Adaptive Notch Filter - LMS Algorithm

Started by zrimkunas in comp.dsp12 years ago 4 replies

Hi everyone, I am working on a simple adaptive notch filter that will be used to cancel one sinusoid. Additionally, the filter coefficient is...

Hi everyone, 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 close to the unit circle at the same angle as the zero. The transfer function is: H(z) = (1 - (1


Notch filter design using poles and zeroes(2nd order) Newbie question

Started by terrp in comp.dsp13 years ago 15 replies

Hi all, 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...

Hi all, 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 spectra using SPTOOL MATLAB, nothing seems to be removed from the spectra. I tried playing with the v...


Frequency domain notch filter

Started by skaggio in comp.dsp11 years ago 24 replies

Hello everybody. A question from not an expert of notch filters. Suppose I have to filter out a specific frequency noise (e.g. a 50...

Hello everybody. 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 out. I would like to compute the Fourier transform of the signal, put the 50Hz frequency amplitude to...


notch filter to remove 20Hz noise of ECG signal

Started by ashcroft2006 in comp.dsp14 years ago 8 replies

Hey 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...

Hey 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 me. im removing noise from an ecg signal and i need to remove this noise at 20hz using a notch filter, sampling rate is 200hz. the matlab code is: > > > a=[x x x] > > > b=[x x x] > > > zfiltered=filter(a,d,z) 'z' is my noise corrupted signal, but ho


IIR phase compensator

Started by itsh11 in comp.dsp11 years ago 3 replies

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...

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 design? Many thanks


IIR filter on a fixed set of data points

Started by ahgu in comp.dsp9 years ago 9 replies

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,...

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 remove the harmonics on the initial 250 samples. What can I do to filter the whole data set? thank you, ...


60 Hz Hum removal

Started by Rob Hutchinson in comp.dsp15 years ago 62 replies

What is the preferred method for removing 60 hz hum from a signal without wiping out signal info around 60 hz? A 60 hz notch filter would not...

What is the preferred method for removing 60 hz hum from a signal without wiping out signal info around 60 hz? A 60 hz notch filter would not be useful because it would attenuate the signal as well. I'm interested in doing this for sampled data, so all filtering would be done in the digital domain. Thank you in advance, Rob


Filtering out 60Hz + harmonics

Started by Tic Tac in comp.dsp6 years ago 21 replies

Hello all, I am quite new to the world of DSP and am required to perform some DSP for my student job using Matlab. I need to remove the noise...

Hello all, I am quite new to the world of DSP and am required to perform some DSP for my student job using Matlab. I need to remove the noise ofthe utility frequencies (60 Hz + harmonics) from a number of signals. I have been reading many forums on this topic and tried many possible solutions but was unable to get it to work properly. For instance, I tried using a notch filter and also tried...


How to write a NotchFilter procedure

Started by gpdsoft in comp.dsp13 years ago 21 replies

Hi, I have do write a procedure in Delphi that take an array of samples and filters it. The filter I've to do is Notch to cut out the 50Hz...

Hi, I have do write a procedure in Delphi that take an array of samples and filters it. The filter I've to do is Notch to cut out the 50Hz from my signal. My signal is sampled at 50Khz for a max of 250 millisec. Do you know where I can find the code or something similar to quickly implement this function in my software? Thnx a lot


Help with THD+N calculations

Started by rcoup in comp.dsp13 years ago 6 replies

Hi All, My DSP knowledge is far too rusty and I've been hitting my head against the wall for a little while now. Archives of the group have...

Hi All, My DSP knowledge is far too rusty and I've been hitting my head against the wall for a little while now. Archives of the group have got tantalisingly close but I'm still confused. The fun situation is calculating THD+N of an audio signal. What I'm doing: a. 1KHz sine wave is my fundamental. b. Notch Filter to remove the 1KHz c. FFT (8Kpoint) d. Get magnitudes (mag[] = sqrt(re[...


Any shape IIR Filter

Started by sparafucile17 in comp.dsp12 years ago 27 replies

Let's say I have a particular freqeuncy response that I can draw out on paper and I want an IIR filter to do just that. What is the best...

Let's say I have a particular freqeuncy response that I can draw out on paper and I want an IIR filter to do just that. What is the best approach to this problem. I know there are simpler ways to do this using FIR filters but let's say my constraint was to only use the IIR form. For example the frequency shape I need is a smiley face or a parabola. It's not a peak-notch type or a band-reject...


How to design this complex filter?

Started by canmc in comp.dsp15 years ago 27 replies

Hi all. I'm in trouble on designing this linear time-invariant digital filter. It has five peaks and two notches in its magnitude...

Hi all. I'm in trouble on designing this linear time-invariant digital filter. It has five peaks and two notches in its magnitude frequency response. Each peak/notch is specified by its frequency, 3dB-bandwidth and magnitude. These parameters and of course, the sampling frequency, can be chosen arbitrarily. Please give me any solution. Thanks in advance. Best regards, Canmc


Frequency notch filter with 5 possible coefficients

Started by DSP_student in comp.dsp11 years ago 18 replies

Hello everyone, I would like to know whether it's possible to find a sequence of FIR coefficients of even length such that each coefficient can...

Hello everyone, 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 packed with all sorts of low-correlation sequences, and some of them (like legendre sequences) possess...


Power line hum question

Started by DJT in comp.dsp11 years ago 17 replies

Hello everybody! I'm trying to remove power line hum from a signal and have two questions concerning this. 1. How high do the harmonics go?...

Hello everybody! I'm trying to remove power line hum from a signal and have two questions concerning this. 1. How high do the harmonics go? (In general, round numbers) 2. How much frequency variations (+/-) should I expect in the fundamental 50 Hz component? The reason for asking these questions is that I am having a hard time choosing between a fixed comb-notch filter and someth...


Adaptive solution +power line harmonics

Started by DJT in comp.dsp11 years ago 3 replies

Hello sorry to bother you again but I am confused. What I understand from the threads I read is that a comb notch filter is a good choice to...

Hello sorry to bother you again but I am confused. What I understand from the threads I read is that a comb notch filter is a good choice to remove power line interference, assuming that the frequency is stable... But say that the frequency fluctuates significantly, some say that there is no use to go adaptive, while some claim that it should work... I don't see why the suggestion of m...


Halfing biquad filter response

Started by jungledmnc in comp.dsp11 years ago 20 replies

Hi there, I need to half the impact of the classical biquad filter from the r-b-j's cookbook, so that when I use the filter twice, the response...

Hi there, I need to half the impact of the classical biquad filter from the r-b-j's cookbook, so that when I use the filter twice, the response would be the same as the original filter. For peak and shelf filters it seems that halfing the dB factor is enough. At least the difference seems to be very small, almost inaudible. But LP, HP, BP or notch... I tried to manipulate Q, but no value se...


books on adaptive filtering

Started by Jack Ace in comp.dsp14 years ago 5 replies

hi everybody. I'm looking for one (or more) textbook on adaptive filter theory. I'm not interesting in advanced topics, I'm interesting in a...

hi everybody. 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 studied the subject on a such valuable book? e.g. I implemented some adaptive notch filters startin...


"Comb notch" filters missing a tooth

Started by Rune Allnor in comp.dsp16 years ago 6 replies

Hi all. 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...

Hi all. 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 woodpecker. Now, the owl makes a very distinctive sound that shows up qiute well in a spectrogram (...


Line noise removal

Started by acat in comp.dsp9 years ago 13 replies

Hi, I need some help on removing the line noise (50Hz) from my data. I know the method conventionally adopted is to fit a sinusoidal function...

Hi, I need some help on removing the line noise (50Hz) from my data. I know the method conventionally adopted is to fit a sinusoidal function and subtract from the original data --- the so called notch filter, I have two more questions with this: 1. In my data, not only 50Hz, but the frequencies nearest to it also have abnormal values, this may be caused by the leakage of the power in 50 Hz (I ...


Quantization of fixed point filter coefficients.

Started by Anonymous in comp.dsp14 years ago 1 reply

This post is a retake on the 'quantization of filter coefficients' issue. Let me present the issue in the following. I have designed a...

This post is a retake on the 'quantization of filter coefficients' issue. Let me present the issue in the following. I have designed a filter (notch, cut-off 55.7Hz) which is a bi-quad. The transfer function is : H(z) = 1 + (-1.992)*z^-1 + (0.994)*z^-2 -------------------------------- 1 - (-1.836)*z^-1 - (0.852)*z^-2 Now I am implementing this filter on a 16-bit fix...