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...

I need to get a smooth as possible envelope of a real-time speech signal. I have tried rectification + filtering and squaring + filtering...

I need to get a smooth as possible envelope of a real-time speech signal. I have tried rectification + filtering and squaring + filtering and also a Hilbert Transformer (to creat the imaginary part of a complex waveform which I can take the magnitude of) but neither of these methods appear to give a smooth envelope. Is there a tried and tested best method? Hardy

hi everyone, million thanks in advance...i would like to know about EMD which is part one of hilbert huang transform. the procedure of EMD is...

hi everyone, million thanks in advance...i would like to know about EMD which is part one of hilbert huang transform. the procedure of EMD is : 1)For a signal X(t), let m1 be the mean of its upper and lower envelopes as determined from a cubic-spline interpolation of local maxima and minima. The locality is determined by an arbitrary parameter; the calculation time and the effectiveness of the ...

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...

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...

Anyone heard of it? Is it the next best thing since sliced bread and the Fourier Transform? I just discovered it and it seems to have some...

Anyone heard of it? Is it the next best thing since sliced bread and the Fourier Transform? I just discovered it and it seems to have some interesting potential... "The HHT technology is a highly efficient, adaptive, and user-friendly set of algorithms capable of analyzing time-varying processes. Designed specifically for nonlinear and nonstationary signals, HHT can be used to analyze ...

Hi, I read a paragraph of "Theory and application of digital signal processing" of L. R. Rabiner on page 70. It says: For v(n), a...

Hi, I read a paragraph of "Theory and application of digital signal processing" of L. R. Rabiner on page 70. It says: For v(n), a Hiltert transformed signal, its Fourier transform V(e^ (jw)) has the property V(e^(jw))=0 pi < w

Hi, I'm implementing a HT with sampling freq 2 Mhz to convert a real signal to complex.. The required passband goes down all the way to...

Hi, I'm implementing a HT with sampling freq 2 Mhz to convert a real signal to complex.. The required passband goes down all the way to 50 hz. I have a couple of questions which hopefully the experts can give me some advice. First of all. to get the required frequency response and to keep the passband ripple as low as possible, I need a large number of taps and this uses a lot of resources...

clear all; Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 seconds. t=0:1/Fs:T; f0 = 50; ...

clear all; Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 seconds. t=0:1/Fs:T; f0 = 50; % Initial frequency is 50hz f1 = 100; % f(t) changes from 50hz to 100Hz in 2 seconds. freq = f0 + (f1-f0)/T*t; y = sin(2*pi*freq.*t); % Chirp signal y(t) Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 sec...

hi I've a doubt whether there is any possibility to implement HHT on hardware.Whether it can be implemented on FPGA.Thanks in advance

hi I've a doubt whether there is any possibility to implement HHT on hardware.Whether it can be implemented on FPGA.Thanks in advance

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...

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...

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...

Hi Group I have a non quadrature signal at 3 MHz. I need to convert it to quadrature (I and Q) at 1.5 MHz to that I can reduce the...

Hi Group I have a non quadrature signal at 3 MHz. I need to convert it to quadrature (I and Q) at 1.5 MHz to that I can reduce the sampleing rate at the next stage. What do you experts think would be the best method in terms of resources (for an fpga) or is there much difference? 1 Hilbert Transformer 2 Multiply by a 1.5 Mhz sine and cosine and then filter to get rid of the 4.5 Mhz sum...

Hi everyone, I have a question. Assuming we have a band-limited signal from 10-20 MHz with a sampling frequency of 100 MHz. Let's say I want...

Hi everyone, I have a question. Assuming we have a band-limited signal from 10-20 MHz with a sampling frequency of 100 MHz. Let's say I want to obtain the envelope of this signal. One method would be to use a Hilbert filter, to obtain the quadrature signal and then do the whole envelope = sqrt(I^2 + Q^2). However, let's assume that I need to use the absolute value instead. What would happe...

Dear Sir/Madam, I have some problems with calculating the instantaneous frequency of a signal which may be so frequent and familiar, but I...

Dear Sir/Madam, I have some problems with calculating the instantaneous frequency of a signal which may be so frequent and familiar, but I couldn't solve them. The signal is a special ground motion acceleration, El Centro. For the purpose of controlling a structure the frequency in real time is needed. Using Hilbert transformation results in two problems. The first one is over shooting and the se...

Hi everybody, this is my first post on DSP related. I have a chirped nonlinear sinoidal signal (between 40 and 140 Mhz) and I am trying to...

Hi everybody, this is my first post on DSP related. I have a chirped nonlinear sinoidal signal (between 40 and 140 Mhz) and I am trying to obtain frequency information as a function of time from it. I tried model signal based signal processing on it but without much success as finding the global minimum proved difficult. Anyways one way literature described solving my problem is using an...

I want to say "a system is to a signal as a function is to a variable". Indeed, I'm pretty sure that when you've gotten fully spun off into...

I want to say "a system is to a signal as a function is to a variable". Indeed, I'm pretty sure that when you've gotten fully spun off into Hilbert spaces and other esoteric stuff (which I can't say that I fully understand), the above statement is pretty close if not entirely true. But before I go and say it publicly in a seminar, I want to pass it by you sharks (uh, folks) and see if ...

Hi all,=20 I am a new member. Could you help me to answer the question that is: the fu= nction "remezord" in Matlab uses what formula to...

Hi all,=20 I am a new member. Could you help me to answer the question that is: the fu= nction "remezord" in Matlab uses what formula to estimate the order of filt= ers and why the function can not be used to estimate the order of Hilber tr= ansformer and FIR differentiator? Thank you so much,

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...