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...

How to change voice (disguise) without formant, pitch analysis,etc, using only simple common DSP method as Hilbert transform and...

How to change voice (disguise) without formant, pitch analysis,etc, using only simple common DSP method as Hilbert transform and forth. Unfortunetly I have some mips only. Maybe this problem hasn't decision in these circumstances ?

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, 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)...

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...

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,...

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...

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...

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 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...

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 ...

I have Chirp sub bottom profile traces, these are just a form a seismic trace, basically a time series with amplitudes. Normal seismic...

I have Chirp sub bottom profile traces, these are just a form a seismic trace, basically a time series with amplitudes. Normal seismic traces have positive and negative amplitude values. These traces only have positive values, I belive this is called a instantaneous amplitude trace. How can I process this so it returns to a conventional seismic trace with positive and negative values? From my...

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...

Hi all, I have designed ssb modulator with HT in Matlab Simulink. The unwanted components (lower or higher) has attenuation 35dB. My...

Hi all, I have designed ssb modulator with HT in Matlab Simulink. The unwanted components (lower or higher) has attenuation 35dB. My question is can I get better results with this SSB method. Thank you in advance, Amir

Hello All, I have an application where I know the group delay of a minimum phase system and I need to calculate the matching magnitude response...

Hello All, I have an application where I know the group delay of a minimum phase system and I need to calculate the matching magnitude response of the system. I know how to calculate the magnitude from the phase response via the Hilbert Transform but in this case I'm not able to obtain the phase response. Calculating the phase from the group delay is not an option as errors accumulate during t...

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

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

OK, say I have the signal: signal=(sin(2*%pi*77.7811 .* x) + .39*sin(2*%pi*74.7811 .* x) + .34*sin(2*%pi*81.3811 .* x)); From this I know:...

OK, say I have the signal: signal=(sin(2*%pi*77.7811 .* x) + .39*sin(2*%pi*74.7811 .* x) + .34*sin(2*%pi*81.3811 .* x)); From this I know: the real part of the signal is identical to the signal, and the imaginary part is: Q = (-cos(2*%pi*77.7811 .* x) - .39*cos(2*%pi*74.7811 .* x) - .34*cos(2*%pi*81.3811 .* x)); Since the Hilbert transform of a sin(x) = -cos(x). Using this, I can...

Is it possible forego the joys of the Costas loop and demodulate a scalar QPSK signal using a bandpass filter, a Hilbert transformer and the...

Is it possible forego the joys of the Costas loop and demodulate a scalar QPSK signal using a bandpass filter, a Hilbert transformer and the atan2(im,re) function?

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...