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 ...

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, 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...

I need to perform detection and estimation on a signal which contains amplitude-modulated pulses. The model for the signal is y[n] = e[n]...

I need to perform detection and estimation on a signal which contains amplitude-modulated pulses. The model for the signal is y[n] = e[n] + sum(Ek[n-Tk]) where e[n] is noise, Ek[] is a pulse (product of hamming-esque window and a sinusoid), and Tk the corresponding lag. The problem is that the envelope gain, carrier frequency, and event duration are all random variables. I can visu...

Hi Guys Hope you guys can help. I'm just getting into DSP (indirectly really). I'm trying to create a quadraphonic SQ decoder in software....

Hi Guys Hope you guys can help. I'm just getting into DSP (indirectly really). I'm trying to create a quadraphonic SQ decoder in software. To do this I need to shift the enitre audio range through +90 or -90 degrees. I belive after a lot of reading and an immense amount of confusion I need a hilbert transform implemeted via a FIR filter. What I've got is is a PCM (.wav) file with loads...

Let's assume there is a certain periodic sequence m (t). Further from it calculated s(t)=[m(t)+j*hilbert(m(t))]*exp(j*2*pi*f*t). Then it is...

Let's assume there is a certain periodic sequence m (t). Further from it calculated s(t)=[m(t)+j*hilbert(m(t))]*exp(j*2*pi*f*t). Then it is calculated it ACF r(m)=(1/(Ncorr-m))*sum(?? n=0 ?? Ncorr-1-m){conj(Vj)*Vj(n+m)}. Further in the module of this ACF the maximal peak is searched, we shall assume with number Mmax. Let's designate frequency of sampling as Fs. If I correctly und

Hello, Using " highpass | rectify | lowpass | downsample" for ages to demodulate signald, I've just found that demodulation using...

Hello, Using " highpass | rectify | lowpass | downsample" for ages to demodulate signald, I've just found that demodulation using Hilberttransformation seems to be very easy (speaking matlab/octave): function y=demodulate( x ) y=sqrt(x.^2 + real(hilbert(x)).^2); endfunction (I implemented "hilbet" myself, using fft) That's the "how", easy ;) But I do not understand the "why". I u...

My understanding is that you phase shift one signal by 90 degrees with a hilbert tranformer than sample - is that right? With a carrier based...

My understanding is that you phase shift one signal by 90 degrees with a hilbert tranformer than sample - is that right? With a carrier based system you need only use sin and cos and then sample giving I and Q. The advantage appears to be that you can sample at B (bandwidth) rather than 2BHz. Can we extend this and phase shift by pi/4 and sample with 4 ADCs? In general we would get a sam...

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

Could some one please help me in the following matter? Given a set of tap-coefficients, how can it be found that the filter is a Hilbert...

Could some one please help me in the following matter? Given a set of tap-coefficients, how can it be found that the filter is a Hilbert transformer (HT) or not? On a website (http://www-users.cs.york.ac.uk/~fisher/cgi-bin/mkfscript), the phase response of a designed HT is plotted and looks like magnitude response of a half band low pass filter. However when I design a HT in matlab u...

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

I've seen how to do the opposite (get phase response from a bandlimited magnitude response). Can the opposite be done? If so, what is...

I've seen how to do the opposite (get phase response from a bandlimited magnitude response). Can the opposite be done? If so, what is the algorithm? thanks

Hi all, I am doing a simple demodulation of a sampled RF signal. I set my digitizer at a fixed 1 GS/s. I don't know what my carrier will be,...

Hi all, I am doing a simple demodulation of a sampled RF signal. I set my digitizer at a fixed 1 GS/s. I don't know what my carrier will be, but it will be less that 200MHz. The signal is pulse amplitude modulated, and I want to recover the original signal. To do this, I have applied a Hilbert transform, and then derived the magnitude which gives me a good approximation of the encoded signa...

I'm trying to generate sets of all-pass FIR filter coefficients to apply varying degrees of phase shift (say from 0 to pi/2) to filter...

I'm trying to generate sets of all-pass FIR filter coefficients to apply varying degrees of phase shift (say from 0 to pi/2) to filter input signals. Can anyone suggest a good way to go about calculating these? Will it be similar to the derivation of a Hilbert Transform? Thanks very much, Jim Singleton This message was sent using the Comp.DSP web interface on www.DSPRelated.com

Hi, Frank, reading thru the new posts this morning you asked me to "elaborate" on Hilbert transform applications. I posted a...

Hi, Frank, reading thru the new posts this morning you asked me to "elaborate" on Hilbert transform applications. I posted a moderately long message in reply to your request. After posting my message, I read three other recent posts of yours asking people to "elaborate" on various aspects of DSP. Now I'm suspicious. Frank, what are your intentions? Why are you asking ...

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 all, I am again looking at the AD9786 datasheet: (http://www.analog.com/UploadedFiles/Data_Sheets/34595181746736AD9786_prc.pd f) and in...

Hi all, I am again looking at the AD9786 datasheet: (http://www.analog.com/UploadedFiles/Data_Sheets/34595181746736AD9786_prc.pd f) and in particular on Fig.13, trying to understand how the chip works and whether I can generate a SSB signal with it (supposedly I should be able to). I wrote a simple MATLAB script (see below) to simulate the image rejection algorithm implemented in the AD97...

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...

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 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...