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265 fir questions.

Question 1 $$ H(e^{j\omega})=\sum_{n=0}^{N-1}h[n]e^{-jn\omega} =\mathbf{c}^H(\omega)\cdot \mathbf{h} \tag{1} $$ $$ =\mathbf{h}^H\cdot\mathbf{c}(\omega) \tag{2} $$ $$H(\mathbf{h})=\sum_{k=1}^Kh[k]e^{...

Is this a FIR algorithm?

1 answers, 37 views fir
I'm kind of out of my element here trying to understand this algorithm and I wanted to see if this is a FIR algorithm or not? If this isn't the right place to ask this, can someone point me to a place?...

For converts any causal LTI digital filter into state-space form,there is the following procedure : 1-The general causal IIR filter $$ y[n] = b_0 u[n] + b_1 u[n-1]+... + b_N u[n-N]- a_1 y[n-1] - ...

I found many information in this thesis "Algorithms for the Constrained Design of Digital Filters with Arbitrary Magnitude and Phase Responses",i want to understand it. This question is a ...

I'm currently working with a dataset of $5000$ pulses of $N=15000$ samples each. I managed to implement the RLS algorithms with a FIR M-Tap filter such that $M\leq 15000$ ($150$ seems to achieve the ...

I want to design a FIR filter by using 'cfirpm' function , which can i use the group delay and grid frequency, my code is : ...

What is the output of the convolution meaning which is used in the FIR filter? how can conclude the output of the FIR filter in the output according to the input $x[n]$?

I am trying to derive the mean delay for $M$ filter taps harmonically weighted: $$y[n] = \frac{M\,x[n] + (M-1)x[n−1]... + 1 \cdot x[n−M+1]}{\tfrac12 M (M + 1)}$$. For a uniformly weighted filter, I ...

I'm relatively new to DSP and I'm currently investigating FIR filter and IIR filters. From what I've found FIR filters can be implemented efficiently using the overlap-save method, but I was wondering ...

I work with Magnetic Resonance Spectroscopy data. Data (complex signals) obtained from Bruker scanners (one of the biggest manufacturers of MRS scanners) start with an important group delay - around ...

I'm trying to model a non-linear system using non-linear convolution with Novak's (2010) synchronized exponential sine weep (SESS) that models them with a Generalized Hammerstein (Volterra diagonal). ...

I'm trying to use 5 FIR in a row to improve the S/N ratio on my signal, but I have some difficulties doing it. In fact I have 2 signals, let's call them sig_I and sig_Q. I want to apply my 5 filter ...

I have calculated the transfer function of an FIR filter $$ y[n] = x[n] + α · x[n − R] $$ This is what I have $$ H(z) = 1 + αz^{-R} $$ Now I should plot the square of the amplitude response. So I ...

I am trying to implement a chain of CIC/FIR filters on an ZYNQ FPGA. Using the Xilinx FIR compiler works fine so far but I am unable to properly get all the math. At the moment I have 2 chains of ...

I have an issue with pa olyphase implementation during interpolation. Lets assume I have 256 taps long FIR lowpass sinc filter. In order to interpolate by a factor of 2 I do use two FIR filters (...

Problem: I have the impulse response of a matched filter(therefore its phase and magnitude response. See figure below) of a filter, and I need to implement its response using only off-the-shelf ...

Problem: I'm trying to analyze the behavior of an FIR filter with the following impulse response/kernel: Using Matlab's function grpdelay(myKernel,length(myKernel)), I obtained the following figure:...

Is there a way to convert a FIR to an IIR filter with the most similar behavior?

Sorry if my question is some FIRFiltering basics, but I couldn't find a specific answer after some search. I am trying to design a decimating low pass filter in C (actually in OpenCL with GPU) with ...

Phase response is the relationship between the phase of a sinusoidal input and the output signal passing through any device that accepts input and produces an output signal, such as filter.https://en....

I want to find many difference 'h' with same group delay and length filter using this code : https://gist.github.com/mattdsp/17f508d2b368db61f47689e4622d8841/c1518f22a5aa451951601b56c5d9f72743868bfe ...

I am working though a dsp past paper and I have come across the questions "find the impulse response in the time domain" & "find the transfer function in the time domain" I know its really simple ...

The graph below shows a linear phase response of FIR Hilbert transformer with a constant diff of -0.06135923. I am not sure if I understand the concept of the phase response correctly, but as the ...

I am using a DSP processor to sample signals at 108kHz and I only want to get the DC part out of it. Such a high sampling rate is used because I want to use oversampling to reduce quantization noises. ...

I was reading an article about ultrasonic signals, and I read in it that it is possible to get the envelope of a signal by using a band pass FIR filter. My question: Is it possible to get the envelope ...

As you know, a nonlinear phase filter (in passband), distorts the frequency contents of passband region of signal. In general, Why do we use such filters in the case they cause distortion in the ...

Knowing that computing an FFT is faster if the amount of samples is a power of 2 I have always tried to pad the inputs to Matlab's FFT with zeros until the next power of 2 is achieved. Matlab's ...

Kindly help me to design a FIR filter for eliminating the 750 Hz sinusoidal beep from audio. I shall be thankful to you

I have a confusion about what does a two pass FIR (bandpass) filter with order 40 means?? Passband frequencies are [8 13]. Type 2 FIR is same as two pass?? I have check some previous literature ...

I compared the results of Hilbert Transform as provided in scipy.signal.hilbert() (which is uses I/FFT as you can see in the source code) with its approximation implemented as FIR filter using coeffs ...

I have this design , Can you tell me about the type of filter and the algorithm just viewing this design, or should there be other information?

Can anyone point me towards a formula for a good (for any values of "good") no-lag causal bandpass filter? I am doing sound processing and I need to identify starting points of certain sounds (target ...

I would like to appy multiple filters to a signal at once. Let's assume I got two frequency bands I want to extract. I compute a Butterworth Filter for 300-500Hz and one for 700-800Hz. I would like to ...

I am pretty new to DSP and I need some help to pinpoint my mistake. I am using Matlab to simulate an embedded environment. I have three signals all centered on 50Hz (6.4KHz sampling rate) which have ...

I am trying to implement a FIR filter on FPGA and trying to have a solid understanding of the FIR filter tap delay and sampling frequency. Does the “one tap” delay equal to “1/Fs (sampling frequency)”...

Suppose I have a sinusoidal signal of 50Hz sampled at 6.4kHz and I wished to filter it at: Band pass: 160Hz, Bands stop: 800Hz, Apass:0.1db, Astop: 106db For a FIR filter order of 40 I would get 41 ...

So I have a transfer function $ H(Z) = \frac{Y(z)}{X(z)} = \frac{1 + z^{-1}}{2(1-z^{-1})}$. I need to write the difference equation of this transfer function so I can implement the filter in terms of ...

Say I have a signal which is guaranteed to have a frequency between 110-120 Hz but is corrupted by interference signals that're very close to this frequency range. For example, let the interference ...

I need to interpolate a complex valued bandlimited periodic function using local interpolation. I can have the signal sampled at any frequency I want over at equispace intervals. I am aware that for ...

I'm trying to design a digital low pass filter with a narrow transition band. My sampling rate is 25 kHz, the cut off frequency is 60 Hz & the transition band width is 4 Hz. I'm looking for about ...

$$h[n]=\begin{cases}a^n & \text{if } 0 \le n < N \\ 0 & \text{otherwise}\end{cases}$$ And for which values of $a$ the filter is stable I know that the transfer function will be $$H(z)=\...

I'm newbie novice in digital signal processing. If one has a signal with some noise superimposed in time series, for which type of information/analysis one would use moving average or LP filter in ...

I'm trying to design an equalizer for spectral flatness over a 100MHz signal for an LTE channel emulator. A channel emulator simply multiples channel coefficients with an external signal from a signal ...

In explication ''the geometric interpretation of least squares'' Typically, the number of frequency constraints is much greater than the number of design variables (filter coefficients). In these ...

Following this paper , I am trying to make a least-squares algorithm in MATLAB, but for type I (I know about firls()). ...

I want to design a digital FIR filter to cancel the phase of an analog Butterworth filter (i.e. the phase is smooth) i.e. the filter has unity gain with just an inverted phase component. The ...

What's the fastest way (if possible in the browser thanks to an online tool, or if not possible easily, with Python), to get the frequency response curve (x-axis: Hz, y-axis: dB), when giving just: ...

I'm running an FIR filter hamming window order 161 bandpass with pass frequency range 20Hz-200Hz, but output doesn't sound like the filter is working. It's just distorted output. Following is the ...

I'm trying to generate coefficients for a FIR low pass filter to be used in an FPGA. I'm using python (scipy.signal) to attempt to do this, but am having trouble getting coefficients in a usable form....

So I'm reading the datasheet for MAX98357A. It's an audio amplifier that receives digital signal using I²S bus and has integrated digital low-pass filters of the IIR and FIR types. In addition to ...

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