60 Hz Hum removal

"Fred Marshall" <fmarshallx@remove_the_x.acm.org> wrote:

>In a noise canceller, the adaptive filter part of the system receives an
>input that attempts to be a reasonable replica of the interference.  The
>input is *not* the same as the "signal" or system "input".  This replica
>doesn't have to be perfect in amplitude, phase or harmonic content.  The
>adaptive filter adjusts the amplitude and phase of the replica so that it
>can be subtracted from the "signal" - thus removing the interference perhaps
>perfectly and not removing any of the desired signal ... unless the desired
>signal has components that are correlated with the interference.

The assumption here is that there is no way to make the noise
canceller adjust to frequency differences?  Say for instance that the
noise will be somewhere in the range of 59 to 61 Hz.  But due to
system contraints, the reference input is not the actual power line
signal, or derivate ... but a mathematical simulation of it at 60 Hz.
The noise cancellor could only cancel out the noise at 60 Hz then, in
that case ... not at 60.2 or 58.2 or anywhere else it drifts, right?

Power line noise will drift around 60 Hz.  Here in the North America,
it can be somewhere on the order of 0.02 Hz, but more in some other
countries.  If someone can't use the power line as a reference input,
is it still possible to set up an adaptive noise cancellor?

Regards,

Robert

www.gldsp.com

<r_obert@REMOVE_THIS.hotmail.com> wrote in message 
news:hqnon0l3tnr19kqcdl51nddkrinlnvekn3@4ax.com...
> "Fred Marshall" <fmarshallx@remove_the_x.acm.org> wrote:
>
>>In a noise canceller, the adaptive filter part of the system receives an
>>input that attempts to be a reasonable replica of the interference.  The
>>input is *not* the same as the "signal" or system "input".  This replica
>>doesn't have to be perfect in amplitude, phase or harmonic content.  The
>>adaptive filter adjusts the amplitude and phase of the replica so that it
>>can be subtracted from the "signal" - thus removing the interference 
>>perhaps
>>perfectly and not removing any of the desired signal ... unless the 
>>desired
>>signal has components that are correlated with the interference.
>
> The assumption here is that there is no way to make the noise
> canceller adjust to frequency differences?

I didn't elaborate much but I didn't mean to imply that.  The assumption I 
made is that the "reasonable replica" might be from sensors that pick up a 
version of the noise - i.e. a linear combination of noises - such that 
amplitude and phase as a function of frequency may not be the same as the 
noise in the signal but a linear relationship exists.

So, if the noise changes frequency then so does the output of the canceller 
accordingly.  I wasn't talking about synthesized noises but rather filtered 
versions of the real noise in real time.  Otherwise, just as an example, 
let's assume that the noise is periodic.  Then we could record one period of 
the noise and then use that as a reference to be subtracted (with suitable 
amplitude and phase changes).  But, this wouldn't be a real time replica and 
not nearly as useful.

Fred 

"Fred Marshall" <fmarshallx@remove_the_x.acm.org> wrote:

>
><r_obert@REMOVE_THIS.hotmail.com> wrote in message 
>> The assumption here is that there is no way to make the noise
>> canceller adjust to frequency differences?
>
>I didn't elaborate much but I didn't mean to imply that.  The assumption I 
>made is that the "reasonable replica" might be from sensors that pick up a 
>version of the noise - i.e. a linear combination of noises - such that 
>amplitude and phase as a function of frequency may not be the same as the 
>noise in the signal but a linear relationship exists.

Yeah, that would be the standard arrangement.

>
>So, if the noise changes frequency then so does the output of the canceller 
>accordingly.  I wasn't talking about synthesized noises but rather filtered 
>versions of the real noise in real time.  Otherwise, just as an example, 
>let's assume that the noise is periodic.  Then we could record one period of 
>the noise and then use that as a reference to be subtracted (with suitable 
>amplitude and phase changes).  But, this wouldn't be a real time replica and 
>not nearly as useful.

Right.

The reason I asked is because of facing a situation where I have the
signal + noise, but don't have access to anything correlated to the
noise, i.e. the power line, etc.  But I know the noise is from power
line sources ... or at least the portion that I'm looking to remove
... and that it is sinusoidal ( with harmonics ), and in the range of
59-61 Hz.  So the initial option appears to be fabricate a 60 Hz
sinusoidal signal as the reference input.  But since it's uncorrelated
to the actual power line noise, it will not drift with it.  And that
likely means that the noise canceller/adaptive filter will only remove
noise at 60 Hz.

I guess the apparently unsolvable dilema is trying to remove power
line noise that drifts in frequency, without having an input reference
signal that is at least somewhat correlated to it.  

Regards,

Robert

www.gldsp.com

<r_obert@REMOVE_THIS.hotmail.com> wrote in message
news:8g5rn0t10ma69ml48do6fs72ltvjphiqir@4ax.com...
> "Fred Marshall" <fmarshallx@remove_the_x.acm.org> wrote:
>
> The reason I asked is because of facing a situation where I have the
> signal + noise, but don't have access to anything correlated to the
> noise, i.e. the power line, etc.  But I know the noise is from power
> line sources ... or at least the portion that I'm looking to remove
> ... and that it is sinusoidal ( with harmonics ), and in the range of
> 59-61 Hz.  So the initial option appears to be fabricate a 60 Hz
> sinusoidal signal as the reference input.  But since it's uncorrelated
> to the actual power line noise, it will not drift with it.  And that
> likely means that the noise canceller/adaptive filter will only remove
> noise at 60 Hz.
>
> I guess the apparently unsolvable dilema is trying to remove power
> line noise that drifts in frequency, without having an input reference
> signal that is at least somewhat correlated to it.

There's always the good ol' notch filter.  But usually there are harmonics
present as well, which complicates the matter...

"Jon Harris" <goldentully@hotmail.com> wrote:

><r_obert@REMOVE_THIS.hotmail.com> wrote in message
>news:8g5rn0t10ma69ml48do6fs72ltvjphiqir@4ax.com...
>> "Fred Marshall" <fmarshallx@remove_the_x.acm.org> wrote:
>>
>> The reason I asked is because of facing a situation where I have the
>> signal + noise, but don't have access to anything correlated to the
>> noise, i.e. the power line, etc.  But I know the noise is from power
>> line sources ... or at least the portion that I'm looking to remove
>> ... and that it is sinusoidal ( with harmonics ), and in the range of
>> 59-61 Hz.  So the initial option appears to be fabricate a 60 Hz
>> sinusoidal signal as the reference input.  But since it's uncorrelated
>> to the actual power line noise, it will not drift with it.  And that
>> likely means that the noise canceller/adaptive filter will only remove
>> noise at 60 Hz.
>>
>> I guess the apparently unsolvable dilema is trying to remove power
>> line noise that drifts in frequency, without having an input reference
>> signal that is at least somewhat correlated to it.
>
>There's always the good ol' notch filter.  But usually there are harmonics
>present as well, which complicates the matter...

Yeah, that's what's being used right now.  But the notch is narrow,
and so the attentuation suffers as the power line noise drifts ...
plus the issue about missed harmonics, as mentioned.

Regards,

Robert

www.gldsp.com

r_obert@REMOVE_THIS.hotmail.com wrote:

>"Fred Marshall" <fmarshallx@remove_the_x.acm.org> wrote:
>
>  
>
>><r_obert@REMOVE_THIS.hotmail.com> wrote in message 
>>    
>>
>>>The assumption here is that there is no way to make the noise
>>>canceller adjust to frequency differences?
>>>      
>>>
>>I didn't elaborate much but I didn't mean to imply that.  The assumption I 
>>made is that the "reasonable replica" might be from sensors that pick up a 
>>version of the noise - i.e. a linear combination of noises - such that 
>>amplitude and phase as a function of frequency may not be the same as the 
>>noise in the signal but a linear relationship exists.
>>    
>>
>
>Yeah, that would be the standard arrangement.
>
>  
>
>>So, if the noise changes frequency then so does the output of the canceller 
>>accordingly.  I wasn't talking about synthesized noises but rather filtered 
>>versions of the real noise in real time.  Otherwise, just as an example, 
>>let's assume that the noise is periodic.  Then we could record one period of 
>>the noise and then use that as a reference to be subtracted (with suitable 
>>amplitude and phase changes).  But, this wouldn't be a real time replica and 
>>not nearly as useful.
>>    
>>
>
>Right.
>
>The reason I asked is because of facing a situation where I have the
>signal + noise, but don't have access to anything correlated to the
>noise, i.e. the power line, etc.  But I know the noise is from power
>line sources ... or at least the portion that I'm looking to remove
>... and that it is sinusoidal ( with harmonics ), and in the range of
>59-61 Hz.  So the initial option appears to be fabricate a 60 Hz
>sinusoidal signal as the reference input.  But since it's uncorrelated
>to the actual power line noise, it will not drift with it.  And that
>likely means that the noise canceller/adaptive filter will only remove
>noise at 60 Hz.
>
>I guess the apparently unsolvable dilema is trying to remove power
>line noise that drifts in frequency, without having an input reference
>signal that is at least somewhat correlated to it.  
>
>Regards,
>
>Robert
>
>www.gldsp.com
>  
>
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.

The amount of decorrelation delay needs to be greater than the period of 
the lowest-frequency signal-component to be passed.


Hope this is of use,
regards,
John

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

<r_obert@REMOVE_THIS.hotmail.com> wrote in message 
news:c0asn0l0rqr2vscq25eatglsvqbm7a7448@4ax.com...
> 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.

I've heard of that being used in a line enhancer (using the output of the 
adaptive stage itself).  If you use it as described above then you have to 
make sure that the filter doesn't adapt too quickly or too slowly - as the 
60Hz component has been described as changing as well.  So, the signal of 
interest would have to be moving faster than the 60Hz component that you 
want to remove.
This would be a special case of interference and signal of interest that may 
not always hold - even from time to time in the same system.

Maybe a silly implementation - but just for illustrative purposes - would be 
a verrrry slowly changing >> non-changing adaptED filter: just set a 
bandpass filter amplitude and phase to cancel the 60Hz interference.  Then, 
as long as the interference doesn't change, it will be cancelled by 
subtraction.  However, if the interference changes, the performance might 
quickly degrade.  And, the adapted filter would act on the signal.... if I 
recall the context of this thread.

Fred 

"Fred Marshall" <fmarshallx@remove_the_x.acm.org> wrote:

>
>I've heard of that being used in a line enhancer (using the output of the 
>adaptive stage itself).  If you use it as described above then you have to 
>make sure that the filter doesn't adapt too quickly or too slowly - as the 
>60Hz component has been described as changing as well.  So, the signal of 
>interest would have to be moving faster than the 60Hz component that you 
>want to remove.

I'll take that at face value, but am not sure it's understood why.

>This would be a special case of interference and signal of interest that may 
>not always hold - even from time to time in the same system.

For power line noise, I believe the amplitude and phase could change
quicker than the signal of interest ... but most likely not in
frequency.

>
>Maybe a silly implementation - but just for illustrative purposes - would be 
>a verrrry slowly changing >> non-changing adaptED filter: just set a 
>bandpass filter amplitude and phase to cancel the 60Hz interference.  Then, 
>as long as the interference doesn't change, it will be cancelled by 
>subtraction.  However, if the interference changes, the performance might 
>quickly degrade.  And, the adapted filter would act on the signal.... if I 
>recall the context of this thread.

Interesting idea.

Thanks,

Robert

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