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filtering in freq domain

Started by sundar in comp.dsp19 years ago 15 replies

Hi all, To begin with thanks to all those who have made this wonderful group. I am a mechanical engineer working on hilbert transforms. I...

Hi all, To begin with thanks to all those who have made this wonderful group. I am a mechanical engineer working on hilbert transforms. I have a time domain signal sampled at 2100pts/sec for 60secs and has signals at 3 freqs. I do a fft, get into my freq domain and then filter all freq except the freq of interest. My question is a) What is the best way to filter in freq domain. I...


Envelope Detector using Hilbert Transform

Started by w106pjs in comp.dsp19 years ago 7 replies

All I have been kind going through previous threads in this group on similar concern and question that I have. Still I feel my feet is not on...

All I have been kind going through previous threads in this group on similar concern and question that I have. Still I feel my feet is not on the ground yet...with this issue.. Background: I have working with ultrasound signals (300 Khz) from a solid state sensor to a target at 3". The received signals from the sensor (300 Khz) are digitized through a sampling card and imported into MATLAB,...


Slightly OT: Iterated Integration by Parts

Started by Clay in comp.dsp19 years ago 5 replies

Hello All, I had put in the back of my Hilbert paper a little bit about a method called Iterated Integration by Parts. My then Calc prof, Dr....

Hello All, I had put in the back of my Hilbert paper a little bit about a method called Iterated Integration by Parts. My then Calc prof, Dr. Boal, taught this trick to me years ago. I've never seen it in any Calc book - and I have quite a few. I was wondering if any of you have seen this method before and if so, was it just shown to you or was it in a book? Just curious. I'm surprised it ...


Filter latency and TDOA

Started by Manny in comp.dsp16 years ago 1 reply

Hi, I'm working on an application for TOA and TDOA for a set of sensor data. After matched filtering, I'm running my data through a...

Hi, I'm working on an application for TOA and TDOA for a set of sensor data. After matched filtering, I'm running my data through a 39-tap Hilbert Transform to ultimately obtain the signals' envelope and store the in-phase and quadrature components of the peaks for subsequent processing. Due to the filter latency, very close sources are not detectable. So I thought of using circular buf...


What makes quadrature mixing possible?

Started by miladsp in comp.dsp9 years ago 40 replies

so I've been reading quadrature signals, quadrature mixing and hilbert transform and while most of it makes sense I find quadrature mixing a...

so I've been reading quadrature signals, quadrature mixing and hilbert transform and while most of it makes sense I find quadrature mixing a bit confusing. What I can't get my head around is the fact that we can transmit in-phase and quadrature parts of the signal on the same physical channel. What are exploiting here? Am I right in thinking that we're essentially transmitting a complex signal usi...


Minimum phase vice versa minimum phase

Started by Uli Brueggemann in comp.dsp19 years ago 10 replies

Hello, I'm actually confused with minimum phase. I know actually two applications for minimum phase: 1. A FIR filter is designed and the...

Hello, I'm actually confused with minimum phase. I know actually two applications for minimum phase: 1. A FIR filter is designed and the result typically is a time domain signal (or taps) with a symmetric structure = linear phase This filter can be converted to a minimum phase filter e.g. by methods described at dspguru. So for example Hilbert transforms are often used. If I do a FFT t...


AM SSB demodulation

Started by Vikram Chandrasekhar in comp.dsp20 years ago 12 replies

Hello, I am trying to demodulate an AM SSB (single side band) baseband signal. Are there any standard techniques/papers for AM SSB...

Hello, I am trying to demodulate an AM SSB (single side band) baseband signal. Are there any standard techniques/papers for AM SSB baseband demodulation in the presence of arbitrary phase/frequency offsets? To elaborate my query, let s(t)={m(t)+j*H[m(t)]}*e^(j*2*pi*fc*t) where H[ ] denotes the Hilbert transform, fc denotes the carrier frequency and s(t) is the complex analytic signal be ...


Re: Complex version of an impulse

Started by Jerry Avins in comp.dsp20 years ago

Bergers wrote: ... > We have to be careful with the terms used in describing signals. Analytic > signals and complex (I and Q) signals...

Bergers wrote: ... > We have to be careful with the terms used in describing signals. Analytic > signals and complex (I and Q) signals are not necessarily the same. Reference > pages 58-59 of Radar Detection by DiFranco and Rubin. DiFranco and Rubin define > an analytic signal as y(t) = s(t) + jx(t), > where x(t) is the Hilbert transform of the real signal s(t). The Fourier > transfo


Single Sideband modulation using QAM64

Started by ipruzhinin in comp.dsp9 years ago 2 replies

Hi All! It's clear how to build system when some real signal (1st real sequence) goes to input of Gilbert transform giving us the second (imag)...

Hi All! It's clear how to build system when some real signal (1st real sequence) goes to input of Gilbert transform giving us the second (imag) sequences of samples. http://www.dsprelated.com/blogimages/RickLyons/SSB-Lyons.pdf The quiestion is - how to use Hilbert transform in QAM modulator, because modulator block gives us complex signal and then RRC filter processs the signal. Thanks...


How to get envelope from AM signal without phase shift

Started by WWalker in comp.dsp14 years ago 243 replies

Hi, Does any one know how to extract the envelope of an amplitude modulated signal without a phase shift, distortions, and able to determine...

Hi, Does any one know how to extract the envelope of an amplitude modulated signal without a phase shift, distortions, and able to determine the envelope in between the signal cycles. One way that almost works is to simply devide the signal by the carrier but, this technique is too sensitive to phase noise. I have also tried using the Hilbert transform but, I get some leakage distortions. Th...


Analytic Signal Generation in the Frequency Domain

Started by BobM in comp.dsp17 years ago 26 replies

Hi All, I've been trying to generate an analytic signal (Hilbert transformation) in the frequency domain using the method outlined in the...

Hi All, I've been trying to generate an analytic signal (Hilbert transformation) in the frequency domain using the method outlined in the IEEE Marple paper in the Transactions on Signal Processing "Computing the Discrete-Time Analytic Signal via FFT" (9/1999). I'm also using Rick's UDSP 2nd edition book as a reference, which outlines the same method. The problem I'm haivng is that the r...


90 degrees phase shift

Started by Giuseppe Sbarra in comp.dsp19 years ago 24 replies

Hi, I'm an hold analog engineer , actually developing a DSP based application and I need to change of 90 degrees the phase of a signal on the...

Hi, I'm an hold analog engineer , actually developing a DSP based application and I need to change of 90 degrees the phase of a signal on the range of 50 - 250 Hz aving the system a 200uSec sampling rate. I have considered the Hilbert FIR filter but for the moment I cannot get it to work not even reducing the sampling rate. In particular I nedd to phase shift by 90 degrees a signal (voltage)...


EJ: Remez / PM for nonlinear phase

Started by Fred Marshall in comp.dsp20 years ago 5 replies

Eric, I just now focused on your suggested process: "generate the usual Remez input grid, except now create real and imaginary components. ...

Eric, I just now focused on your suggested process: "generate the usual Remez input grid, except now create real and imaginary components. Send the real portion into the Remez routine configured for normal filter design, and send the imaginary component into the Remez routine with the Hilbert bit set (or however your nearest Remez function needs to be set to do this). Take the resulti...


negative frequency and Hilbert transform

Started by fisico30 in comp.dsp15 years ago 3 replies

hello forum, I might need some help understanding the usefulness of the complex analytic signal. What we measure are real valued signals...

hello forum, I might need some help understanding the usefulness of the complex analytic signal. What we measure are real valued signals whose Fourier transform can be one-sided, or, if we used complex sinusoids, two-sided and symmetric. Then the real signal is made of positive and negative frequencies in equal amount. The negative sinusoids dont really a physical meaning, I guess. Comple...


Analytic signal

Started by fisico32 in comp.dsp14 years ago 3 replies

Hello Forum, in AM modulation, a carrier (pure monochromatic signal) is multiplied by the message signal m(t). In angle modulation, the phase...

Hello Forum, in AM modulation, a carrier (pure monochromatic signal) is multiplied by the message signal m(t). In angle modulation, the phase or derivative of the phase is modulated according to m(t). In both case we have a real signal. What is the advantage of associating the analytic signal to the real modulated signal? The analytic signal has its imaginary part equal to the Hilbert tra...


Wavelets in vector space without Cauchy-Schwartz inequality

Started by Edward Jensen in comp.dsp14 years ago 2 replies

Dear all. I am studying wavelets and multi-resolution analysis (MRA) at the moment. One of the key problems with time-frequency analysis in...

Dear all. I am studying wavelets and multi-resolution analysis (MRA) at the moment. One of the key problems with time-frequency analysis in general is the uncertainty principle with states that the localization in time and in frequency has a lower bound (called Heisenberg boxes in signal processing). My book presents MRA in a Hilbert space (L^2([0;1])) and proves the uncertainty relat...


constant 90-degree phase shift using parallel allpass filter networks

Started by Robert Adams in comp.dsp16 years ago 15 replies

One method of creating a quadrature signal is to pass an input signal through two parallel allpass networks, where the outputs of the...

One method of creating a quadrature signal is to pass an input signal through two parallel allpass networks, where the outputs of the two networks differ in phase by 90 degrees. Each allpass network is typically high-order to obtain reasonable accuracy in terms of holding the phase difference over frequency. One advantage of this technique compared with the more usual anti-symmetric Hilbert f...


Non minimum/Maximum phase filter question

Started by westocl in comp.dsp11 years ago 5 replies

From my knowlege of phase/Magnitude response of digital filters, minimum and maximum are the only filters whose Magnitude response...

From my knowlege of phase/Magnitude response of digital filters, minimum and maximum are the only filters whose Magnitude response completely specifies its phase response (some kind of log hilbert relationship.. or whatever). Does this infer that given a system that is non minimum and non maximum phase and given an infinite amount of taps, that one could synthesize any arbitrary frequency res...


Using Z-transform with continous time signals

Started by Communications_engineer in comp.dsp15 years ago 23 replies

Hello, can we use Z-transform on continuous time signals to find their z-domain analysis. (what do we really realize in z-domain and how is it...

Hello, can we use Z-transform on continuous time signals to find their z-domain analysis. (what do we really realize in z-domain and how is it different from frequency domain analysis) and also can we use Laplace for discrete time signals. Also what kind of analysis/information can we get from i) Laplace Transform ii) Z Transform iii) Fourier Transform iv) Hilbert Transform (PS. n...


Hilbert Transformers, Question about delay needed to see 90 deg phase shift

Started by Mark in comp.dsp19 years ago 6 replies

The short definition of a HT is that it provides a +90 deg phase shift for negative frequencies and a -90 deg phase shift for...

The short definition of a HT is that it provides a +90 deg phase shift for negative frequencies and a -90 deg phase shift for positive frequencies. But phase shift relative to what?, relative to the HT input? Apparently not. I used QED to design a HT and looked at the phase response (for positive frequencies) expecting to see about a -90 deg phase shift over the range of frequencies, bu...