## Slightly OT: Iterated Integration by Parts

Started by in comp.dsp15 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 ...

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

Started by in comp.dsp15 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...

## Evelope detection using Hilbert transformer and carrier ripple

Started by in comp.dsp15 years ago 17 replies

I have a question about envelope detection. I am an "analog" guy trying to understand the advantage of using the Hilbert transformer (HT)...

I have a question about envelope detection. I am an "analog" guy trying to understand the advantage of using the Hilbert transformer (HT) method for obtaining the envelope of a carrier signal. I have read that the standard DSP approach is to use a Hilbert transformer (HT) to create a 90 deg shifted version of the carrier then: envelope = sqrt (I^2+Q^2) Does the output of th...

## FIR Hilbert transformers

Started by in comp.dsp14 years ago 23 replies

Hi all.. 1. Is it possible to design a FIR hilbert transformer for the follwing specifications.. Signal freq = 220khz Samples freq =...

Hi all.. 1. Is it possible to design a FIR hilbert transformer for the follwing specifications.. Signal freq = 220khz Samples freq = 12Mhz The purpose here is to generate the envelope over the signal freq. of interest (220khz) The fear I have, as the signal frequency is close to DC with this unusally high sample rate, the length of the designed FIR hilbert transformer may be very high. ...

## AM signal detection with Hilbert?

Started by in comp.dsp14 years ago

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...

Started by in comp.dsp14 years ago 1 reply

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...

## Analytic Signal Generation in the Frequency Domain

Started by in comp.dsp13 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...

## Compacting of arbitrarily separated bands: Hilbert transformer

Started by in comp.dsp12 years ago

I am doing a project in which I have to implement filters to do multiple band-pass filtering and then compact the arbitrarily separated bands....

I am doing a project in which I have to implement filters to do multiple band-pass filtering and then compact the arbitrarily separated bands. I believe using Hilbert transformers after the filtering would be a good idea to shift the bands to their required locations. However, this has to be done in real time, and hence the algorithm has to be fast( I think this rules out the optimal method?). I a...

## Help to understand the formula

Started by in comp.dsp12 years ago

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

## PING Rick Lyons re Hilbert Xform & Envelope Detectors

Started by in comp.dsp12 years ago 3 replies

Hi Rick, I sent you an e-mail about a week ago re the above. Did you get it? Mark

Hi Rick, I sent you an e-mail about a week ago re the above. Did you get it? Mark

## Instantaneous frequency with Hilbert transforms!!

Started by in comp.dsp11 years ago 5 replies

Sorry for post again if posted before!! About the Instantaneous frequency with Hilbert transforms. I still haven't found a clear answer....

Sorry for post again if posted before!! About the Instantaneous frequency with Hilbert transforms. I still haven't found a clear answer. This is the code I have so far in matlab: [x,fs] = wavread('xx.wav'); %a real speech signal; xx = hilbert(x); %analytic signal xx; n = length(xx); pha = angle(xx)/2/pi; ff = diff(pha); %ff is wh...

## Window selection for Hilbert Transformer

Started by in comp.dsp11 years ago 5 replies

I am generating FIR Hilbert transformers with the following code: function b = hlbt(n) if (rem(n, 2) == 0) ...

I am generating FIR Hilbert transformers with the following code: function b = hlbt(n) if (rem(n, 2) == 0) usage("n must be odd"); endif n = n + 1; i = [-n/2+1:n/2-1]; b = 2/n * (sin(pi*i/2).^2).*cot(pi*i/n); b(n/2) = 0; b(abs(i)

## Envelope of Speech

Started by in comp.dsp11 years ago 5 replies

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

## causal and minimum phase system

Started by in comp.dsp11 years ago 3 replies

I am a little confused with causal and minimum phase system. If a system is a minimum phase system, then the phase of system could be decided...

I am a little confused with causal and minimum phase system. If a system is a minimum phase system, then the phase of system could be decided by phase = hilbert(log(abs(f))). So the real(f) = abs(f)*cos(phase) and imag(f) = abs(f)*sin(phase) On the other hand, a minimum phase system is a causal system, so that imag(f) = hilbert(real(f)). So, here there are two real/imag number. Which on...

## Hilbert Transformer questions..

Started by in comp.dsp10 years ago 16 replies

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...

## chirp linearity using Hilbert Transforms

Started by in comp.dsp10 years ago 1 reply

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...

## Wavelets in vector space without Cauchy-Schwartz inequality

Started by in comp.dsp10 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...

## Analytic signal

Started by in comp.dsp10 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...

## Hilbert Transformer Designed Using the Frequency-Response Masking Technique

Started by in comp.dsp10 years ago 8 replies

Hello! I have read article "Optimum Masking Levels and Coefficient Sparseness for Hilbert Transformers and Half-Band Filters Designed Using...

Hello! I have read article "Optimum Masking Levels and Coefficient Sparseness for Hilbert Transformers and Half-Band Filters Designed Using the Frequency-Response Masking Technique", Yong Ching Lim, NOVEMBER 2005. There're in example two filters Hb(z) and H1(z). I guess they derived by multiplication classical hilbert impulse function h(n)=[1-cos(Pi*n)]/Pi*n on different windows. What kind o...

## instantaneous frequency

Started by in comp.dsp9 years ago 3 replies

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...