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Re: Applying hilbert transform close to DC

Vladimir Vassilevsky wrote: > The correct way to do 90 shift by band shifting is: > > 1. multiply the signal by exp(iwt) > 2. filter the upper (or lower) sidebands from I and from Q. This is > where the filters come in and where the delay will be incurred. > 3. multiply the signal by...

Re: Low freq "analog" of Nyquist? ( possibly naive question )

"Fred Marshall" wrote in message news: ... [snip] > Very nice treatment of a lot of good stuff Thanks. > However, in the context of the discussion, we were talking about arbitrary > time-limited signals and not systems - if that distinction matters. So, > even though the exponenti...

Hilbert transform & analytic signals

Hi Guys, I've been modeling (with MATLAB) the Hilbert transform for use in generating the analytic signal (a complex signal) corresponding to a real signal x(n). That is, I'm computing a complex signal whose real part is x(n) and whose imaginary part is the Hilbert transform of x(n)....

Hilbert IIR filter implementation

Hello, I wanted to know how to implement Hilbert transform using IIR filter. Any reference or article or suggestion will be great. I wanted to implement it on a DSP processor. Hardware structure, filter coefficient????? waiting for reply With regards praveen ...

Re: Hilbert transform in MATLAB

R.Lyons@_BOGUS_ieee.org (Rick Lyons) writes: > On Fri, 10 Feb 2006 15:58:34 -0600, "Leah" > wrote: > > > I'm doing some real time work in MATLAB as well. I discovered that you > > cannot use the Hilbert transform for real time data because the Hilbert > > function is a non-causal filte...

Understanding modern TxDAC's?

Hi all, Let's say I want to synthesize a 40 MHz wide transmit band centered at 70 MHz, from a baseband sampled at 100 MHz. I played with an AD9772 DAC and found that I can do it in the Direct IF mode if I use the first image (Fs-Fin), but the problem is that the second image (Fs+Fin) is too cl...

Re: What kind of filter is this?

It's a pair of 16-tap, linear phase FIR filters. The code is optimized to reduce the number of multiplications to 1 for each pair of equal magnitude coefficients. The y filter is even and the z filter odd. Given that, what you said about your test, and because there are two of them, the filter...

All-pass filter specified by phase shift amount?

In my experience 1st and 2nd order all-pass filters are usually specified by their frequency, i.e. the frequency at which the phase shift is half of the maximum. For example, see r b-j's audio cookbook (http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt) for a 2nd order di...

Re: Phase response and causality

Hi Rune, I'm sure it will be obvious as soon as I say it -- By duality, a filter is causal if and only if its frequency response is analytic, i.e., the real and imaginary parts of the frequency response are Hilbert transform pairs. I don't know at the moment whether there is a nice way to ex...

Re: Oppenheim & Schafer

On 12 Nov 2003 13:58:47 -0800, allnor@tele.ntnu.no (Rune Allnor) wrote: > > Does anyone know why the two different editions are available? > > And what differnces there may be betwen the two? According to one > > customer review at amazon there seems to be quite substantial > > differences ...

Re: How to implement a filter with arbitrary amplitude curve and phase curve?

In article ktYub.6631$J%S1.3172@twister01.bloor.is.net.cable.rogers.com, jason han at jhanb208@rogers.com wrote on 11/20/2003 00:40: > I am currently working on a project which needs to design a custom filter > with arbitrary amplitude curve and arbitrary phase curve. okay, first thing you...

Re: FIR

In article br2nhr$287lql$1@ID-82263.news.uni-berlin.de, Bhaskar Thiagarajan at bhaskart@deja.com wrote on 12/08/2003 15:39: > > "Fred Marshall" wrote in message > news:1sqdnXk0xro9V0miRVn-gQ@centurytel.net... > > > > "Bhaskar Thiagarajan" wrote in message > > news:br2gmc$257eu...

Re: Remez algorithm and filter lengths

Hello Rune, Specifically the Remez algorithm is one for finding the best fit polynomial in a Chebyshev sense (I..e, minimize the maximum error). And yes the polynomial order is prespecified. Now in the world of DSP, many call the Parks-McClellan algo a Remez algo. But actually the PM algo is one...

Re: Fast response filter?

"Fred Marshall" a écrit dans le message de news:LZednWzJw5Rd-5DdRVn-hA@centurytel.net... > Patrick, > > ??? Being linear phase means that it's definitely not minimum phase. I > don't think either Bob or I suggested anything different. Indeed, that's why I canceled my msg. I didn't...

Re: FIR roots and frequency response

Matt Timmermans wrote: > "Bob Cain" wrote in message > news:c0s60402rrh@enews3.newsguy.com... > > > Actually, I've always thought [...] > > From what I'm reading in the thread I'm still not sure > > whether that is right or dead wrong. :-) > > > Yes, what you say works fine ...

Re: Mirror Image of Magnitude Response

Fred Marshall wrote: > "I. R. Khan" wrote: > > I have a real impulse response h of an FIR filter, and its frequency > > response H is pure imaginary (if I rotate h to left by half of its > length). > > I want the impulse response of a filter whose magnitude response is > > 1-Abs[H]....

Re: Minimum phase filter design using cepstrum methods?

"Michael Numminen" wrote in message news: ... > Hi Ron! > > To your question, see below. > > "Ronald H. Nicholson Jr." skrev i meddelandet > news:c3ahbu$5cp$1@blue.rahul.net... > > I've run across a few web pages describing how to convert an arbitrary > > FIR filter to a m...

Re: Complex version of an impulse

> Subject: Re: Complex version of an impulse > From: Jerry Avins jya@ieee.org > Date: 4/16/2004 11:46 AM Eastern Daylight Time > Message-id: > > robert bristow-johnson wrote: > > > In article da4d20d8.0404152109.7ed6b813@posting.google.com, Impulse at > > impulse@e.coolworks.com wrot...

Removing high frequency hum from audio

This should be a familiar problem to many of you. I recorded animal vocalizations at 100kHz, and they have a frequency content from 200Hz to about 35kHz. Unfortunately (what with old age...) I didn't hear that a TV monitor in the room generated a very strong 16kHz and 32kHz. It's quite stable in...

Re: How can I create filter with such a low lag?

"Vadim" wrote in message news:e58bda6b.0405061617.54b050bc@posting.google.com... > I need to create a lowpass digital filter with a very low lag. What > type of filter could provide such a lag (look at link)? > http://shareftp.narod.ru > > red points is the original data, blue and gr...

Envelope Follower

Hi, i´m new to dsp (since few weeks) , so i hope anyone could help me: I´m coding a audio dynamik-tool (vst) and need a good Envelope-Follower/Detection. I´ve allready tried some different ways: 1. Simply abs and lowpass filter - problems: - ripples in low frequencies, - not linear i...

most effecient hilbert transform method

What would be the most effecient method of hilbert transofrming audio? my filter program genertates too many taps for the low frequency performacnce i require. I don't want to use FFT due to memory constraints in the DSP. Thanks in advance ...

Re: FIR Filter

phuture_project wrote: > I've just designed a FIR bandpass filter centered on 9 kHz with 401 > coefficients. > > My aim is to recover the max of the incoming signal (which is at 9 > kHz). I thought that searching for the max of the y(n) i'll recover > this maximum but actually this ma...

comp.dsp FAQ [1 of 4]

Archive-name: dsp-faq/part1 Last-modified: Tue Oct 19 2004 URL: http://www.bdti.com/faq/ FAQs (Frequently asked questions with answers) on Digital Signal Processing The world-wide web version of the comp.dsp FAQ is maintained and sponsored by Berkeley Design Technology, Inc....

Minimum phase vice versa minimum phase

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. b...

Designing IIR hilbert transform pair

I'm looking at designing a pair of all-pass filters whose outputs are 90 degrees phase shifted from each other. The purpose is to create an analytic signal but avoid the processing required for the true FIR-based Hilbert filter, which is quite large for wide-band audio signals (e.g. 20-20kHz, 48...

Re: baseband filtering

Folks, I have the following so far for question (2) below. Basically, I wanted to modify my filter with real coefficients h(t) such that i modify only the positive frequencies of my (complex) signal s(t). So far I have the following: Let H(f) represent the FT (Fourier Transform) of h(t). ...

Hilbert transform of ARMA filter

Suppose I have an ARMA filter of order N with known A and B coefficients. Does there exist another ARMA filter of order N whose impulse response (and output) is the Hilbert transform of the original filter? If so, what is the expression relating the two sets of coefficients? Thanks! ...

Re: Crossover networks. Can someone recall the name?

On Wed, 14 Dec 2005 15:50:46 -0600, Greg Berchin wrote: > On Wed, 14 Dec 2005 22:22:28 +0100, Andrew Reilly > wrote: > > > a perfect (for whatever > > definition of perfect actually works in practice) brick-wall crossover > > ought to improve the off-axis behaviour by ensuring that th...

Re: ringing: minimum vs linear phase

Mark wrote: ... > I agree a listening test would be interesting. > > But I'll also muse for a bit. > > My thought experiment is to think about the transient (impulse > response) of 5 kHz low pass filters fed with impulses. > > I classify the filters as min phase vs linear phase a...

Hilbert transform question

Hi Guys, I have a question about Hilbert transformer applications. First, we can build Hilbert transformers using a tapped-delay line structure (like a tapped-delay line FIR filter.) An ideal Hilbert transformers (HT) would have a freq magnitude response that's flat over the HT's ...

Hilbert Vs bandpass filter in VSB Modulation

Hi, I am implementing a RF modulator for video signals. I am caught in dilemma of choosing between two different implementations strategies, One that involves taking hilbert transform of the video and then modulating it while the other modulates video to IF and then bandpass it to get the VS...

FIR filter phase shifter

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, Ji...

Re: Frequency Dependant Resistor

Bob Cain wrote: > What kind of thing, within linear systems theory, is a > frequency dependant resistor? If I construct the IFFT of a > frequency domain function is purely real I get what looks > like a linear phase filter rotated so that the peak is at > the front which seems like nons...

Re: FFT re-scaling or interpolation

Hi Andor, "Andor" wrote in message news:1118904008.586422.316930@z14g2000cwz.googlegroups.com... > [...] "Calculation of a constant Q spectral transform". [...] > I'm wondering, is this equivalent to applying Goertzel filters with the > k's logarithmically spaced, where each filter's N...

Re: FIR filtering in the Fourier domain (Free Filter Design Program)

"Philip de Groot" wrote in message news:Xns968DAAD70D456groot877zonnetnl@137.224.11.5... > Hello, snip > > 1. How exactly to multiply the Fourier transform with the FIR > coefficients. > 2. Where to obtain the FIR coefficients (there are many websites; please > recommend). ...

Envelope Detector using Hilbert Transform

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 receiv...

Re: Alternative to Hilbert Transform

holtkamp wrote: > When there is a single dominant frequency present (and good fringe > contrast), I can use a Hilbert transform to give me a time dependent phase > (and thus the frequency from the time derivative of same), but the Hilbert > transform seems to be very sensitive to DC offse...

How to shift frequencies in a non-flat manner

Following somebody's advice, I'm making a topic to expose my problem without talking about possible solutions. But first of all I'd like to restrict this topic to solutions on how to implement frequency shifting (that doesn't involve performing a DFT and shifting bins), I don't want to hear ab...

Re: Phase measurement techniques

Al Clark wrote: > I have a customer that needs to measure the phase difference between two > signals. > > Here are some of the parameters: > > 1. The waveform is very oversampled. > 2. Each signal may have a different amplitude > 3. Averaging the result is allowable. > 4. The ...