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  • * Increasing filter size improves detection at the expense of localization
    14 KB (2,253 words) - 12:21, 9 January 2009
  • ==[[ECE 301 Fall 2007 mboutin Filter Types|Types of Filters]]== {{:ECE 301 Fall 2007 mboutin Filter Types}}
    264 B (30 words) - 13:07, 9 December 2008
  • | align="right" style="padding-right: 1em;" | Monday || 03/30/09 || Filter bank interpretation and efficient computation of STDTFT ||
    6 KB (689 words) - 07:59, 2 August 2010
  • ...ndow shows the network that has been selected from the query field and the filter panel.
    4 KB (713 words) - 16:47, 30 November 2011
  • *<math>\omega_c</math>: Cutoff frequency of a filter (<math>\omega_c > 0</math>). (For instance, lowpass filters are nonzero in
    2 KB (406 words) - 11:08, 12 November 2010
  • ...mum amplitude one, minimum amplitude zero and period P. The ideal low pas filter is a band that goes from -fc to fc with constant amplitude one. Once filte ...but the digital filter is in the frequency domain. How do you apply this filter? Any ideas are helpful. Thanks!
    844 B (152 words) - 18:26, 11 February 2009
  • ...trum of the rect (which is a sinc). This is effectively a coarse low pass filter to remove high frequency aliased replicas. <br> ...c^2</math> in frequency. This can also be thought of as a coarse low pass filter but with a sharper cutoff than zero-order hold. <br>
    906 B (143 words) - 12:40, 4 March 2009
  • Plot of the frequency response of the average filter: Plot of the frequency response of the filter:
    950 B (132 words) - 11:52, 28 April 2009
  • == Filter Design== ...ise.www.ecn.purdue.edu/VISE/ee438L/lab5/pdf/lab5a.pdf First lab on digital filter design]
    8 KB (1,226 words) - 11:40, 1 May 2009
  • ...mboutin_DFT_windowedfilter|Graph of the magnitude of the DFT of a windowed filter]]
    7 KB (1,067 words) - 12:05, 25 June 2010
  • ...e sufficiently close, then the signal can be reconstructed using a lowpass filter. ...ntinuous curve and represent an interpolation formula for an ideal lowpass filter H(jw):
    851 B (151 words) - 11:38, 8 November 2008
  • ...)</math>, we can simply low-pass filter <math>x_p(t)</math> as long as the filter, ...ath>\omega_m < \omega_c < \omega_s - \omega_m </math>. Also, the low-pass filter must have a gain of <math>T</math>. This can be represented graphically as
    3 KB (582 words) - 06:11, 16 September 2013
  • To recover, first we need a filter with amplited T when |W| < Wc.<br><br>
    500 B (101 words) - 16:21, 9 November 2008
  • To recover, first we need a filter with amplited T when |W| < Wc. <math> x_{p}(t) ---->Filter, H(w) -----> x(t)</math>
    1 KB (238 words) - 16:44, 9 November 2008
  • To recover, first we need a filter with amplited T when |W| < Wc. <math> x_{p}(t) ---->Filter, H(w) -----> x(t)</math>
    1 KB (238 words) - 05:31, 16 November 2008
  • *<math>\omega_c</math>: Cutoff frequency of a filter (<math>\omega_c > 0</math>). (For instance, lowpass filters are nonzero in
    1 KB (178 words) - 19:31, 23 November 2008
  • <math>\omega_c </math> Cut off frequency for a filter Pretty much apply a low pass filter to <math>x_p(t)</math>
    2 KB (349 words) - 12:09, 10 November 2008
  • X(t) can be recovered exactly from Xp(t) by using a low pass filter with gain T and cut off frequency greater than Wm but less than Ws - Wm.
    1 KB (274 words) - 06:49, 16 September 2013
  • ...alled the reconstructed signal <math>x_r(t)</math>, we must use a low pass filter with a gain of T, and a frequency in the range of <math>(\omega_m,\omega_s- We can use a transfer function to approximate this filter, and it is:
    3 KB (543 words) - 17:23, 10 November 2008
  • ...can be used to recover the modulated signal being a common one the low pas filter,
    1 KB (207 words) - 17:11, 10 November 2008
  • ...(t) can be recovered from its sampled version using an appropriate lowpass filter?
    2 KB (340 words) - 17:29, 10 November 2008
  • ...in reality must of course be truncated usually by a low-pass or band-pass filter.
    548 B (84 words) - 17:56, 10 November 2008
  • Where <math>H(\omega)</math> is a filter with gain equal to the period of the signal and a cutoff frequency of <math
    711 B (130 words) - 19:44, 10 November 2008
  • To demodulate, multiply <math> cosw_{c}t </math> and use low pass filter with gain of 2.
    1 KB (270 words) - 12:35, 16 November 2008
  • To recover x(t), use a low pass filter with gain <math> \frac {1}{a_0}=\frac {T}{\delta} </math>, and cut-off freq
    949 B (174 words) - 16:11, 16 November 2008
  • ...m <math>x(t)cos^2(\omega_c t)</math>, we run the signal through a low pass filter with a gain of 2 and cut off frequency <math>\omega</math>. <math>\omega</m
    2 KB (356 words) - 08:49, 17 November 2008
  • <math>\omega_c</math>: Band frequency for the lowpass filter used to recover the original signal from it's samples. Must be greater tha
    2 KB (279 words) - 12:53, 17 November 2008
  • ...signal one of the signals so that it is centered at zero and use a lowpass filter with a gain of 2 and cutoff frequency greater than <math> \omega_m</math> b
    2 KB (344 words) - 15:55, 30 November 2010
  • <math>x(t)\cos^2(\omega_ct)</math> --> a lowpass filter with a height of 2 and <math>\omega_m<\omega_c<2\omega_c-\omega_m</math> --
    2 KB (336 words) - 17:26, 17 November 2008
  • ...constructed function <math>x_r(t)</math> can be recovered using a low pass filter created from the multiplication of the original step function generator <ma ...lution of <math>h_1(t)</math> and <math>h_2(t)</math> represent a low pass filter with a gain of T and a cutoff frequency <math>\omega_c</math> between <math
    2 KB (411 words) - 17:16, 17 November 2008
  • ...by modulating y(t) with the same sinusoidal carrier and applying a lowpass filter to the result. Applying the lowpass filter to w(t) corresponds to retaining the first term, <math>\frac{1}{2}x(t)</mat
    837 B (153 words) - 19:06, 17 November 2008
  • ...er than Wm and less than Ws - Wm. We process this signal through a lowpass filter and receive the output signal that is exactly x(t).
    805 B (160 words) - 20:06, 17 November 2008
  • ...mple values. This impulse train is then processed through an ideal lowpass filter with gain T and cutoff frequency greater than \omega_M and les than \omega_
    21 KB (3,312 words) - 11:58, 5 December 2008
  • ...mple values. This impulse train is then processed through an ideal lowpass filter with gain T and cutoff frequency greater than \omega_M and les than \omega_
    2 KB (254 words) - 07:05, 8 December 2008
  • ...to the central limit theorem this smoothing can be approximated by several filter steps that can be computed much faster, like the simple moving average.
    10 KB (1,594 words) - 11:41, 24 March 2008
  • ...eed a 2D convolution function. So you can take the 2d fft multiply by your filter and invert. However this seems like allot of work, and I have never implime ...hat way. The example file gives you an idea of how to do this for a simple filter. Good luck!
    10 KB (1,738 words) - 22:44, 7 April 2008
  • *Week 7-8: Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering) ***Prof. Bouman's lecture notes on digital Filter design: [https://engineering.purdue.edu/~bouman/ece438/lecture/module_1/1.7
    9 KB (1,237 words) - 09:29, 5 October 2009
  • *What did you do to apply the lowpass filter? -- [[User:weim|weim]] * firpm(500,... generates a 500 order Parks-Mc Clellen filter of cutoff. MATLAB uses 1 as pi and 1e-12 and 2e-12 are the cutoff freque
    4 KB (543 words) - 07:02, 25 August 2010
  • See also deconv, conv2, convn, filter and, in the signal
    1 KB (204 words) - 22:28, 1 July 2009
  • This can be accomplished either by a band-pass filter OR using the following system.
    538 B (101 words) - 17:58, 29 July 2009
  • ...y modulating y(t) with the same sinusoidal carrier and applying a low pass filter to the
    629 B (112 words) - 18:18, 29 July 2009
  • ...ck market! That's right. It is possible, from simple spectral analysis to "filter" out long-term and short term trends from any time varying data. This inclu * How? LOW PASS FILTER
    7 KB (1,251 words) - 11:54, 21 September 2012
  • ...ess the answer. Am I supposed to take the inverse transform of the digital filter and relate it to time? My final answer would be composed of a lot of rects
    4 KB (628 words) - 15:47, 30 November 2010
  • After Step 3, the signal is ready to be put through a discrete filter. ...s the maximum frequency of the signal equal to the cutoff frequency of the filter and will allow us to determine a sampling frequency that will satisfy the N
    8 KB (1,452 words) - 06:49, 16 September 2013
  • ...y research interests include Signal Processing, (mainly Spectral Analysis, Filter Design and Image Processing), Digital Systems Design and Psychology (sounds
    2 KB (268 words) - 09:43, 14 May 2010
  • ...t this can be applied is in signal reconstruction, where a low pass analog filter is used on the output of a digital-to-analog converter to attenuate unwante ...ained how up-sampling can be used to relax requirements on analog low pass filter design while decreasing signal distortion.
    5 KB (840 words) - 19:08, 22 September 2009
  • ...act, the signal can be reconstructed back by passing it through a low-pass filter with a cut-off frequency of <math> \frac{1}{2T} </math> and a gain of <math
    3 KB (527 words) - 11:50, 22 September 2009
  • To recover the signal, we will require a low pass filter with gain '''<math>T\,\!</math>''' and cutoff, '''<math>\frac{1}{2T}</math>
    3 KB (484 words) - 09:47, 23 September 2009
  • ...ecover the signal, the sampled function is simply multiplied by a low-pass filter with height equal to the sampling period T, <math>{H_r}(f)</math>, to isola ...age:ReconstructFilter.png|500px|thumb|center|Sampled function with lowpass filter]]
    2 KB (436 words) - 19:51, 22 September 2009
  • ...D in the circle is how many zeros we fill between samples and the Low Pass Filter removes the extraneous copies of the signal beyond W/D shown in the output *So, intuitively, we want to use an LPF (low pass filter) to "remove" these high frequency spikes.
    5 KB (847 words) - 11:54, 21 September 2012
  • If we imagine that our slit aperture can be thought of as a spatial filter that only passes light through the slit and blocks light everywhere else, t
    2 KB (286 words) - 10:19, 23 September 2009
  • *To filter this out, we can apply a low-pass filter with a cutoff frequency of around 2000 Hz. *So adding a high pass filter should remove the bear's voice completely.
    5 KB (822 words) - 11:54, 21 September 2012
  • Two basic filters used are 1> '''Average Filter''' == '''Average Filter''' ==
    3 KB (426 words) - 06:03, 14 October 2009
  • title('Image using Average Filter');
    782 B (107 words) - 06:01, 14 October 2009
  • title('Image using Average Filter');
    787 B (108 words) - 06:06, 14 October 2009
  • ...er:mboutin|Prof. Boutin]]: graph of the magnitude of the DFT of a windowed filter= Consider the ideal low-pass filter
    1 KB (212 words) - 11:50, 24 October 2011
  • The end result after modeling is a transfer function that is an all-pole filter with a gain and a time delay. As noted above, the transfer function is usually an all-pole filter. We can observe what the resonances, or formants, are just by looking at t
    5 KB (841 words) - 15:26, 10 April 2013
  • periodic filter phoneme - Generally, the vocal tract transfer function is an all-pole filter
    2 KB (390 words) - 07:46, 14 November 2011
  • ...the series for tons of translations and dilations, uses a faster approach: filter banks. ==Multi-Resoltion Analysis using Filter Banks==
    10 KB (1,646 words) - 11:26, 18 March 2013
  • The following pictures show the original image (Lena), the image of the filter, and the filtered image (done with conv2). Note that the FFT2 are plotted a ...e filter = 1/16*[1 2 1; 2 4 2; 1 2 1], commonly referred to as an "average filter":'''</u>
    8 KB (1,397 words) - 11:23, 18 March 2013
  • periodic filter phoneme - Generally, the vocal tract transfer function is an all-pole filter
    2 KB (387 words) - 07:47, 14 November 2011
  • Plot of the frequency response of the average filter: Plot of the frequency response of the filter:
    1 KB (163 words) - 12:50, 26 November 2014
  • ...g how to handle the boundaries is as important as knowing how to apply the filter to the image (at least in the context of this course). -[[User:crtaylor|Ry
    945 B (155 words) - 17:55, 8 December 2009
  • ...of some non-linear filters for face-enhancement, such as the Perona-Malik filter that preserve sharpness and details, whilst removing noise at the same time
    235 B (33 words) - 22:21, 6 December 2009
  • ...though I got good grade on ECE301. And the second part is basically about filter. Before you take ECE438, I also recommend that you should figure out how to
    17 KB (3,004 words) - 08:11, 15 December 2011
  • h(x,y) → filter FREE → filter image based only on pixels: {H, K, P}
    5 KB (811 words) - 16:19, 19 December 2009
  • ...ion evolves into a great estimation method. The more I know about particle filter for object tracking, the more I get impressions about power of random sampl
    6 KB (884 words) - 16:26, 9 May 2010
  • ...al ( this is of course easier said than done because we then have to use a filter of the appropriate length and so on, which means another two pages of math) ...h cut off frequencies that represented those weekly variations. A low pass filter is one that passes low frequency signals but attenuates signals that have f
    13 KB (2,348 words) - 13:25, 2 December 2011
  • **[[Practice_Question_5_ECE438F10|Practice Question 5 (filter design)]]
    9 KB (1,221 words) - 11:00, 22 December 2014
  • *Week 7-8: Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering) ***Prof. Bouman's lecture notes on digital Filter design: [https://engineering.purdue.edu/~bouman/ece438/lecture/module_1/1.7
    9 KB (1,331 words) - 07:15, 29 December 2010
  • ...system is an all pole filter cascaded with a time delay. The poles of this filter determine the location of the local maxima of the voiced phonemes we pronou
    1 KB (151 words) - 12:53, 8 November 2010
  • ...o get rid of aliases, what is the cutoff frequency of digital LPF(Low-Pass Filter)?}\,\!</math>
    2 KB (315 words) - 10:39, 11 November 2011
  • ...ore, the cut-off frequency, <math>f_c</math> of the discrete-time low-pass filter (LPF) is <math>f_c=\frac{\pi}{L}</math>, in general.
    3 KB (467 words) - 19:52, 20 September 2010
  • ...s relationship, we concluded that, under certain circumstances, a low-pass filter could be applied to this upsampling so to obtain the signal
    1 KB (220 words) - 16:07, 22 September 2010
  • ...rder to get rid of aliasing, what is the cut-off frequency of the low pass filter? Explain your answer. Assume that the input signal X(f) and the continuous time filter H(f) are both band limited to 1/(2T).
    2 KB (373 words) - 10:41, 11 November 2011
  • Thus, the cut-off frequency of the LP filter is <math>\frac{\pi}{L}</math>.
    666 B (121 words) - 12:21, 29 September 2010
  • ...it affects the reasoning here). And finally sending it through a low pass filter, the "extra" rects get filtered out so when you end up with non-zero freque
    5 KB (778 words) - 09:11, 1 October 2010
  • ...plies that the reconstructed signal <math>x_r(t)</math> is the output of a filter when we input the impulse train of <math>x(t)</math> with period <math>T</m ...math>\text{sinc}(t/T)</math>, whose frequency response is a ideal low-pass filter with the cut-off frequency of <math>1/(2T)</math>.
    4 KB (751 words) - 04:56, 2 October 2011
  • Q2. Suppose that the LTI filter <math>h_1</math> satifies the following difference equation between input < Then, find the inverse LTI filter <math>h_2</math> of <math>h_1</math>, which satisfies the following relatio
    3 KB (462 words) - 10:42, 11 November 2011
  • ...aracteristics (Filter A) and another filer with band-pass characteristics (Filter B). The behavior of these two filters will be further studied when we retur
    867 B (122 words) - 16:21, 8 October 2010
  • ...at a specific low-pass filter (filter A) and a specific band-pass filter (filter B). We noticed the two different ways of writing the transfer function (as
    2 KB (393 words) - 07:25, 25 October 2010
  • equation (moving average filter) :a. Find the impulse response h[n] for this filter. Is it of finite or infinite duration?
    4 KB (661 words) - 11:22, 30 October 2011
  • Q1. Consider the following second order FIR filter with the two zeros on the unit circle as shown below. The transfer function for this filter is given by <math> H(z) = (1-e^{j\theta}z^{-1})(1-e^{-j\theta}z^{-1})=1-2\c
    3 KB (480 words) - 10:42, 11 November 2011
  • d. If we further look at the frequency response of this filter, :therefore, when <math>\theta=\pi/2</math>, it is a bandstop filter.
    2 KB (437 words) - 12:00, 19 October 2010
  • Today in the lecture, we continued talking about filters and filter design using the transfer function. It seems like many students find it dif
    1 KB (174 words) - 03:53, 21 October 2010
  • ...y with period <math>2 \pi</math>. It is important to remember this for any filter in discrete-time. In the last part of the lecture, we saw how the heat equa *[[ECE_301_Fall_2007_mboutin_Filter_Types|Ideal filter types in continuous-time]]: do not forget to repeat periodically every <ma
    2 KB (260 words) - 12:42, 22 October 2010
  • Topic: Filter Design Define a two-pole band-pass filter such that
    2 KB (322 words) - 13:00, 26 November 2013
  • d. Determineif the filter represented by the difference equation is FIR or IIR. Give reasons for your
    3 KB (479 words) - 10:42, 11 November 2011
  • d. Filter represented by this difference equation is IIR. Because the transfer functi
    2 KB (441 words) - 05:42, 28 October 2010
  • Q2. Consider a causal FIR filter of length M = 2 with impulse response Q5. Define a two-zero band-stop filter such that
    3 KB (462 words) - 10:42, 11 November 2011
  • Suppose the transfer function of the filter has the form Where <math>z_1,z_2</math> are zeros of the filter.
    2 KB (279 words) - 17:23, 3 November 2010
  • Q1. Consider a causal FIR filter of length M = 2 with impulse response
    3 KB (561 words) - 10:43, 11 November 2011
  • Q4. Consider a 3X3 FIR filter with coefficients h[m,n] <br/> a. Find a difference equation that can be used to implement this filter.<br/>
    3 KB (398 words) - 10:43, 11 November 2011
  • ...that <math class="inline">h\left(t\right)</math> acts as a crude low-pass filter that attenuates high-frequency power.
    3 KB (498 words) - 07:16, 1 December 2010
  • d. Describe how the filter behaves when <math>\lambda</math> is positive and large. <br/> e. Describe how the filter behaves when <math>\lambda</math> is negative and bigger than -1. <br/>
    3 KB (515 words) - 10:43, 11 November 2011
  • d. For large values of <math>\lambda</math>, the filter performs sharpening.<br/> e. For -1 < <math>\lambda</math> < 0, the filter performs blurring.<br/>
    2 KB (275 words) - 13:34, 28 November 2010
  • ...quency domain perspective. We looked an an example in detail (the low-pass filter illlustrated on top of [[ECE_438_Fall_2009_mboutin_plotCSFTofbasicfilters|t
    808 B (107 words) - 10:56, 29 November 2010
  • ...ral input signal <math>x[m,n]</math> we get the difference equation of the filter. b. Place the center of filter (i.e. where m=0,n=0) upon the pixel of image. Multiply h[m,n] with x[m,n] o
    2 KB (391 words) - 07:16, 30 November 2010
  • ...lting in huge disturbance to the game, I feel strong obligated to design a filter to remove the noise. Each filter is used to remove corresponding frequency component.
    3 KB (409 words) - 08:53, 11 November 2013
  • ...ms]] from [[2011 Spring ECE 301 Boutin|ECE301]]!) are utilized to convert, filter, and combine these signals and produce the images used in diagnostics.
    17 KB (2,368 words) - 10:53, 6 May 2012
  • ...y that the Fourier transform is a non-zero constant multiple of a low-pass filter with gain 1 and cutoff <math>3 \pi</math> and conclude from there; you woul
    3 KB (431 words) - 10:28, 11 November 2011
  • ...'c''</sub>''t'').</span> Then feed the resulting signal through a low pass filter with a gain of 2 and a cutoff frequency of <span class="texhtml">ω<sub>''c b) Multiply by cos(w<sub>c</sub>t) then pass it through a Low Pass Filter with a gain of 2 and a cutoff f of w<sub>c</sub>
    2 KB (400 words) - 10:31, 11 November 2011
  • ...ignal can be recovered by filtering the sampled signal using the following filter: ...DT, one must first convert the samples to a pulse-train, and then low-pass filter. -pm </font>
    9 KB (1,462 words) - 07:01, 22 April 2011
  • ...</sub> = 1000π</span> and gain 2. The frequency response of this low pass filter is: Note that the cut-off frequency of the low pass filter can actually be anywhere between <span class="texhtml">ω<sub>''M''</sub></
    12 KB (2,109 words) - 05:58, 22 April 2011
  • **[[Practice_Question_5_ECE438F10|Practice Question 5 (filter design)]] ....m.zip zpgui3.m] A MATLAB GUI showing the effect of poles and zeros during filter design.
    10 KB (1,359 words) - 03:50, 31 August 2013
  • *[[ECE438_Week9_Quiz|LTI system and filter design]] <br/>
    900 B (121 words) - 10:39, 11 November 2011
  • *Week (7)-8: Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering) ***Prof. Bouman's lecture notes on digital Filter design: [https://engineering.purdue.edu/~bouman/ece438/lecture/module_1/1.7
    9 KB (1,341 words) - 03:52, 31 August 2013
  • =Image processing on an Android phone - Lowpass filter an Image - C++ code= ...ore used to. The filtering itself is not really complicated: We just use a filter in form of a matrix and perform a convolution. As this is not really an iss
    3 KB (486 words) - 11:58, 20 April 2012
  • ...revious lecture, we observed that, under certain circumstances, a low-pass filter could be applied to this upsampling so to obtain the signal
    1 KB (213 words) - 06:24, 11 September 2013
  • ...this by computing the frequency response and the transfer function of that filter. In particular, we noted how the location of the poles and the zeros of the [[Category:Filter]]
    998 B (143 words) - 06:27, 11 September 2013
  • **[[Practice_Question_5_ECE438F10|Practice Question 5 (filter design)]]
    9 KB (1,273 words) - 20:52, 15 October 2011
  • ...n the location of the poles and the zeros of the transfer function of this filter and the amplitude of its frequency response. [[Category:Filter]]
    953 B (132 words) - 06:27, 11 September 2013
  • equation (moving average filter) :a. Find the impulse response h[n] for this filter. Is it of finite or infinite duration?
    5 KB (916 words) - 03:56, 31 August 2013
  • Today we discussed the relevance of "filter design" in today's world, including some open problems for which research i [[Category:Filter]]
    1 KB (221 words) - 06:28, 11 September 2013
  • ...e unit impulse response of an ideal filter in order to obtain a causal FIR filter. A MATLAB plot of the example presented in class can be viewed on [[ECE_438
    1 KB (164 words) - 06:30, 11 September 2013
  • ::#low pass filter the repeated part
    7 KB (1,108 words) - 06:02, 23 September 2014
  • A LPF (low-pass-filter) will usually be used before down-sampling to reduce aliasing. In this&nbsp ...f 4, 8, and 16. In this project we are using FIR (finite impulse response) filter.<br>The audio signal we use is part of Waving Flag, the theme song of 2010
    10 KB (1,707 words) - 10:44, 6 May 2012
  • *Filter design **[[Practice_Question_5_ECE438F10|Practice question on filter design]]
    6 KB (801 words) - 22:04, 19 April 2015
  • ...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
    2 KB (301 words) - 06:32, 11 September 2013
  • ...sample by factor of 5 then down sample by factor 3. To avoid aliasing, the filter was build by MATLAB embedded function "fir1" with order= 20, cut-of frequen
    2 KB (389 words) - 06:37, 25 September 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 filter
    2 KB (213 words) - 06:32, 11 September 2013
  • Consider the following FIR filter: a) Write a difference equation that can be used to implement this filter.
    2 KB (270 words) - 03:59, 31 August 2013
  • ...other type of filter introduced by Perona and Malik. We observed that this filter is not linear and that it allows one to smooth out an image without blurrin
    1 KB (157 words) - 06:33, 11 September 2013
  • Therefore the filter can be separate into two 1-D filters.
    3 KB (355 words) - 13:42, 4 December 2011
  • =Image processing on an Android phone - Lowpass filter an Image - Java code= ...roject was to lowpass filter an image on an Android phone using a Gaussian filter. Therefore it should be possible to either take a picture with the integrat
    7 KB (1,278 words) - 11:57, 20 April 2012
  • ...tion, signal mixing, encoding and decoding of audio signals, and real-time filter implementation. Additionally, I was involved with the development of a filt
    4 KB (676 words) - 12:21, 9 February 2012
  • ...ng a proof of the visit to a doctor with a date and time on it, so it will filter out such students who use this excuse frequently and know all the details h
    6 KB (1,023 words) - 09:24, 16 March 2012
  • ...cy filtering removes noise, but also blurs images as a result. The unsharp filter accentuates the edges of images, in an emboss like feature. The parts of th
    1 KB (196 words) - 17:45, 21 April 2013
  • so filter out them. /* Filter in daily returns on earning reports dates */
    11 KB (1,577 words) - 08:35, 23 April 2012
  • 3. \text{ Multiply step 2 by the filter } H(\rho) = |\rho| = f_c \left [ rect(\frac{f}{2f_c}) - \Lambda(\frac{f}{f_ 2. \text{ Filter the projections } \rho_{\theta}(r) \text{ with } h(r) \text{, where } H(\rh
    17 KB (2,783 words) - 01:51, 31 March 2015
  • ...t from probabilistic noise-reducing filters such as the Bayesian or Kalman filter.
    8 KB (1,176 words) - 15:15, 1 May 2016
  • ** Low Pass Filter: Smoothing (less sharp edges or details but reduces some static noise) ** High Pass Filter: Sharping (clear edge and enhance details but also emphasize noise)
    3 KB (555 words) - 08:09, 9 April 2013
  • b) Create Gaussian filter of size 5x5 with mean 0 and standard deviation 3. c) Plot Fourier Transform of filter’s impulse response in 3D.
    4 KB (573 words) - 10:15, 15 May 2013
  • ...l has probability of {1/10,2/10,4/10,2/10,1/10}. The signal goes through a filter, Z=2X^2+1. Z: output after the filter<br>
    2 KB (299 words) - 18:13, 27 February 2013
  • filter <math>h(m,n)</math> is a <math>(2N+1)\times(2N+1)</math> filter, and for each location we need 2 multiplies, so in total, we need <math>2(2 ...ated offline, if we consider that <math> a_j b_i </math> are merged in the filter <math> h(m,n)</math>, then will need <math> (2N+1)^2 </math> multiplies to
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  • ''b) Create Gaussian filter of size 5x5 with mean 0 and standard deviation 3.'' ''c) Plot Fourier Transform of filter’s impulse response in 3D.''
    2 KB (348 words) - 10:50, 11 March 2013
  • The upper is the Gaussian filter, while bottom is the unsharp.
    1 KB (174 words) - 11:34, 11 March 2013
  • ...eed a 2D convolution function. So you can take the 2d fft multiply by your filter and invert. However this seems like allot of work, and I have never implime ...hat way. The example file gives you an idea of how to do this for a simple filter. Good luck!
    10 KB (1,756 words) - 08:05, 9 April 2013
  • ...this process. These artifacts can be reduced by filtering with a high pass filter prior to back-projection. For this reason, the process is also commonly ref # Filter the projections to obtain <math>g_{\theta}(r) = h(r)*p_{\theta}(r)</math>.
    9 KB (1,486 words) - 07:25, 26 February 2014
  • ...function is basically the frequency response of an ideal digital low pass filter. So if you were to build a low pass filter, its impulse response would be a sampled sinc in time as shown in figure 6.
    10 KB (1,726 words) - 07:26, 26 February 2014
  • ....m.zip zpgui3.m] A MATLAB GUI showing the effect of poles and zeros during filter design.
    8 KB (1,096 words) - 06:44, 14 December 2013
  • *Week (7)-8: Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering) ***Prof. Bouman's lecture notes on digital Filter design: [https://engineering.purdue.edu/~bouman/ece438/lecture/module_1/1.7
    9 KB (1,353 words) - 09:04, 11 November 2013
  • ...nd last questions in part 2 of this lab, does anybody know they the RLC BP filter isn't used for the two-tone test, or why the '''in-band''' third-order inte ...in frequency) to the input signal. Thus, it would be difficult to design a filter to separate the input frequencies from these in-band distortion terms.
    1 KB (204 words) - 10:09, 15 January 2014
  • ...gnal in the frequency domain first. (Recall that you just need to low-pass-filter the ideal sampling.) Then invert the Fourier transform to get the reconstru
    2 KB (362 words) - 13:59, 26 September 2013
  • ...(t) was band-limited with <math>f_{max}<\pi /D</math> and apply a low-pass filter with gain D and cut-off <math>\pi/D</math> to <math>x_2[n]</math>.
    2 KB (335 words) - 05:55, 27 September 2013
  • 1. Use the signal generators and filters in the lab to generate and filter noise and various types of periodic signals. ...rfectly band-limited to 0-10MHz, is passed through a perfectly rectangular filter of bandwidth 18 kHz, gain 3 dB, and center frequency 455 kHz. If the RMS vo
    14 KB (2,228 words) - 12:03, 15 January 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 (480 words) - 09:13, 27 September 2013
  • We need a high pass filter that filters our everything below 60 Hz. ...equency component. In order to remove the annual cycle, we need a low pass filter.
    6 KB (1,018 words) - 12:18, 30 September 2013
  • ...izing a few important facts about LTI systems, we defined a first filter, "Filter A", which we found had low-pass characteristics.
    2 KB (294 words) - 05:58, 14 October 2013
  • ...ssion on the topic of filtering, we defined another simple filter, called "filter B", and analysed its properties using the concept of frequency response and
    2 KB (300 words) - 05:47, 16 October 2013
  • 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_1/1.7
    6 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 describ
    1 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 filter
    3 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 discussed
    2 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 10
    13 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)
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  • ...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 is
    7 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 H
    10 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 see
    3 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}</math
    2 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 \tex
    6 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&nbsp; prevent aliasing.&nbsp; LPF eliminates signal components which has
    7 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 the
    4 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 e
    3 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 ha
    6 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}</m
    3 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 t
    3 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 co
    2 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 as
    5 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 fun
    2 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 describ
    6 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
  • *Week 8: [[Media:lab7aECE438F15.pdf| Lab 7a - Digital Filter Design]] *Week 9: [[Media:lab7bECE438F15.pdf| Lab 7b - Digital Filter Design]]
    2 KB (303 words) - 13:45, 5 October 2015
  • *** [[Media:lab7aECE438F15.pdf| Lab 7a - First Lab on Digital Filter Design]] *Week 10-(11): Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering)
    10 KB (1,356 words) - 13:19, 19 October 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 (384 words) - 11:44, 21 April 2015
  • ...t its CTFT X(f) is zero when when |f|>1,400 Hz. You would like to low-pass-filter the signal x(t) with a cut off frequency of 800Hz and a gain of 7. Let's ca ...you to describe the digital filter that is equivalent to the given analog filter. In other words, how can you process the signal in the discrete-time domain
    3 KB (499 words) - 16:04, 22 September 2015
  • where the frequency response H(f) corresponds to a band-pass filter with no gain and cutoff frequencies f1=200Hz and f2=600Hz.
    5 KB (779 words) - 18:19, 25 September 2015
  • INSTRUCTOR'S NOTE: THERE IS A MISTAKE BELOW. THE AMPLITUDE OF THE DT FILTER SHOULD NOT BE MULTIPLIED BY 1/TS. -> Corrected! ...t its CTFT X(f) is zero when when |f|>1,400 Hz. You would like to low-pass-filter the signal x(t) with a cut off frequency of 800Hz and a gain of 7. Let's ca
    3 KB (475 words) - 15:23, 20 October 2015
  • ...spectrum while preserving the one at baseband. Conceptually, the simplest filter is
    7 KB (1,181 words) - 19:17, 19 October 2015
  • where the frequency response H(f) corresponds to a band-pass filter with no gain and cutoff frequencies f1=200Hz and f2=600Hz. Note that the cutoff points for second filter, <math>H_1(\omega)</math>, were found using <math>w_0=(f_0/f_s)2\pi</math>.
    6 KB (945 words) - 11:40, 19 October 2015
  • equation (moving average filter) :a. Find the impulse response h[n] for this filter. Is it of finite or infinite duration?
    4 KB (625 words) - 13:17, 16 November 2015
  • b) Could the vocal tract be modeled using an FIR filter? Explain.
    3 KB (449 words) - 11:39, 20 November 2015
  • Consider the following filter: a) Write a difference equation that can be used to implement this filter.
    4 KB (416 words) - 11:50, 20 November 2015
  • ...o separate the output from the lungs from the vocal tract. The vocal tract filter is then modeled by a series of values. This allows the sound and the freque The impulse train was then combined with the LPC coefficients using Matlab's filter function.
    9 KB (1,777 words) - 23:23, 21 November 2015
  • ...ent <br />of the frequency bin, leading to a unchanging bandwidth for each filter. This means that bins in the higher frequencies have a higher <br /> quali
    4 KB (655 words) - 23:26, 22 November 2015
  • ...y implementing a comb filter. The difference equation for this simple comb filter can be written as follows: ..., which changes periodically. The difference equation for this simple comb filter can be written as follows:
    2 KB (362 words) - 22:18, 29 November 2015
  • ...single delay will achieve this effect. The difference equation for the FIR filter can be written as follows [3] : ...ay D, which changes periodically. The difference equation for this flanger filter can be written as follows:
    3 KB (405 words) - 23:49, 29 November 2015
  • ...order to enhance the look of an image. However, a 3X3 median filter is the filter we are going to use in this operation. The Median filter matlab code is provided below
    2 KB (316 words) - 00:15, 30 November 2015
  • ...order to enhance the look of an image. However, a 3X3 median filter is the filter we are going to use in this operation. The Median filter matlab code is provided below
    2 KB (316 words) - 00:25, 30 November 2015
  • Consider the following filter: a) Write a difference equation that can be used to implement this filter.
    10 KB (1,157 words) - 22:56, 2 December 2015
  • *** [[Media:lab7aECE438F15.pdf| Lab 7a - First Lab on Digital Filter Design]] *Week 10-(11): Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering)
    10 KB (1,357 words) - 17:02, 14 September 2016
  • *Week 9: [[Media:lab7aECE438F15.pdf| Lab 7a - Digital Filter Design]] *Week 10: [[Media:lab7bECE438F15.pdf| Lab 7b - Digital Filter Design]]
    2 KB (301 words) - 14:05, 26 September 2016
  • ...he patient is being monitored. You are going to design a high-pass digital filter to eliminate the 60 Hz interference and everything at frequencies below 60 a) Sketch the CTFT of the (analog) high-pass filter that is needed in this case
    3 KB (460 words) - 09:11, 7 September 2016
  • ...onship between DT signal processing and CT signal processing (for a simple filter) once more, this time with a focus on the effect of changing the sampling f ...at its CTFT X(f) is zero when when |f|>1000 Hz. You would like to low-pass-filter the signal x(t) with a cut off frequency of 900Hz and a gain of 7. Let's ca
    4 KB (636 words) - 08:16, 21 September 2016
  • :d) the system is an FIR filter; :e) The system is an IIR filter;
    3 KB (481 words) - 15:35, 8 November 2016
  • equation (moving average filter) :a. Find the impulse response h[n] for this filter. Is it of finite or infinite duration?
    3 KB (503 words) - 15:44, 8 November 2016
  • * Bandpass Filter ...equency. Based on the diagram, we can determine how to design our bandpass filter.
    8 KB (1,120 words) - 00:27, 26 November 2016
  • d) the system is an FIR filter;<br /> e) The system is an IIR filter;<br />
    8 KB (1,336 words) - 15:40, 27 November 2016
  • b) Could the vocal tract be modeled using an FIR filter? Explain.
    3 KB (460 words) - 13:20, 18 November 2016
  • ...aim that these devices bleed onto other channels and should have a lowpass filter on the output (mine is inline with the coax). There is also a UHF band ava
    11 KB (1,666 words) - 02:18, 30 November 2016
  • 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 describ
    7 KB (1,236 words) - 17:19, 29 November 2016
  • *** [[Media:lab7aECE438F15.pdf| Lab 7a - First Lab on Digital Filter Design]] *Week 10-(11): Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering)
    10 KB (1,357 words) - 09:45, 8 January 2017
  • ...project I have illustrated how edge detection works. I have used Gaussian filter to blur the images to remove some for accurate edge line detection. [[File:Filter math.jpeg|1024px|thumbnail]]
    3 KB (498 words) - 18:53, 30 November 2016
  • ...ge strength as noise is cancelled. This is how the equation of a Laplacian filter looks:
    2 KB (258 words) - 04:20, 28 November 2016
  • [https://drive.google.com/open?id=0B6zfOg44IFTPNDFKVndCeWVsdm8 LAPLACIAN FILTER BLUR DETECTION]
    140 B (21 words) - 05:40, 28 November 2016
  • | [[Digital filter using the clustering method]]
    3 KB (421 words) - 16:18, 10 December 2017
  • | [[ BRITA Filter ]]
    3 KB (448 words) - 23:55, 23 April 2017
  • *Week 9: [[Media:lab7aECE438F15.pdf| Lab 7a - Digital Filter Design]] *Week 11: [[Media:lab7bECE438F15.pdf| Lab 7b - Digital Filter Design]]
    2 KB (325 words) - 18:20, 30 March 2017
  • ...he patient is being monitored. You are going to design a high-pass digital filter to eliminate the 60 Hz interference and everything at frequencies below 60 a) Sketch the CTFT of the (analog) high-pass filter that is needed in this case
    4 KB (658 words) - 14:50, 1 February 2017
  • ...ne the frequency ranges of both the signal and the noise so an appropriate filter could be applied. Spectrograms for each section were created with the MATLA ...d a clearer result. This filter was implemented using MATLAB's butterworth filter command.
    6 KB (984 words) - 21:39, 23 April 2017
  • ...f differently (or symmetrically) weighted indexes. This will represent the filter that we will be implementing for an edge detection. ...user wanted to exaggerate the edge, then the user would need to change the filter values of -2 and 2 to higher magnitude. Perhaps -5 and 5. This would make t
    7 KB (1,184 words) - 20:31, 23 April 2017
  • ===BRITA Filter: Audio Filtration and FFT Visualizer=== ...ss, HPF, LPF, etc) with different Q and cutoff values to achieve the exact filter response desired. Very simple filters, such as the HPF and LPF discussed in
    6 KB (1,048 words) - 16:58, 24 April 2017
  • a. Yes! This can be accomplished by passing the signal through a low-pass filter with a gain of 5 and a cutoff at <math> \frac {2 \pi 2000} {9000} </math>
    7 KB (1,194 words) - 19:21, 24 April 2017
  • *Week 9: [[Media:lab7aECE438F15.pdf| Lab 7a - Digital Filter Design]] *Week 10: [[Media:lab7bECE438F15.pdf| Lab 7b - Digital Filter Design]]
    3 KB (418 words) - 21:57, 7 January 2018
  • Take the high pass filter y[n] = 1/2*(x[n]-x[n-1]) as an example. Figure2: Fourier Transform of a high pass filter
    2 KB (310 words) - 21:40, 30 November 2017
  • Take the high pass filter y[n] = 1/2*(x[n]-x[n-1]) as an example. Figure2: Fourier Transform of a high pass filter
    2 KB (310 words) - 21:44, 30 November 2017
  • ...ference point of 4.5V with 0V, and I assumed that the frequency responses' filter characteristics should remain the same. ...n offers a Low-Pass Filter while the rightmost position offers a High-Pass Filter, which fits the User's Manual description of "Max Bass" and "Max Treble."
    5 KB (752 words) - 16:44, 2 December 2017
  • ...ference point of 4.5V with 0V, and I assumed that the frequency responses' filter characteristics should remain the same. ...n offers a Low-Pass Filter while the rightmost position offers a High-Pass Filter, which fits the User's Manual description of "Max Bass" and "Max Treble."
    3 KB (499 words) - 03:53, 2 December 2017
  • | FIR Filter || ROC of transfer function is the whole complex plane except z=0 and/or z=
    967 B (143 words) - 13:22, 3 December 2017
  • ...and design in addition to how concepts learned in ECE 438 apply to actual filter design. ...The step response is the integrated impulse response and will show how the filter will affect the signal in the time-domain. On the other hand, the frequency
    6 KB (897 words) - 16:44, 7 December 2017
  • *Week 6: [[Media:lab7aECE438F15.pdf| Digital Filter Design 1]]
    3 KB (375 words) - 13:15, 25 January 2018
  • ...rain does not overlap then the signal can still be recovered by applying a filter.
    2 KB (332 words) - 03:22, 23 November 2018
  • ...des the audio signal into smaller pieces, these are called frames. An MDCT filter is then performed on the output. ...1024-point FFT, and then the psychoacoustic model is applied. Another MDCT filter is performed on the output.
    5 KB (752 words) - 17:40, 2 December 2018
  • ...mind that when there is a specific frequency channel specified, a Bandpass filter must be added at the end of the pulse train example in order to get the pro ...mind that when there is a specific frequency channel specified, a Bandpass filter must be added in before multiplying by the carrier.
    2 KB (405 words) - 23:42, 2 December 2018
  • ...find out which frequency is needed, you can use the frequency obtained to filter out other frequencies that creates noise.
    3 KB (555 words) - 22:02, 2 December 2018
  • ...ution of the causal sinewave of length N in Equation (1) with a causal FIR filter of length L, where L < N.<br/>
    8 KB (1,474 words) - 16:37, 24 February 2019
  • ...ters, <math>f_0[n]</math> and <math>f_1[n]</math>,form a Quadrature Mirror Filter (QMF). Specially, <br> The DTFT of the halfband filter <math>h_0[n]</math> above may be expressed as follows:<br>
    4 KB (738 words) - 15:34, 19 February 2019
  • e) Describe what ths filter does and how the output changes as <math>\lambda</math> increases.<br>
    2 KB (338 words) - 16:48, 19 February 2019
  • This is a sharpen filter. The image will become more sharpen as <math>\lambda</math> increases.
    2 KB (282 words) - 11:58, 25 February 2019
  • ...come hardly visible. The goal of this mini-project was to apply a low pass filter learnt in class and observe its effectiveness in removing the excess chalk ...r (and popular) filters to implement at this scale. Secondly, the low-pass filter is versatile enough to extend applications to any writing on a background,
    7 KB (1,006 words) - 19:22, 7 April 2019
  • ...spike), we can try to apply something we already know about - a band-pass filter! To do this, we are going to go to '''Effects -> Filter and EQ -> FFT Filter...'''. This will display a simpler Fourier Transform of our signal at a spe
    6 KB (1,077 words) - 20:29, 7 April 2019
  • In both cases, we must first low pass filter each audio file
    1 KB (185 words) - 17:30, 25 April 2019
  • | [[ece438f19achakrabortybonus|Finding the Optimal Filter Size for Median Filtering]] | [[ece438f19mvermabonus|Image Convolution with Different Filter Applications]]
    4 KB (467 words) - 02:18, 10 December 2019
  • *Week 9: [[Media:lab7aECE438F15.pdf| Lab 7a - Digital Filter Design]] - [[Media:lab7agradingrubric.pdf|Lab7a cover sheet with grading ru *Week 10: [[Media:lab7bECE438F19new.pdf| Lab 7b - Digital Filter Design]] - [[Media:lab7bgradingrubric.pdf|Lab7b cover sheet with grading ru
    4 KB (572 words) - 11:16, 3 December 2019
  • *** [[Media:lab7aECE438F15.pdf| Lab 7a - First Lab on Digital Filter Design]] *Week 10-(11): Filtering (Systems defined by Difference equations, Filter Design, DFT view of Filtering)
    10 KB (1,356 words) - 18:52, 20 August 2019
  • ...cy content from the original signal by first passing it through a low pass filter to ensure the final downsampled signal satisfies the Nyquist criterion. See <math>x_1[n]</math> → Low pass filter with cutoff <math>\frac{𝜋}{D}</math> and gain 1 → Downsample by factor
    16 KB (2,611 words) - 14:11, 12 November 2019
  • The Karplus-Strong algorithm uses a noise burst as an initial condition, a filter of the designer's choice, and a positive feedback loop to produce a signal ...s at the bottom of the article show what happens to the output when an IIR filter with poles near an A string's harmonic frequencies is chosen.
    3 KB (546 words) - 23:43, 1 December 2019
  • ...MATLAB to remove the frequencies above 11 kHz. Next, we design a high-pass filter to remove the frequencies below 7.5 kHz. See the MATLAB code and listen to [[File:batnoisesmatlab.PNG|unframed|MATLAB code used to filter recording of nighttime noises to hear bat noises more clearly]]
    2 KB (280 words) - 23:27, 1 December 2019
  • <big>'''Finding the Optimal Filter Size for Median Filtering'''</big> ...contained salt and pepper noise. This page will demonstrate how the median filter size affects the filtered image in terms of-<br />
    6 KB (1,000 words) - 23:15, 30 November 2019
  • <big>'''Image Convolution with Different Filter Applications'''</big> ...ically then applied to the image matrix. We add the product of the flipped filter matrix together to get the value of that pixel of the filtered image. This
    4 KB (624 words) - 09:11, 6 December 2019
  • ...We can therefore model the vocal tract as a series of uniform tubes that filter the signal.
    5 KB (849 words) - 02:10, 10 December 2019
  • ...ddition, the transform can detect unwanted periodic patterns in images and filter them out.<br /><br />
    12 KB (2,051 words) - 14:20, 5 December 2020
  • ...g at a decreasing rate (or vice-versa). This process is called the Laplace filter in image processing. The original image, before application of the Laplace filter.
    2 KB (392 words) - 23:11, 6 December 2020
  • ...//docs.opencv.org/3.4/d5/db5/tutorial_laplace_operator.html OpenCV Laplace Filter Image Processing]
    2 KB (242 words) - 00:36, 7 December 2020
  • ...ented with a variety of anti-aliasing tools, such as low-pass filters that filter out high frequencies.
    3 KB (444 words) - 23:32, 6 December 2020

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Alumni Liaison

BSEE 2004, current Ph.D. student researching signal and image processing.

Landis Huffman