John Monro <johnmonro@delete.optusnet.com.au> wrote:>Robert, >The only reference I have on this is: > Ifeachor & Jervis, Digital Signal Processing, A Practical >Approach,Addison Wesley, 1993, > p.544, chap. 9.2.2, "Other configurations of the adaptive >filter." . >The following page gives the topology of a number of different adaptive >filter applications, including the one I described. Although chapter 9 >very good in its general coverage of adaptive filters, unfortunately >there are no design notes given for these particular variations,Great, I'll see if I can't hunt that down.> >Some time ago there appeared on the market a "signal enhancer" for Ham >Radio use, which may be of interest to you.. I don't recall its actual >name or maker, but it appeared to be based on two adaptive filters, in >different configurations. One filter acted as discussed, to attenuate >continuous tones in the received signal, exploiting the fact that this >interference is more coherent than the speech. The part I particularly >liked was that the other filter attenuated the wide-band interference >by exploiting the fact that the 'hiss and crackle' interference is LESS >coherent than the speech. Ifeachoe & Jervis show this configuration as >well.Interesting. Is encouraging that there are proven practical applications. Sometimes there are great ideas proposed in papers or textbooks, but that run into snags in actual application. And then the question is sometimes not whether it can be made to work practically, but how much effort is required!> >Your intended aproach seems the right way to go.Great, that's just how I'll approach it.> >Good luck, >regards,Thanks, same. Robert www.gldsp.com
60 Hz Hum removal
Started by ●July 1, 2004
Reply by ●October 28, 20042004-10-28
Reply by ●October 29, 20042004-10-29
"John Monro" <johnmonro@delete.optusnet.com.au> wrote in message news:<417f0405$0$32541$afc38c87@news.optusnet.com.au>...> r_obert@REMOVE_THIS.hotmail.com wrote:>> >John Monro <johnmonro@delete.optusnet.com.au> wrote:> >> >> >> >>r_obert@REMOVE_THIS.hotmail.com wrote:> >>> >>Robert,> >>> >>One technique is to use a delayed version of the signal itself as> >>the> >>reference input to the adaptive filter. This delay, often called a> >>'decorrelation delay' ensures that the adaptive filter does not try to> >>null out the changing part of the signal, but is able to null out the> >>long-term ('coherent') part of the signal, i.e. the 60Hz component with> >>its associated harmonics.> >>> >>> >> >That sounds promising.> >> >> >> >>The amount of decorrelation delay needs to be greater than the> >>period of> >>the lowest-frequency signal-component to be passed.> >>> >>> >> >Ah. So if the lowest frequency of the signal to-be-passed is 1 Hz,> >the delay of the reference signal to the actual signal has to be at> >least 1 second.> >> >> >> >>Hope this is of use,> >>> >>> >> >Yes, it has thanks. If you can think of a reference or paper to> >describe this in more detail, that'd be great. But I'll just> >initially approach it as a normal LMS or RLS adaptive filter> >arrangement, but with the reference input simply delayed, per your> >suggestion and the 'lowest-frequency signal-component' criteria.> >> >> >> >>regards,> >>John> >>> >>> >> >Same,> >> >Robert> >> >www.gldsp.com> >> >> Robert,> The only reference I have on this is:> Ifeachor & Jervis, Digital Signal Processing, A Practical> Approach,Addison Wesley, 1993,> p.544, chap. 9.2.2, "Other configurations of the adaptive> filter." .> The following page gives the topology of a number of different adaptive> filter applications, including the one I described. Although chapter 9> very good in its general coverage of adaptive filters, unfortunately> there are no design notes given for these particular variations,>> Some time ago there appeared on the market a "signal enhancer" for Ham> Radio use, which may be of interest to you.. I don't recall its> actual name or maker, but it appeared to be based on two adaptive> filters, in different configurations. One filter acted as discussed,> to attenuate continuous tones in the received signal, exploiting the> fact that this interference is more coherent than the speech. The> part I particularly liked was that the other filter attenuated the> wide-band interference by exploiting the fact that the 'hiss and> crackle' interference is LESS coherent than the speech. Ifeachoe &> Jervis show this configuration as well.>> Your intended aproach seems the right way to go.>> Good luck,> regards,> John>The person responsible for the signal enhancer you are refering to was Al Clark. He now runs Danville Signal Processing which is where the recent comp.dsp conference was held. dsp@danvillesignal.com I don't think Al makes/sells these anymore. Yes an adaptive line enhancer is a good choice for adaptively separating the correlated component of a signal from the uncorrelated. There are many papers especially in the ieee journals on this technolgoy. Most Adaptive filter text books cover this as well. One draw back on the ALE is the bandwidth of the enhanced signal. The length of the adaptive filter determines the bandwith of the ALE. I have generated what I call a recursive ale to really narrow the bandwith of the ALE. I also noticed a paper at an IEEE conference that refered to his recursive ALE as a feedback ALE or FALE. I'm not kidding. If you search on FALE in the ieee conference papers it shows up. An ALE, RALE or FALE will detect all periodic signals (within reason). The drawback of the RALE of FALE is the reponse time to readapting to changing periodic signals. In your case this doesn't seem to be a problem. The frequency changes very slowly. You are in trouble if there are periodic signals or signals with a longer correlation than your decorellation delay that you want to keep. Feeding an adative filter with the correlatred output of the ALE will work great. Maybe you already spoke of this next method. I don't know I just tuned in. You can Bandpass filter the incoming signal at 60 Hz(or what ever the fundamental frequency is). Then determine the phase component by zero crossing detection (yuk) or using a digital PLL or adative PLL to determine the real fundamental. This can be used to generate a sin and cos of you fundamental. Then generate the harmonics by multiplying the phase comp and generating more sine and cos waves. This gives you separate signals for each frequency. These sin cos pairs can be used to adapt a quadrature filter (only 1 tap per sin and 1 tap per cos) using the original input as the error. It may not be more efficient than the ALE method but it gives you more control over which tones are removed because you can have a separate mu (covergence factor) for each. Please note that when using a signal subtraction method like this a small convergence factor should be used on the filter determining the amplitude and phase of the signal to be subtracted. You can use a large convergence factor but it will result in a non-linear filter which can actually change the frequency of the reference signal if your reference doesn't exactly match the error. Maybe that's not so bad. Mark Allie John and Robert I appologize for sending this response to you directly and not using just the newsgroup.>>
Reply by ●October 30, 20042004-10-30
<allie@engr.wisc.edu> wrote:>The person responsible for the signal enhancer you are refering to was Al >Clark. He now runs Danville Signal Processing which is where the recent >comp.dsp conference was held. dsp@danvillesignal.com I don't think Al >makes/sells these anymore. > >Yes an adaptive line enhancer is a good choice for adaptively separating the >correlated component of a signal from the uncorrelated. There are many >papers especially in the ieee journals on this technolgoy. Most Adaptive >filter text books cover this as well. One draw back on the ALE is the >bandwidth of the enhanced signal. The length of the adaptive filter >determines the bandwith of the ALE. I have generated what I call a recursive >ale to really narrow the bandwith of the ALE. I also noticed a paper at an >IEEE conference that refered to his recursive ALE as a feedback ALE or FALE. >I'm not kidding. If you search on FALE in the ieee conference papers it >shows up. An ALE, RALE or FALE will detect all periodic signals (within >reason). The drawback of the RALE of FALE is the reponse time to readapting >to changing periodic signals. In your case this doesn't seem to be a >problem. The frequency changes very slowly. You are in trouble if there are >periodic signals or signals with a longer correlation than your >decorellation delay that you want to keep. Feeding an adative filter with >the correlatred output of the ALE will work great. Maybe you already spoke >of this next method. I don't know I just tuned in. You can Bandpass filter >the incoming signal at 60 Hz(or what ever the fundamental frequency is). >Then determine the phase component by zero crossing detection (yuk) or using >a digital PLL or adative PLL to determine the real fundamental. This can be >used to generate a sin and cos of you fundamental. Then generate the >harmonics by multiplying the phase comp and generating more sine and cos >waves. This gives you separate signals for each frequency. These sin cos >pairs can be used to adapt a quadrature filter (only 1 tap per sin and 1 tap >per cos) using the original input as the error. It may not be more efficient >than the ALE method but it gives you more control over which tones are >removed because you can have a separate mu (covergence factor) for each. >Please note that when using a signal subtraction method like this a small >convergence factor should be used on the filter determining the amplitude >and phase of the signal to be subtracted. You can use a large convergence >factor but it will result in a non-linear filter which can actually change >the frequency of the reference signal if your reference doesn't exactly >match the error. Maybe that's not so bad. > >Mark AllieMark, Thanks for the additional information. Robert
