Questions tagged [infinite-impulse-response]

Infinite impulse response (IIR) is a property of signal processing systems.

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152 views

How many coefficient sections should I select when designing filters with MATLAB FDATOOL?

I attach a picture about the results of a filter design in Matlab. It says that it has 12 sections. One section contains the coefficients of the transfer function. If I'm implementing this filter in ...
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1answer
237 views

Can a Goertzel filter response be mapped to a Direct Form II Biquad IIR Filter

I am looking at the structure of the Goertzel Filter, and it seems like I can map its coefficients to the standard implementation of the CMSIS DSP Biquad Cascade IIR Filter. Is this possible or should ...
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1answer
108 views

Is this sampling frequency right?

I designed a Butterworth second order bandstop filter on Matlab with the following parameters: Notch frequency: 150 Hz (the frequency I want to remove) Bandwidth = 8 Hz, that is Fc1=146 Hz Fc2=154 ...
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81 views

should block diagrams in Direct Form 1/2 contain empty branches to fit structure?

When creating a block diagram in a well-recognized structure (Direct Forms 1&2 in particular), does one remove branches where the coefficient is 0? This picture includes a coefficient of $b_1=0....
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164 views

Can I combine an IIR (an EWMA) with a maximum?

I'd like to have a exponentially weighted moving average (EWMA) which is raised to the value of the input signal whenever this is higher than the filter output. This taking of the maximum shall take ...
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126 views

Can I use IIR or FIR to get a sliding mean?

My output signal shall be computed by (meta code): o = 0 # or some arbitrary initial value for i in input: o = (o * 99 + i) / 100 print o I call this a "...
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1answer
78 views

historical context of impulse response-convolution

Hi All: I have zero background on the history of LTI systems so I was wondering when the concept of LTI and the fact that an LTI signal can be be completely described by its impulse response came ...
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1answer
507 views

Design Lowpass Filter With -7.5 dB/Octave Rolloff

I am working on a project that needs a lowpass filter with a -7.5 dB/Octave rolloff. This is somewhere between the rolloff of a one-pole and two-pole filter, so I wanted to ask what some standard ...
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616 views

Get difference equation of a filter from MATLAB transfer function coefficients

I need to get the difference equation of a specific elliptic filter. I calculated the transfer function coefficients in MATLAB with: ...
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Matlab `filtfilt` provides excessive transient

I observed that filtfilt suffers from an undesired behavior when I provide IIR bandpass filters having steep transition bands. Specifically, the output signal exhibits excessive transient response; ...
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How to transform a Fractional Order Laplace Transfer Function into a digital filter?

I'm working with loudspeaker impedance analysis. Electrical behavior of loudspeakers can be modeled with RLC networks. But real loudspeakers have components, that exhibit some non-linear and frequency ...
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why is that filter FIR not IIR [duplicate]

how is this filter an fir filter should not it be an iir because of the y[n-1 ] term
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Which IIR form has lowest round-off noise?

We are making some audio effects using IIR, and seeking ways to optimize its round-off noise. After searched some materials, most of them tells transposed form II has lowest round-off noise, and many ...
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37 views

Incorrect Butterworth filter outcome using Rational Transfer Funcion

I've been playing with Second-Order Butterworth high-pass filter for a while. I've calculated the coefficients using Matlab function and I wanted to use them to filter simple sine wave generated by ...
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2answers
108 views

Filtering without using any transforms

A composite signal is defined as $$x(t) = \sin(2\pi 10t) + \sin(2\pi 40t) + \sin(2\pi 60t) + \sin(2\pi 50t)$$ Separate the four frequencies without using any of the transform techniques.
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Stable digital filter with leading zeroes

I'm looking construct a stable pole-only filter where the feedback coefficients start with a block of zeroes, i.e. \begin{align} a_0 &= 1\\ a_i &= 0, \textrm{ for}\ 1 \le i \lt k\\ a_i &\...
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1answer
286 views

Is initializing a digital filter's output with no "momentum" a non-trivial task?

I'm working on implementing a filter with a very slow step response. This filter is implemented as a cascaded second-order-section filter (transposed direct form 2). I'm using the output of this ...
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3answers
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Is there a common name for the first order IIR averaging filter?

Is there a term to describe this IIR averaging/smoothing filter? $ y[n] = \alpha x[n] + (1 - \alpha) y[n - 1] $
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1answer
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Complex IIR to Real IIR

I have created an IIR design algorithm that generates complex coefficients (there is no symmetry in the poles and zeros). However, the IIRs will be used to filter a real signal. Is there a closed form ...
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1answer
195 views

Does this equation have a closed form

I am on chapter 6 of "Understanding digital signal processing", 3rd Edition by Richard G. Lyons. But I am stuck on question 6.15. Here is the network in question. I wrote down the difference ...
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1answer
59 views

Audio signal series' beginning points(about 50) distort after a bandpass filter

I am using some bandpass filters to detect a certain freq of the sampled audio series in noisy environment. Everything is OK except this: I found that for many bandpass filters I tried(butterworth, ...
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2answers
631 views

Convolution with an impulse response does not give the same result as the frequency response

Context I am trying to design a linear-phase FIR filter with the following frequency and phase responses: Designing the filter Given its characteristics (peak filter at 1kHz with 9dB gain and Q=7), ...
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886 views

Design a linear-phase FIR filter approximating the magnitude of a given IIR filter

I have a biquad IIR filter from which I want to get a linear-phase FIR. I saw that related question but the OP does not care much about phase. I have troubles getting the exact same magnitude curve ...
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Whats is the difference between FIR/IIR filters and Chebyshev/Butterworth filters

I am new to Signal Processing. From my understanding -- FIR/IIR just refer to the placement of poles and zeros in the z-domain helping us achieve convolution, if FIR and ??? in IIR. Chebyshev and ...
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1answer
95 views

Generic response of an IIR filter from its transfer function

How can I derive the general response of an IIR filter from its transfer function? I know that: $$H(z)=\frac{1}{1+\sum\limits_{m=1}^N{a_m z^{-m}}}$$ Thus: $$Y(z)=X(z)H(z)=\frac{X(z)}{1+\sum\limits_{...
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383 views

Typical filter orders

I am on my first DSP project where I want to implement lowpass/highpass filters for ECG application. I have started from MATLAB to give me orders and coefficients for my desired specs. I prefer to go ...
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Digital Filter Design Accuracy [duplicate]

Having only dealt with digital filter scarcely, the question dawned to me when I used the firls function in matlab to design an equalizer with a certain gain response. In general, can we prescribe ...
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1answer
776 views

How to implement a FIR filter with correction of group delay on c#?

I read respect to digital filters, and implemented both IIR and FIR filters and I found that this type of filters have a called "group delay". Now I know that the IIR have a non-linear group delay and ...
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257 views

Fixed point Error Spectrum Shaping (noise shaping) for biquads

Based on that article and all the resources on the internet that advise the use of noise shaping for fixed-point filter implementations, I tried to make a few simulations on Matlab for the biquad I'm ...
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436 views

Converting floating point IIR to fixed point arithmetic, given its difference equation

Is there some general way to perform the task given in the question? I have a filter given as follows: ...
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1answer
325 views

Impulse response of a continuous system sampled with zero-order hold

I've a continuous system $$F(s) = \frac{K}{Ts+1}.$$ I sample it with zero-order hold with sampling period $T_s$. The discrete system transfer function is $$ \begin{aligned} G(z) &= % \frac{z-1}{z}...
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How to design all-pass filter by specifying phase requirements?

I want to delay the part of signal above (or below) specified frequency, but I don't find any books or tutorials to tell how to design is. The commonly used biquad APFs (such as the one introduced in ...
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2answers
932 views

Trouble with fixed-point arithmetic 2nd order IIR implementation in c

DSP newcomer here! I am tinkering around with a TI DSP and am trying to implement a second order IIR filter in C. Input is 16 bit 2's complement, as is the output, the accumulator is 32 bit wide. I ...
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272 views

IIR filtering vs IFFT filtering by zero binning and making bandpass IIR filters with high sample rates?

I am having some issues making thin IIR filters that work at the sample rate of my data and was wondering if there is a way to improve my method? I am also curious about the deficiencies in using the ...
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1answer
871 views

How do scaling of the coefficients for the poles of a biquad filter affect the gain?

Given a digital filter consisting of a chain of digital biquad filters. Given a digital biquad filter with a normalized Direct form 1 difference equation: $ y_n = b_0x_n + b_1x_{n-1}+b_2x_{n-2} - ...
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1answer
270 views

Behavior of tanh IIR filters

If we insert a tanh function (or any other activation function) between the feedback summation and the unit delays, how will such an IIR filter behave for values of $|a| < $1 and $a > ±1$ ? The ...
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2answers
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Memory for FIR and IIR [closed]

I have doubt regarding memory of FIR and IIR, I have read that output of FIR depends upon present and previous inputs, while output of IIR depends upon, present and previous inputs as well as previous ...
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Is there a tool for automatically converting filter implementation forms?

I am looking for alternate tools that will automatically convert filter implementation forms (or signal flow network diagrams in general) to simplest or alternate forms; for example for going between ...
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2answers
568 views

Manually Windowing an IIR Filter in Matlab

I am currently trying to window an analog IIR filter in order to obtain an FIR filter. Though I realize the fir1 function exists within Matlab, I was wondering to see if it could be done step-by-step ...
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1answer
345 views

Amplitude loss during resampling with IIR

I have a problem in that I am resampling a signal and loosing amplitude. The steps I do are the following, in c++: Upsample/interpolate signal by 16x by padding with zeros. Run an IIR Bessel 2nd ...
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309 views

Software implementation recipe of cascaded Biquad IIR filter

I have been given the task of measuring the performance of Biquad IIR filters on a new CPU that we are evaluating. What I know is that the Biquad filters can be cascaded to form higher order filters. ...
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3answers
629 views

IIR biquad real-time filter just output noises

I've implemented a IIR filter but it outputs only noise (like FM radio out of tuning) and I can't see what's wrong. This is the processing function: ...
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2answers
262 views

Simplify equation of single pole IIR transfer function

Example - Consider the causal stable IIR transfer function $$ H(z)=\frac{K}{1-\alpha z^{-1}}, \quad 0 < \lvert \alpha\rvert 1 $$ where $K$ and $\alpha$ are real constants Its square-...
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Appropriate model for IIR filter to compare with spectral ratios

A quick, basic question. I have measured spectral ratios - comparing a measured signal to an 'ideal' signal. I am fitting a specific, simple IIR filter to that data for transfer functions. So, if I'...
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1answer
272 views

Analog Filter Prototype to Direct Form Second Order Cascade

I've been using the Faust programming language a lot lately for experimenting with DSP, and I've been digging into their implementation of an Elliptic (Cauer) Lowpass filter. Here's an example of such ...
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1answer
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Designing an IIR to generate a specific data sequence

I know that it's possible to design an IIR with specific poles and zeroes to create specific frequency responses. Is it also possible to design an IIR such that when you give it an impulse, it ...
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1answer
884 views

Biquad filter from Matlab to C using "tf2sos"

This is a biquad implementation I use: ...
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1answer
751 views

4th order high-pass filter on a DSP: standard or biquads?

I have two C implementations of 4th order high-pass filter (fs = 16 kHz, cut-off=100 Hz, designed using Matlab ...
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1answer
2k views

If a filter has a non-linear phase lag will the output signal be distorted?

EDIT: From here (https://www.mathworks.com/help/signal/ref/butter.html) it appears that an Butterworth filter has a non-linear phase lag at the normalized frequency of 0.5 onwards. Does that mean that ...
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Filter Order Rule of Thumb

Is there a rule of thumb for calculating the approximate filter order of a FIR/IIR based on passband vs sample rate? I seem to remember reading about it once in Undergrad. I'd think you could make a ...

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