I'm receiving a signal at 100kHz and need to do two things at the same
-notch the 2k frequency.
-rough detection of the presence of 2k frequency.
So basically the notch filter is always there, but I also want to know if
something is being notched.
I could obviously have in para...
I am trying to obtain a FIR notch filter which is very narrow. For
example, a 50 Hz notch filter to remove mains noise from a 1250Hz signal
Doing this with an IIR filter was easy, I could specify my "notch region"
as small as [49 51]Hz.
I am now trying to do the same with a FIR fi...
I am searching a mathematical solution for a notch filter.
I measured a signal with a ADC and I want to kill one frequency (50Hz)
out of the measurement data.
Does anyone of you know a mathematical function for a notch filter ?
Say I generate a digital 2nd order notch filter. With the lattice
structure, one coefficient controls the notch frequency and the other the
bandwidth. I can get VERY good attenuation of the notch frequency, pretty
much only depending on wordlength used. The overall response looks like one
I'm trying to find an even lengthed sequence of FIR coefficients where
every coefficient will be one of the five: 1, -1, 0, j, -j.
I'm looking for a sequence whose FFT will have one notch (or at least a
20dB attenuation relative to other frequency samples) at a certain
frequency. I'm hop...
> Im a student and need help using matlab. im extremeley new to signal
> processing so this is a pretty basic question, im hoping someone can help
> im removing noise from an ecg signal and i need to remove this noise at
> 20hz using a notch filter, ...
I've found in MATLAB help that a notching comb filter has following
H(z) = b*(1+z^(-N))/(1+a*z^(-N))
where N is the number of notches. How to determine coefficients a and b
with respect to real-world parameters such as sampling frequency, notch
frequency and notch b...
I am trying to design a second order digital IIR band stop (notch) filter
with the following specs:
3dB cut off frequencies: 55Hz and 65Hz
I want the notch at 60Hz with atleast 90dB attenuation at the 60Hz.
Sampling frequency: 200hz
I tried various filter configurations like a Butterworth or...
Hope no introduction for why I am here (I have a problem.....)
Presently i am working on a ofdm simulink implementation where I need to
increase the single carrier notch depth (notch is created by putting some
carriers to zero)which at present is 13dB(because of 'no window' or ...
I'm having problem getting rid of a 2kHz sine wave from a digitized signal
at 128kHz. I do know the exact frequency of the 2kHz (I generate it in the
first place with a DDS), but the phase/amplitude are unknown (it goes
through DAC-> ADC.
Now, there seems to be two approach to do this, and ...
A question from not an expert of notch filters.
Suppose I have to filter out a specific frequency noise (e.g. a 50 Hz
noise) from an acquired signal. Suppose my filter has not to work online,
i.e. I can collect signal samples (its a digitalized signal) and then
filter noise o...
> Thanks for your comments. I appreciate it.
I was involved in an IRIG-B demod about 20 years ago. As I recall, the
first thing we did was to remove DC, then square the signal, and run
that through a 2KHz notch filter. Since IRIG-B is primarily a 1KHz
I need to implememt a notch filter to filter 2175 Hz tone on TI C54x DSP.
Does anyone know if there is source code in TI's application report? or can
you point me to any source code as example?
Thanks in advance!
I was given a matlab project to use a unit circle with 2poles and
2zeroes to create a notch filter. The component to be removed is 50 Hz,
sampling frequency is 200 Hz.
I tried putting 2 zeroes at z=1, angle= pie/2;
and 2 poles at z=0.9, angle = pie/2. However, when i look at the s...
Jerry Avins wrote in news:-bmdnb3C7cUcofPZRVn-uQ@rcn.net:
> Al Clark wrote:
> > Tim Wescott wrote in
> > news:8KSdnUsGjuGof_DZnZ2dnUVZ_t6dnZ2d@web-ster.com:
> > > Andor wrote:
> > >
> > >
> > > > Folks,
> > > >
> > > > is it possible to construct a causal a...
On Jun 5, 1:13=A0am, "briwel" wrote:
> I have an FIR filter equation, which is
> y(n) =3D 0.5x(n) + 0.5x(n - 2).
> I have to draw the magnitude of the frequency response of this filter?
> Does anyone know how to do this?
> thanks in advance for any help
This is an easy one si...
I have an IIR notch filter ( i.e. bandstop ), that rings like crazy
whenever a transient signal comes through ... which is very unwanted
for my application. Anybody have any ideas how to get around this? I
have to stick with IIR form filters. I've tried all the various
I have the advantage of seeing the other posters' suggestions and
your repsonse, but how about the following:
y[n] = x[n] - 0.61803 * x[n-1] + x[n-2]
? This is an FIR notch filter with a notch right at 20 Hz.
The 0.61803 term is 2*cos(0.4*p...
I would like to know whether it's possible to find a sequence of FIR
coefficients of even length such that each coefficient can be 1 out of 5
possibilities: 1, -1, 0, j, -j. I'm trying to create a Band-Stop filter
using a sequence of coefficients like this.
The literature is pa...
I'm looking for one (or more) textbook on adaptive filter theory.
I'm not interesting in advanced topics, I'm interesting in a well
explained book able to drive the beginner inside the matter. a book
with some examples or solved exercises would be appreciated.
does anyone st...
I m making a simple PID controller with notch filters in Xilinx
blocks. it always prompt the following errors for the blocks in my
notch filter. if i delete the error block or change the latency to 1,
the same error will happen to another block in the notch.
Is there a general guidel...
On Nov 28, 9:06=A0pm, "acat" wrote:
> Hi, I need some help on removing the line noise (50Hz) from my data. I kn=
> the method conventionally adopted is to fit a sinusoidal function and
> subtract from the original data
No, it isn't.
> --- the so called notch filter, I have two
I want to design an IIR notch filter with linear phase. It is pretty
straight forward to design the IIR notch filter but I am not sure how to
design the phase compensator(all pass IIR filter) that makes the overall
phase of the system linear phase. Can someone throw a few pointers on this
I have a fixed 3600 data points(samples) and I want to apply a notch 2nd
order Butterworth IIR Notch filter to remove the 1/4Fs frequency, the
results start to converge(or filter start to work) after about 250 or so
samples. The issue is the it is a continuous stream of data and I need to
I am working on a simple adaptive notch filter that will be used to cancel
one sinusoid. Additionally, the filter coefficient is complex (I am
looking after the error envelope). So what I have is one zero and one
pole. The zero is fixed to the unit circle and the pole is very clos...
> tim w wrote:
> > Hello all,
> > I need some guidance in programming a laplace transfer function into
> > computer language -- pseudocode for now.
> > The transfer function is a second order function:
> > To^2*s^2 + zeta1*To*s + 1
> > ----------------------...
On 15 Apr 2005 03:46:05 -0700, email@example.com wrote:
> > However, is there a simple method of finding the fewest number of
> > partials required to create a waveform that passes through each of the
> > points?
> > I could "solve" this using search-based techniques such as genetic
Andreas Huennebeck wrote:
> Fred Marshall wrote:
> > One method might be to use the sample at 1002Hz for the reference and
> > forget about a notch filter, just don't include that one sample or the rms
> > value of that one sample and the two adjacent ones in the sum across
> > fre...
I have just "made" some recordings of birds nearby (which is to say that
I ripped them off a CD released by the local ornithology society, but don't
tell anyone) and want to play with these sounds to make an audio DSP demo.
Among the birds are the "Eagle Owl" (Bubo Bubo) and a woo...
This question gets asked all the time and I have read many of the answers
and tried a number of things but I am unable to get satisfactory results.
The issue is that I am recording an audio signal and processing it,
however, when I use a different mic, I get a different background spectrum.
if you knew the exact frequency of your noise, you would remove it via an
adaptive notch filter in which the input signal would be the noise and the
reference signal would be the audio signal with noise. Then the adaptive
filter would remove the noise from your audio signal.
So now my excel spreadsheet implementing R B-J's equations work fine.
R B-J assisted me to build a "Master Volume Control" for the 5 band
equaliser. It adds/subtracts some dB from the reponse of each filter based
on the setting of the Master Volume control, by changing the coefficients (
In article , "Rune Allnor" wrote:
> Praveen wrote:
> > Hi,
> > I have a speech signal that has noise introduced at various
> > frequencies. The FFt magnitude of the speech signal is of the order of
> > 10 power 6 at these frequencies. The whole point is I have to remove
> The difference is that the first radio can handle one and only one
> signal over an astounding span, but it may not be able to handle two
> signals at all. The second radio can be plunked down in a busy spectral
> environment and do OK -- 90dB of dynamic range is a bi...
> My understanding is that if sampled at 2GHz a 1.1GHz signal would
> appear to be a frequency of 900MHz after sampled and an FFT was
> taken. A 1.9GHz signal would appear as a 100MHz signal if you took an
> FFT. If you sampled a signal with two tones wouldn't you have to flip
> the FFT f...
glen herrmannsfeldt wrote:
> Jerry Avins wrote:
> > You can have a continuous function, but on a CRT, plasma display, or
> > stepping-motor plotter, you can't have a continuous plot. The moiré that
> > troubles you is between the actual points calculated alo...