In signal processing, a filter is a device or process that transforms a signal by selectively choosing specific frequencies and leaving others with appropriate gain or as it is.

learn more… | top users | synonyms (1)

0
votes
0answers
10 views

How to find roll-off from its transfer function equation?

I want to find roll-off (unit dB/decade) of highpass filter $H(z) = 1 - a \cdot z^{-1}$. I derived magnitude response to be $1+a^2 - 2a\cdot \cos(\omega)$ To find out roll-off factor from above ...
0
votes
0answers
12 views

Mel vs. Gammatone filterbank

After some research about the problem of segmenting voice from unvoiced sounds I found that most of the papers deal with an operation to simulate the cochlea filtering of the signal. I found two ...
0
votes
0answers
37 views

How to Estimate Attenuation Rate ($\alpha$) of band limited filter impulse response

For 2nd order systems, we could find Impulse Response Attenuation (Damping) Rate ($\alpha$) from Q-factor equations, combining $Q = \frac{f_0}{\Delta f}$ and $Q = \frac{2\pi f_0}{2\alpha}$ for 2nd ...
-1
votes
0answers
21 views

how to code fir filter in matlab using discrete equation?

I want to design a fir filter in matlab using the discrete equation. I won't be using any built in functions. For doing so what will be the coefficient? How can i compute it?
2
votes
0answers
52 views

Design a Filter with Resonance

I want to design a Filter with a variable Cutoff Frequency and Resonance. However in the source I'm using there is a Bi-Quad Filter with two IIR and two FIR Filters designed like this : ...
1
vote
1answer
142 views

Is it necessary to downsample data for filtering of low frequencies?

My data is sampled at $f_s = 32kHz$, and for a particular analysis I'm only interested in signals with certain frequencies from $f_{highpass} = \text{0.1Hz}$ to $f_{lowpass} = \text{300 Hz}$. When I ...
1
vote
1answer
78 views

Question on Chebyshev filter

What does the output signal look like when you insert $2000 Hz$ square pulses to chebyshev filter with parameters: (Matlab syntax). I tried to do it like this but I don't understand where it goes ...
0
votes
0answers
27 views

Generate band limited white noise to be used as excitation in ODE

I am trying to repeat an example I found in a paper. I have to solve this ODE: $25 \ddot{x} + 15 \dot{x} + 330000 x = p(t)$ where $p(t)$ is a white noise sequence band-limited into the 10-25 Hz ...
0
votes
0answers
20 views

Continuous wavelet function spanning piecewise functions

Is there a continuous wavelet function that spans the space of piecewise: linear functions (as Haar wavelet spans the space of piecewise constant functions). polynomial functions?
1
vote
1answer
53 views

Modulation and DeModulation

I have a question regarding Modulation and Demodulation. In question 5.21 (b) from the "Discrete-time Signal Processing" book by Oppennheim, he asks to find out the type of filter used in the attached ...
0
votes
0answers
24 views

Basic Fourier Transform filtering and LTI frequency response using Matlab

Not sure if this is the right place to ask... I'm new to Matlab for LTI signal processing and wondering if anyone can help with something that I'm sure is meant to be basic. I've spent hours and hours ...
0
votes
0answers
10 views

Butterworth filter for quadricorrelator

I am trying to implement a FM demodulator (quatricorrelator) according to this paper (3.1 & 3.2). I am now struggling to figure out the proper values for my filters, as described by the paper: ...
1
vote
1answer
26 views

Splitting a Complex FIR Filter into Real Filters

I'm working on a software defined radio project on a PIC32 MCU. I'm using a CMX 973 quadrature frontend to provide I and Q. I'm independently low passing I and Q with carefully matched analog filters ...
-1
votes
0answers
9 views

Segment audio signals + calculate the variances

I don't know any thing about Matlab, but I need to use it to analyze data as a part of my current project. So, could you please help me to figure how to do the following ? First, I need to segment ...
0
votes
0answers
42 views

Is it possible from 1 order cascading low pass filters to make a butterworth filter

Today I have heard, that is "impossible" to make low pass butterworth filter from cascading 1 order low pass filters(RC). I am trying to discover is that true, but I can find little to none ...
2
votes
1answer
65 views

Applying filter in scipy.signal: Use lfilter or filtfilt?

I saw in a SO thread a suggestion to use filtfilt which performs backwards/forwards filtering instead of lfilter. What is the ...
0
votes
0answers
40 views

Filter for audio sampler

An audio sampler (in "music" terms because I'm not experienced in dsp) is a software which loads an audio file and allows you to play it at different pitches by slowing it down and speeding it up. ...
0
votes
1answer
28 views

Orthornormal Wavelet scaling function coefficient calculation (filter design)

I stumbled upon an equation system as shown in the image below while designing an orthonormal wavelet filter. I tried solving the equation (5.63,5.64,5.65 => 5.66) but no avail. I'm not sure if ...
1
vote
1answer
39 views

3 Band Crossover with Linkwitz Riley Filter

I would like to write a 3 Band "EQ" with BiQuad Filters but for some reason, I get some phase distortion. The problem already got discussed here. I found out that I need to work with Allpass filters ...
0
votes
2answers
37 views

How do I take the real part of this bandpass filter system's output?

I'm stuck on a final step in this problem. Essentially, there are the two systems above, which we'll call System 1 (Fig. 4.26, with ideal lowpass $H(jw)$) and System 2 (with $H_1(jw)$). The ...
0
votes
2answers
41 views

Butterworth filter in python

I'm trying to use a Butterworth filter in Python as described in this thread with these functions: ...
0
votes
0answers
25 views

How to perform Wiener deconvolution in Matlab

I have an output signal y which is an input signal x convolved * with an impulse response ...
0
votes
1answer
55 views

FIR vs IIR complexity

IIR filters have feedback whereas FIR filters don't use feedback. Then why is it said that FIR filters sometimes have the disadvantage that they require more memory and/or calculation to achieve a ...
0
votes
2answers
28 views

How can you intutively figure what the impulse response will behave like?

I was trying to understand convolution from this book and came across this figure: Here, the input signal is a combination of a sine wave and a ramp. In fig a, the input signal is convolved with a ...
0
votes
0answers
37 views

What are the frequency components of a sampled signal after low-pass filtering?

A continuous signal $x_a(t)$ is a linear combination of sinusoids of frequency 250 Hz, 450 Hz, 1 kHz, 2.75 kHz and 4.05 kHz. The signal $x_a(t)$ is sampled at $f_s=$1.5 kHz, and the resulting digital ...
1
vote
0answers
82 views

DSP interview question: use of the identity in development of a significant transform

I'm preparing interview and found this question. But I don't really understand what is the question. Does it ask about Fourier transform or Z transform? How the simple identity $$xy=\frac{1}{2}x^2 ...
1
vote
0answers
50 views

MATLAB implementation Spline Fitting

Let $s(t)$ be a signal that can be approximated by a uniform spline function of order $K$ (say $K=2$): $$s(t)\approx\sum_{n\in \mathbb{Z}}c_n\beta_+^{(K)}(t-n) $$ Suppose that we know the ...
0
votes
1answer
35 views

Perfect reconstruction ( Single Channel )

I was trying to solve the following problem . Given an input x[n], consider upsampling by 2, followed by interpolation with a filter having z-transform H(z) for magnification of the signal. ...
-1
votes
0answers
30 views

Filter design using inverse dft

How do I get the cut-off frequency? A filter is needed to recover $x_2(n)$ from the signal: $$x(n) = x_1(n) + x_2(n) + x_3(n)$$ where: $x_1(n) = \sin(0.5n)$ $x_2(n) = \sin(1.4n)$ ...
0
votes
1answer
56 views

What is the Roll-Off factor of a Root Raised Cosine Filter?

When designing a root raised cosine filter on Matlab, there is an option to set its "roll-off factor". What does this mean? I have been looking around for a while but I can't seem to find a clear ...
1
vote
1answer
85 views

How to extract IF (instantaneous frequency) from Discrete Hilbert Transform?

The continuous time Hilbert transform is $$\hat x(t) := x(t) + j\left( p.v. \left\{\frac{1}{t\pi} \ast x \right\} (t)\right)$$ where $$ \theta(t) = \tan^{-1}\left( \frac{ p.v. \left\{\frac{1}{t\pi} ...
0
votes
0answers
59 views

Transient cancellation in non causal filtering

I am using the exponential moving average filter (IIR): $$y(n)=\alpha{}\cdot{}x(n)+(1-\alpha{})y(n-1)$$ in a foward-backward filtering scheme, in order to eliminate completely any phase response ...
0
votes
0answers
32 views

moving average filter vs up sampled low pass filter

I'm experimenting on different low pass filters in matlab. I've considered a vector x=[1 0 1 0] ; an up sampled low pass filter by a factor of two. And the frequency response turned out to be And in ...
0
votes
0answers
34 views

How to remove modulation frequency and its harmonics

I have amplitude modulated sound file signal and after demodulation using FFT I got fundamental frequency and its harmonics and of course noise. How to remove that fundamental modulation frequency ...
0
votes
0answers
23 views

Does a Laplacian Pyramid and a Gaussian Pyramid use the same Kernel?

Does a Laplacian Pyramid and a Gaussian Pyramid use the same Kernel? I have to program this for an assignment, but it seems like the kernel used in both cases is a gaussian filter. Is that the case? ...
1
vote
0answers
35 views

Direct Form I Filter from Direct Form II

I am looking a way to make outputs match. For len 160 wouldnt the error be significant? For 40 ? I have included some sample output showing the error. Direct Form 2 Original ...
0
votes
1answer
76 views

Strange Results from Biquad Peaking EQ Filter

I have a signal that I am filtering on an MCU (STM32F405) that I get from a codec (CS42L52). It is being sampled at a sampling frequency of 48kHz (actually between 48400 and 48600). I have a ping ...
0
votes
2answers
72 views

FIR vs IIR filter

In my filter design, FIR gives order 14 and IIR gives order 8 or 4. so what are the disadvantages of IIR design?
0
votes
1answer
45 views

Why bother impluse invariance, bilinear transform?

Now that we can operate in the frequency domain by way of FFT. We can implement ideal lowpass, highpass and bandpass digital filters. then why there many textbooks talk about designing digital ...
0
votes
2answers
53 views

multiplications per second for FIR filter

In my filter the fs=300MHz. and no. of coefficients is 31. I have following queries. (1) what will be the number of multiplications per second? (2) can multiplication be done in one clock cycle? (3) ...
0
votes
2answers
54 views

A filter in IOS to eliminate a certain frequency range

I am making an application where I record sound on an IOS device using the built-in microphone. I want to process the sound in such a way that only frequencies in the 16-18 kHz range remain; the other ...
1
vote
2answers
46 views

Signal corrupted using peak (biquad) filter at low frequencies

I'm trying to process an audio stream through a series of peak filters at static frequencies that range from 0 to nyquist. If I start the range from anything below about 25Hz, the audio gets corrupted ...
0
votes
1answer
54 views

FIR filter design

I am designing an FIR filter.my specs are fs=300MHz, Fc=45MHz, Fs=75MHz, passband gain=3dB , stopband attn=>40dB. what parameters I values of a have to provide in firpm for the minimum order filter ...
0
votes
0answers
63 views

DFE for QPSK in matlab

I need to do a DFE for QPSK simulation in matlab. The system, consist in QPSK signal transmitted with power = 1. Then pulse shapping with a square raised-cosine filter. Add AWGN and a filter receptor ...
0
votes
0answers
24 views

PLL as bandpass filter?

I'm working on writing the synchronization for MSK and I need to create two PLL's as band pass filters. How would I design the filter part of the PLL to make the system act as a band pass filter?
0
votes
0answers
27 views

How to smooth out a noisy signal?

I've been waiting for an answer to my question here therfore I decided to ask the question in a different way. Could anyone please help me to smooth out the signal below!!. I've no experience of ...
0
votes
0answers
27 views

Unstable Cascade of Allpass Filters

I'm working with cascaded second-order allpass filters. Each filter is characterized by conjugate pole/zero pairs. In a long (>50) cascade of identical filters, the pole locations of the overall ...
1
vote
1answer
27 views

Iterative Kalman filters and system parameters estimation

i am working recently on a project in which i want to implement a Kalman filter as being an observer, and i couple this observer with a state feedback controller that produces control actions ...
1
vote
1answer
74 views

Discrete algorithm for low pass filter

I am working on a position controller for a marine vessel. I have an measurement signal containing the y-position of the vessel that consists of both low frequency (<.1 rad/s) and high frequency ...
-1
votes
1answer
45 views

Can all pass filers have a pole/zero at the origin of the root locus

I came across this root locus diagram in my DSP class. It has a pole at the origin, another pole within the unit circle and a zero just outside the unit circle. I understand that the poles and zeros ...