This is excellent sir!!! I am taking a filters course at the undergraduate level EE, and my professor is really good, but this is a nice supplement and summary of the differences between FIR and IIR.
Actually it is ok as written. Note that the sum in the denominator starts from k=0, and we define a_0 = 1. It would be equivalent to write "1 + sum" where the sum starts at k = 1.
Barry Van Veen ı have a question about fır filter can you help me about it? gokhan.arslan.763@gmail.com you can send mail to me and ı will send question to you on gmail. Thank you so much
This is a good video. I think that it would have been useful to say and show a diagram that in order for any filter to be able to recognize a band of frequencies, then the SHAPE of the impulse function of the filter must have a kernel containing the signal to be selected. * A Low pass filters would have an decaying impulse function which contains the SHAPE related to frequencies to be selected * A high pass filter would have an IMPULSE followed by the inverted shape of the impulse of a low pass filter. * A bandpass filter would have a decaying impulse that encloses an oscillatory signal related to the frequency to be selected. Then the FIR and the IIR Filters would have provided this shape which is shown on flat paper as a flat function but in reality, it is a three-dimensional rotating function going around a time axis After all the signal e^-jwt or e^jwT may be looked at as time locations or rotating vectors are in the form of cos ( wt) + j.sin( wt) which is a rotating vector that operates in the frequency and time domain. It would be a remarkable digital filter if it is shown in three dimensions where the pulses rotate and the delay and adders operate in three dimensions. It is also useful to look upon the Laplace and Convolution function as being three dimensional as we deal with e^jwt and not only the flat functions sine(wt) and cosine ( wt)
Never understood the difference between these two. Just 1 min into the video, my doubt is cleared and i realized i have already used them by name of moving average and software filter.
Thank You so much for this and for al people visiting this, do check out the link that shows up in the video..Really neat explanation of complex topics..Thank You from India Sir...
thanks for the video! I just have one question: at 1:50 you say that FIR filters have poles at z=0. I don't really understand why. Isn't a pole a value that makes the denominator 0? And since here the denominator is 1...
The system function (z-transform of impulse response) for a causal FIR filter is a polynomial in z^{-1}. For example, a simple one is H(z) = 0.5 - 0.5z^{-1}. This is equivalent to H(z) = (0.5z - 0.5)/z - in this form you clearly see the pole at z=0. Note that in the first form with z^{-1} when z = 0, the z^{-1} blows up.
Barry Van Veen So when do we use FIR and IIR? It seems that they both cannot be used for the same problem. When we say filters, I take it to mean band pass band stop notch filters, does it mean something more. How do I use this concept of FIR and IIR????
Had to deal with this in 1997, with limited internet and very few books left in the Uni library. Between this and working fuzzy logic out, I'm clearly smarter than I thought at the time. Since then, I'm clearly not. :D
Interesting video. I had trouble making low pass filter (analog) and high pass filter, also analog. This may solve many troubles in electronic design. I need that for SSB (Single Side Band) generation utilizing 'Weaver method'. Yesterday tried to get that, and ended using three inductors for LPS, plus one for HPS, but the problem is that result is not that great. I got -60 dB attenuation at 3 kHz with LPS, but not that great for HPS for audio filtering before IQ mixers. Just... I am newbie in DSP, and don't know what is what in terms of varialbes (k=0?, M, n, and other things). Just wanted to implement that formula on my STM32, but this concept is not clear, or I am not yet familiar with that math. Do you have any link that may help solve this beginners nightmare? Thank you in advance.
Would've been nice if you briefly went over what all the different letters stand for in your equations. Not everyone learns the mathematical notation of filters at their university, you know :)
Oops sorry i didn't notice that the limits for y[n] in the eq. above H(z) were from 1 to n, so it is 1+sum in deno. if k=1,2,3..N which is apparantly equal to sum if k=0,1,2,3...N
I implemented a fir filter with code, and I understand summation notation usually, but this is hard for me to get. I'm only 5 minutes into the video tho.
This video contains insane amount information. I actually like this type of videos. Not wasting viewers time at all.
this put the pieces of my entire life together
i feel that
I love the comparison of the FIR and IIR. It makes prefect sense. This is a fantastic video! It's really clear and helpful.
This is excellent sir!!! I am taking a filters course at the undergraduate level EE, and my professor is really good, but this is a nice supplement and summary of the differences between FIR and IIR.
136 slides summed up in 12.26 hatts off Sir
Started out overwhelmed, ended with a great understanding. Ty!
Holy f**k, this is some really good content!
Great video. Better and clear than my professor lecture. Thank You!!
Actually it is ok as written. Note that the sum in the denominator starts from k=0, and we define a_0 = 1. It would be equivalent to write "1 + sum" where the sum starts at k = 1.
Barry Van Veen ı have a question about fır filter can you help me about it? gokhan.arslan.763@gmail.com you can send mail to me and ı will send question to you on gmail. Thank you so much
This is a good video. I think that it would have been useful to say and show a diagram that in order for any filter to be able to recognize a band of frequencies, then the SHAPE of the impulse function of the filter must have a kernel containing the signal to be selected.
* A Low pass filters would have an decaying impulse function which contains the SHAPE related to frequencies to be selected
* A high pass filter would have an IMPULSE followed by the inverted shape of the impulse of a low pass filter.
* A bandpass filter would have a decaying impulse that encloses an oscillatory signal related to the frequency to be selected.
Then the FIR and the IIR Filters would have provided this shape which is shown on flat paper as a flat function but in reality, it is a three-dimensional rotating function going around a time axis
After all the signal e^-jwt or e^jwT may be looked at as time locations or rotating vectors are in the form of cos ( wt) + j.sin( wt) which is a rotating vector that operates in the frequency and time domain.
It would be a remarkable digital filter if it is shown in three dimensions where the pulses rotate and the delay and adders operate in three dimensions. It is also useful to look upon the Laplace and Convolution function as being three dimensional as we deal with e^jwt and not only the flat functions sine(wt) and cosine ( wt)
Very good comparison for those who wish to understand DSP and filtering as application
A clear explanation. Saved a lot of time.
Thanks a lot
Finally landed onto the correct video!! ✌
Never understood the difference between these two.
Just 1 min into the video, my doubt is cleared and i realized i have already used them by name of moving average and software filter.
You saved my exam sir
great content .........video tightly packed with lots of information
Thank You so much for this and for al people visiting this, do check out the link that shows up in the video..Really neat explanation of complex topics..Thank You from India Sir...
at 4:11, what determines how many blocks/delays that FIR filter have x[n - m]?
The number of samples (the sample rate) of the initial signal(input)?
Great video! Amazing explanation!
Thank you sir. Very clearly explained...
thanks for the video! I just have one question: at 1:50 you say that FIR filters have poles at z=0. I don't really understand why. Isn't a pole a value that makes the denominator 0? And since here the denominator is 1...
The system function (z-transform of impulse response) for a causal FIR filter is a polynomial in z^{-1}. For example, a simple one is H(z) = 0.5 - 0.5z^{-1}. This is equivalent to H(z) = (0.5z - 0.5)/z - in this form you clearly see the pole at z=0. Note that in the first form with z^{-1} when z = 0, the z^{-1} blows up.
I get it now. Thank you!
Barry Van Veen
So when do we use FIR and IIR? It seems that they both cannot be used for the same problem. When we say filters, I take it to mean band pass band stop notch filters, does it mean something more. How do I use this concept of FIR and IIR????
You mister are truly amazing!
this is so clear and simple! Helped me a lot! thank you!
great video barry
Had to deal with this in 1997, with limited internet and very few books left in the Uni library. Between this and working fuzzy logic out, I'm clearly smarter than I thought at the time. Since then, I'm clearly not. :D
good overall summary to begin with..Cheers!!
Barry Van Veen! You da man! Thanks for explaining the difference between these two.
Thank you Barry,
Very interesting content. Subscribed.
I think at 1:51 in the denominator of the impulse response of IIR filter there is a missing "1+" term
Interesting video. I had trouble making low pass filter (analog) and high pass filter, also analog. This may solve many troubles in electronic design. I need that for SSB (Single Side Band) generation utilizing 'Weaver method'. Yesterday tried to get that, and ended using three inductors for LPS, plus one for HPS, but the problem is that result is not that great. I got -60 dB attenuation at 3 kHz with LPS, but not that great for HPS for audio filtering before IQ mixers. Just... I am newbie in DSP, and don't know what is what in terms of varialbes (k=0?, M, n, and other things). Just wanted to implement that formula on my STM32, but this concept is not clear, or I am not yet familiar with that math. Do you have any link that may help solve this beginners nightmare? Thank you in advance.
Would've been nice if you briefly went over what all the different letters stand for in your equations. Not everyone learns the mathematical notation of filters at their university, you know :)
thanks for this viedo.......now i have idea about FIR & IIF
I somehow ended up here from 'how to draw' to this. I feel quiet dumb now after watching this, haha!
If M equals 2, for an iir filter does that mean it's a 2 pole filter and so on?
Oops sorry i didn't notice that the limits for y[n] in the eq. above H(z) were from 1 to n, so it is 1+sum in deno. if k=1,2,3..N which is apparantly equal to sum if k=0,1,2,3...N
Excellent video!
This was helpful - Thanks!
Hi, great work. Do you code analog computer.
it helped alot.
Thanks!
which software you have used for writing the notes ???
Can you please give an example of a generalized linear-phase lter which is not FIR, and does not exhibits odd or even symmetry?
Hi, great work. Do you code analog computer.? I have work to do? do you accept Job?
Thank you very much ! This is very helpful !
So it's not the case that h[n] (FIR) = b_k = h[n] (IIR)?
Great vid, very easy to understand! Thanks!
Concise & to the point. Thank you!
thank you so much!
Hi! I'm wondering if you can explain the difference between asexual and demisexual?
I implemented a fir filter with code, and I understand summation notation usually, but this is hard for me to get. I'm only 5 minutes into the video tho.
I get it now :d
Thanks a lot! Really helpful
Thank you so much
thanks my good sir
Great explanation! Thanks.
you can improve the user experience by adding a bit of compression on the voice :)
crystal clear, sir!
I want to understand what each varaiable in the formula means where can i find that
Ali Master did you end up finding it?
can u please put the link to download the pdf ?
Thank you
nicely explained the diffrence.!!
that's great explanation . thanx a lot ...
This video lack some background knowledge. Is there an intro to this video?
Thanks a lot
Thanks for video
Holy shit I need to learn math
hahah wtf is this
Thanks, Sir.
thank you so much barry
whoever doesn't understand this is clearly not paying close attention
if you got it,please help me. I could not understand it.
I don't understand why people find it hard to explain it like that? Are they trying to hide the truth from everyone? It sounds so easy here.
ok if u got it ,kindly explain it to me.i have got my exam and there is no one to help me.
thanks for the video
GREAT!!!!
great,thanks alot
Great!
thank you soo much :))
well it is very difficult way to explain .a beginner like me can not understand.
good
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I would love if you slow down when you talk. its hard to keep with all this information dump .
Hey could you try this software? Encounter: 'Circuit Solver' by Phasor Systems on Google Play.
What is this im accidentally here