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
From: jim on 26 Apr 2010 11:11
Ross wrote: > Are there any techniques or research on how to "correct" early output > from a filter after there has been a sudden change from (relative) > silence to signal? I will not be using the output of the filter in any > audio that is actually listenened to. I will only be tracking the zero > crossings and peaks of the waveform. However, I need to have an > estimate early on of the peak value of the waveform passing through. > Tricky given that the filter takes some time to "get going" (AKA the > filter isn't a brick wall filter). Sounds like not using a filter would instantly solve all the problems. Is there a reason why you haven't tried that? -jim |