Create the page "Filter" on this wiki! See also the search results found.
- equation (moving average filter) :a. Find the impulse response h[n] for this filter. Is it of finite or infinite duration?4 KB (638 words) - 10:04, 16 October 2013
- b) Could the vocal tract be modeled using an FIR filter? Explain.2 KB (353 words) - 21:19, 31 October 2013
- ...proposed modeling the vocal tract as an LTI filter and approximating this filter by a sequence of tubes.2 KB (305 words) - 07:05, 25 October 2013
- **notes on digital Filter design: [https://engineering.purdue.edu/~bouman/ece438/lecture/module_1/1.76 KB (759 words) - 08:10, 11 November 2013
- #an all pole filter.2 KB (281 words) - 05:54, 30 October 2013
- ...If you use the same threshold device as that of Part (a), explain how the filter you designed in Part (a) should be modified to work on the third day.15 KB (2,507 words) - 01:05, 5 November 2013
- (b) No, it must be an IIR filter as it must have poles. As explained in (a), the difference equation describ1 KB (258 words) - 12:16, 7 November 2013
- ...If you use the same threshold device as that of Part (a), explain how the filter you designed in Part (a) should be modified to work on the third day.17 KB (2,710 words) - 10:07, 5 November 2013
- Consider the following FIR filter: a) Write a difference equation that can be used to implement this filter.2 KB (262 words) - 06:50, 15 November 2013
- ...arate it. We then considered another filter (edge detector). Although that filter is not separable, we were able to write it as a sum of two separable filter3 KB (367 words) - 07:27, 15 November 2013
- ...n]. We then proceed to demonstrate how to use the formula using an average filter and a 6x6 digital image. The issue of the boundary conditions was discussed2 KB (315 words) - 06:46, 15 November 2013
- Therefore the filter can be separate into two 1-D filters.4 KB (518 words) - 00:11, 11 December 2013
- ...lt in [http://www.mathworks.com/help/vision/ug/object-tracking.html Kalman Filter] function. ...it is too noisy of choppy. First, we can get rid of noise with a Gaussian filter:11 KB (1,762 words) - 09:42, 13 February 2014
- ....m.zip zpgui3.m] A MATLAB GUI showing the effect of poles and zeros during filter design. ...equency domain view of downsampling (explain why decimator needs a lowpass filter before the downsampling). DEADLINE October 1013 KB (1,944 words) - 16:51, 13 March 2015
- ...p://engineering.purdue.edu/VISE/ee438L/lab5/pdf/lab5a.pdf Lab 5a - Digital Filter Design] ...p://engineering.purdue.edu/VISE/ee438L/lab5/pdf/lab5b.pdf Lab 5b - Digital Filter Design]2 KB (380 words) - 08:18, 2 December 2014
- ...ecn.purdue.edu/VISE/ee438L/lab5/pdf/lab5a.pdf Lab 5a (First lab on digital filter design)] *Week 9-(10): Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering)9 KB (1,320 words) - 04:46, 11 September 2014
- ...r. Be sure to show how you calculated the cutoff frequency for the digital filter. ...is. Be sure to show how you calculated the cutoff frequency of the digital filter.3 KB (486 words) - 06:19, 22 September 2014
- *an all pole filter.2 KB (279 words) - 06:37, 19 November 2014
- ...spectrum while preserving the one at baseband. Conceptually, the simplest filter is7 KB (1,178 words) - 20:16, 18 December 2014
- ...better show how the original signal could not be found by using a low-pass filter. The takeaways are good, but the spelling of "Shannon" should be checked.2 KB (394 words) - 05:38, 15 October 2014
- The middle-bottom plot shows the ideal low-pass filter (LPF) that can be used to recover the original signal from this sampling. I The middle-bottom plot shows the time-domain representation of the low-pass filter with gain <math>T</math> and a cutoff frequency <math>f_{co}</math> of 10 H10 KB (1,650 words) - 19:04, 16 March 2015
- ...d be good because it feels that it is just added, while the reason for the filter is important. The colors on the plots where really helpful but hard to see3 KB (525 words) - 05:39, 15 October 2014
- If we use a low-pass filter with gain <math>T_{s}</math> and cutoff frequency between <math>f_{m}</math2 KB (287 words) - 19:05, 16 March 2015
- ...<math> X(f) </math> (shown in red). Therefore we can use a simple lowpass filter with gain <math> \tfrac{1}{f_s} </math> and cutoff frequency <math> \tfrac{ ...ds to <math> X(f) </math> (shown in red) can be recovered using a bandpass filter with gain <math> \tfrac{1}{2a} </math> and cutoff frequencies <math> a \tex6 KB (1,008 words) - 19:04, 16 March 2015
- ...omain. Also, it explains process of decimation and why it needs a low-pass filter. ...ing occurs. Downsampler is a part of a decimator which also has a low-pass filter to prevent aliasing. LPF eliminates signal components which has7 KB (1,035 words) - 19:07, 16 March 2015
- We need a high pass filter that filters out signals below the frequency 60Hz. ...equency component. In order to remove the annual cycle, we need a low pass filter.3 KB (564 words) - 06:24, 10 October 2014
- If <math>f_{MAX} > 1/{2*T_2} </math> is true, then you must use a low-pass filter before downsampling.4 KB (566 words) - 09:59, 14 March 2015
- ...ould first pass the original <math>x_1[n]</math> signal through a low-pass filter with <math>f_c = 1/(2T_2)</math> BEFORE downsampling.<br><br> Since the possible aliasing is caused by the downsampling, trying to low-pass filter after the downsampling will be too late and won't be able to get rid of the4 KB (641 words) - 09:58, 14 March 2015
- Low-Pass filter of cutoff π/2, gain 2 is applied.<br>2 KB (206 words) - 10:01, 14 March 2015
- ...it's signals every 2π and to prevent aliasing, decimator needs a lowpass filter before the downsampling. =====1 KB (179 words) - 06:07, 13 October 2014
- ...nd it is easy to follow the logic. Also the example helps to explain why a filter is needed and limitations. ...explanation is very clear and concise. The explanation of why the low-pass filter is needed is also good. Great job!4 KB (594 words) - 05:41, 15 October 2014
- ...The motivation for and derivation of the cutoff freqeuncy of the low-pass filter was expressed well. Overall, the content was very good.3 KB (405 words) - 05:41, 15 October 2014
- ...der to remove or at least attenuate the unwanted image spectra, a low pass filter must be placed immediately after upsampling. In the time domain, the effect ...s filter is used. So the combiation of an upsanpler followed by a low pass filter can be referred to as an interpolator.5 KB (790 words) - 10:01, 14 March 2015
- ...so includes low-pass filter, also you did not clearly explain why low-pass filter is needed. Good explanation of derivation formula and clear graphs.<br> You forgot to put low-pass filter before the decimator. Except that everything is very clear and graphs and e3 KB (560 words) - 05:42, 15 October 2014
- ...w your point. You did have very good explanation about how to use low pass filter when doing upsampling. In the end, the practical example is very good to ha6 KB (1,072 words) - 05:42, 15 October 2014
- ...it's signals every 2π and to prevent aliasing, decimator needs a lowpass filter before the downsampling. =====1 KB (192 words) - 10:00, 14 March 2015
- #Send through a LPF (low pass filter) LPF with filter that has cutoffs at <math>\frac{\pi}{D}</math> and <math>\frac{- \pi}{D}</m3 KB (565 words) - 10:01, 14 March 2015
- Sampling at frequencies much larger than Nyquist requires a filter for reconstruction with a less sharp cutoff. A digital LPF can be used to t3 KB (542 words) - 10:00, 14 March 2015
- ...de of the frequency response of that system, it will be clear what kind of filter this is.2 KB (250 words) - 06:41, 3 November 2014
- equation (moving average filter) :a. Find the impulse response h[n] for this filter. Is it of finite or infinite duration?4 KB (640 words) - 06:37, 3 November 2014
- ...ed analyzing Filter B. After that, we compared the effects of Filter A and Filter B on pure frequencies.2 KB (221 words) - 06:21, 5 November 2014
- Today we generalized filter A and Filter B to the case of any causal system defined by a difference equation with co2 KB (221 words) - 07:30, 10 November 2014
- b) Could the vocal tract be modeled using an FIR filter? Explain.3 KB (462 words) - 07:25, 17 November 2014
- ...ll be a two zero filter, using complex conjugates. The Z-transform of this filter is ...there is some true signal at the same frequency as the vuvuzelas. For this filter, it will be r=0.95. Theta is calculated as5 KB (860 words) - 10:26, 20 November 2014
- ...we saw that the DT filter corresponding to the vocal tract is an all pole filter. This is because, according to our sequence of tube model, the transfer fun2 KB (329 words) - 06:44, 24 November 2014
- Consider the following filter: a) Write a difference equation that can be used to implement this filter.5 KB (545 words) - 12:20, 9 December 2014
- ...n]. We then proceed to demonstrate how to use the formula using an average filter and a 6x6 digital image. The issue of the boundary conditions was discussed ...9_mboutin_plotCSFTofbasicfilters|its plot]]. Using the separability of the filter greatly facilitated the computation of its Fourier transform.3 KB (373 words) - 07:08, 24 November 2014
- b) Could the vocal tract be modeled using an FIR filter? Explain. No, it must be an IIR filter as it must have poles. As explained in (a), the difference equation describ6 KB (1,031 words) - 11:27, 29 November 2014
- Consider the following filter: a) Write a difference equation that can be used to implement this filter.10 KB (1,181 words) - 16:07, 2 December 2014
- 2. Filter the projections <math>g_{\theta}(r) = h(r) * p_{\theta}(r)</math><br /> =Projection Filter Analysis=6 KB (927 words) - 19:26, 9 February 2015
