Hilbert transform FIR size
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From: Vladimir Vassilevsky on 7 Aug 2010 00:33 jungledmnc wrote: > Thanks. Could you please point me to some keywords to search for? The keyword is DIY. A +/-45 degree IIR phase shifter is typical numeric optimization problem (minimizing error vector magnitude). For the audio purposes, you will likely need a filter of the order of 6...8. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
From: Tim Wescott on 7 Aug 2010 01:24 On 08/06/2010 05:53 PM, jungledmnc wrote: > Thanks Tim. I was checking about the solution with 2 allpass networks. What > I don't understand is why do I have to use networks? I said "network" when I should have said "filter". > I tried just for > curiosity to use 2 biquad allpasses, found some points where they were > around 90 degrees to each other, but the differences were quite big. Is > that why we have to use multiple sections? Yes. But hopefully you won't need nearly as many as you would for the Hilbert transform. > And how should I compute the coefficients? Good question! I dunno -- or at least I don't know any structured ways. If I needed to do his I'd search around on the web for a bit, then I'd fiddle around with Scilab to find a set of coefficients that really seemed to work. > I read this text: > http://www.katjaas.nl/hilbert/hilbert.html > > There were also "polyphase IIRs" mentioned. I quite don't understand how > they should work. First why is there some 1 sample delay on the second > channel? And again, how could I get the coefficients? There are some raw > numbers, but now explanation how to find them out. I don't know -- I'm not familiar with what the author's trying to say, and my brief perusal of the site didn't really make anything jump out at me. I _can_ say that his "Polyphase IIR" is not the same thing as the usual "polyphase filter" -- so don't get confused if you run across that term and it seems to be a different animal. -- Tim Wescott Wescott Design Services http://www.wescottdesign.com Do you need to implement control loops in software? "Applied Control Theory for Embedded Systems" was written for you. See details at http://www.wescottdesign.com/actfes/actfes.html
From: HardySpicer on 7 Aug 2010 03:38 On Aug 7, 9:33 am, "jungledmnc" <jungledmnc(a)n_o_s_p_a_m.gmail.com> wrote: > Hi, > I want to create a frequency shifter for audio. First I need to get an > analytical signal via a hilbert transformer. I started by checking out how > long the Hilbert FIR would be. Unfortunately I ended with 20ms, which seems > to be related to -3dB at 50Hz (1/0.02). Isn't there another way to do that? > I mean 20ms is a relatively long delay for realtime processing and also 800 > taps would need relatively lots of CPU power. > > Thanks. Hmmm make sure your Hilbert transformer thingy has the right number of turns. Hardy
From: VelociChicken on 7 Aug 2010 07:27 "jungledmnc" <jungledmnc(a)n_o_s_p_a_m.gmail.com> wrote in message news:KYidnQr0LLY8FsHRnZ2dnUVZ_rSdnZ2d(a)giganews.com... > Thanks. Could you please point me to some keywords to search for? Search for: csound hilbert (Not in quotes) VC
From: Rick Lyons on 7 Aug 2010 08:20
On Fri, 06 Aug 2010 16:33:13 -0500, "jungledmnc" <jungledmnc(a)n_o_s_p_a_m.gmail.com> wrote: >Hi, >I want to create a frequency shifter for audio. First I need to get an >analytical signal via a hilbert transformer. I started by checking out how >long the Hilbert FIR would be. Unfortunately I ended with 20ms, which seems >to be related to -3dB at 50Hz (1/0.02). Isn't there another way to do that? >I mean 20ms is a relatively long delay for realtime processing and also 800 >taps would need relatively lots of CPU power. > >Thanks. Hello jungledmnc, I don't know if you have access to IEEE articles, but Clay Turner has an article titled: "An Efficient Analytic Signal Generator" in the "DSP Tips and Tricks" column of the July 2009 issue of the IEEE Signal Processing Magazine. Good Luck, [-Rick-] |