filtering in matlab using 'built-in' filter design techniques

[@mikeshaw714]

Just like to say I find all your videos on FFT/MATLAB brilliant. Very clear and concise, they have definitely helped me understand the topic, a must view for anyone wanting to learn.

[@ddorran]

Thanks for the comment. It's great to know these videos are helping out.

[@manhuynhkha]

Perfect! You're the best teacher that I've ever seen. Good job, dude!

[@rajeshbodade]

Perfect! You're the best teacher that I've ever seen. Good job, Keep it up.

[@Karikato]

Best explanation of the filters I've seen so far. Very good job showing the differences! :)

[@ahhegab2]

Brilliant .. it ended my misery after suffering for a long time. Thanx Dave :)

[@seselian]

You did a very good job, clear and easy to understand.

[@stephenalabi3345]

David I really appreciate ur effort. The lecture is concise, clear and easy to understand. Please can you talk about Basspass filter?

[@ddorran]

I will do something on bandpass filters when I get a chance - its on the to do list! I do have an example of a bandpass filter being used in Why Linear Phase Filters are Used which might be of some help.

[@mostafahoseini8888]

Simple, straightforward and helpful! Thank you.

[@dinghydan]

At last a tutorial that normal people can understand!

[@TheAhmedMAhmed]

wonderful ! really helped me a lot in my project of my 1st Signals and Systems course; Thanks :)

[@6400loser]

You are a goddamn life saver, David Dorran.

[@kishonpatel7161]

thnx a lot David. ur techniques in MATLAB r really outstanding and simple to understand. we will like to have more videos of urs in future. Thnx again :)

[@mshahrul]

cheers david... u help a lot of people with this

[@jimz92raisanz]

impressive, you are really helpful sir , do you have the code for the LOW PASS FILTER WITH FINITE IMPULSE RESPONSE USING LEAST SQUARE TECHNIQUE SIR

[@pratyushpradhan1844]

brilliant explanation . this is what i was looking for.

[@TheGoodphy]

Oh...This is what I "exactly" look for!! Thanks!!! It is really straightforwardly applicable now!

[@hienlevan6726]

Thank you so much for your useful tutorial, David. I am trying to apply filter technique for my real data which is taken every 10 minutes average of raw data including so much noise data and some missing data also. I only know the frequency of system was designed at 20Hz. Could you recommend me how to apply filter method in general and how to define the optimal order of filter and which technique in filter method is better for my case. Thank you so much for your time in advance.

[@markpage5108]

I really like your videos, please keep doing them. I learn something from each one!

[@杰米老师]

I really dont know how to express my gratitude. Thanks so much! do you have videos on how to filter without built-in-functions?

[@ddorran]

+Jamirah Ama Ahmed I don't have any videos explicitly on this topic. However if you understand how a systems poles and zeros influence the frequency response then you can design any discrete filter you want without having to rely on built-in functions. Take a look at kzbin.info/www/bejne/o2a3gWWrfMSCaK8

[@杰米老师]

I downloaded your pdf on basics of DSP! I always run away from Matlab during undergrad and now i need it for signal analysis etc etc for this Masters course I am taking. Your pdf is really helpful. Thank you so much. Wish I could email you or something! I am grateful!

[@ddorran]

+Jamirah Ama Ahmed Thanks for letting me know my material is useful. It's nice to get that type of feedback. Good luck with your studies.

[@vickysteev8964]

Hi, i am Vicki Right now i am learning matlab, but i am just begginer at matlab, right now i want to learn about how to using low and high pass filter, Savitzky-Golay Filtering (SGOLAYFILT), mean and varians, and also find out the peak and valley point from the gyro and accelerometer signal, and all my data i get from excel. can You give me some tutorial or code how to make it I really hope You can help me to figure it out. Best Regards Vicki

[@mutiaaziza1394]

perfect! very good tutorial. thank u so much Mr..

[@aseramohammed4597]

What does it do this function Create signal (a,b)

[@tdawg108]

Thank you for your video, it was a great help. I managed to plot the amplitudes of both the Chebyshev and Butterworth filters but could you please advise me on how to plot the phase of these as well?

[@ddorran]

The following should work: [H w] = freqz(b,a); plot(w,angle(H))

[@govindsahu15]

Sir, If we add higher order filter then there will be some delay in the signal. So, are there any methods to avoids/compensate such delay? Thanks

[@aseramohammed4597]

Sir I wanna aske you about Create_signal_flow(b,a) In line 32 . What should I do to creat this function?

[@hmir9588]

thanks sir, But what about a signal exponentially sinusoid ?! and how to eliminate dc offset and exponential term getting a pure sinusoid? actually DC offset is easy to get rid of it. Waiting for your reply for eliminating exponential term of sinusoid.

[@nurahmedomar]

This is a Digital filter, is there any similar video about an analog filter? Thanks.

[@mustaphaalkhafaaf5512]

hi ,, grate video ,,, one question though ,, after all this how can engineers go and build the filter ,, after all this is just a matlab code , we dont have a real schematics and components ??!?!?! ... cheers

[@mustaphaalkhafaaf5512]

+Mustapha Alkhafaaf it would be useful if we have option to turn to schematics

[@ddorran]

+Mustapha Alkhafaaf by writing the code you have built the filter! Take a look at kzbin.info/www/bejne/qYW7lJVpoJ6qnJo. The schematic is the signal flow diagram. The components are multipliers and adders and memory which all exist in a computer and micro-controllers. In order to filter a continuous signal using a discrete filter you just need to get the signal on to the microcontroller - which is achieved using an analog to digital converter (see kzbin.info/www/bejne/n5PNkGRpeJqXr8k). Once the signal has been filtered by the microcontroller you can send it out of the micro using a digital to analog converter. Note that you could also build a filter using electrical components such as resistors, capacitors and op-amps.

[@mustaphaalkhafaaf5512]

Cheer David for the explanation, you have mentioned the delay , I assume this available only in the processor with DSP capability not all micros or processors have this ... and to be honest with you i dont uderstand why do we need the delay .. why ?

[@ddorran]

The delay is implemented by just storing data in memory - so all micros can implement discrete filters. The delay is a key element and filtering would not be possible without it. Look through the introductory videos to gain insight into this kzbin.info/www/bejne/Y4OwaYmmpNedibs

[@mustaphaalkhafaaf5512]

many thanks

[@GaiaKnight11]

Thank you very much! Excellent explanation!!

[@rimzanrefaideen9168]

Clear and simple. Really helped!!!!!

[@priyeshshah4750]

thanks this video really helped, I would like to know the a FIR method for filtering

[@amineleking9898]

Thank you so much David

[@francescovatrano8687]

Hi David I used successfully what you show in this video but if it's possible i want to pose a question to you about filtered signal. I started with a signal composed by a sinusoid and noise. Than i filtered the signal by removing sinusoid . Now we have only noise and i have do demonstrate that noise have zero mean and it have a probability density function described by a Gaussian function. So I executed the command: sort(signal) in order to obtain a vector starting by lower value to higher value. Than i plotted histogram and the gaussian curve and now i have to do the ki square test but i have not idea how to do this. Can you suggest me a way? Thank you so much.

[@aakashdeep8716]

The filter you created is of amplitude 1. How can we change the amplitude of filter, if let say I want the amplitude to be 6?

[@ddorran]

You can multiply the input or the output by 6. Multiplying all the b coefficients by 6 and then filtering would have the same effect.

[@aakashdeep8716]

Thanks for the support. It worked.

[@Appleiphones1]

i have 3 phase voltages and current.Now how can i obtain Fundamental components of current and voltage signals using DFT?

[@jyotimishra473]

I want to apply bandpass filter on a discrete signal, can you suggest any method for that?

[@dongdli4708]

Thank you very much! It helps me a lot.

[@dipalisinha3427]

sir, i have an eeg signal with the sampling rate 173.61 , duration 23.6 second, i want to implement the band pass filter having the freq. range 0.2 to 35 hz. would u plz help me

[@sultanalhammadi2910]

what if there is more than frequency component and there is noise between them? how would you filter it

[@husseinalgusab4129]

Thank you very much for this video David but i have a question : I have a signal and i had filtered it according to your method but there is some discrete specified frequencies ( about 100 values ) in the range of frequencies which i had already removed, so how can i add these specified frequencies (where lies in the range of removing values of frequencies ) to the original signal so as not to remove these specified values of frequencies thank you very much

[@ddorran]

So you have a filter that, for example, removes frequencies in the range of 100 to 200Hz, but you now realise that you actually need to keep frequencies around 150Hz. I hope I understand you correctly. One technique you could use is to first filter frequencies from 100 to 149Hz, giving you a new signal y1. You could then pass y1 through a filter which removes frequencies between 151 and 200Hz. There are other ways of doing this but this is probably the most straight forward for someone starting out.

[@husseinalgusab4129]

Thank you very very much, i really appreciate your fast answer it's very kind of you and yes, i mean exactly what you understand thank you very much sir

[@sumilangovinden]

Excellent Stuff

[@rollinfever]

Great video but I need some help! I am trying to filter a .wav audio file. I transformed it using fft( ). When I transformed it, the peak magnitude of the noise that I want to filter is around 12000 (not sure the units). How can I make this peak smaller to match your 0 to 1 magnitude which works well with the butter function you used? Or is there any way I can make a filter that has a magnitude of 12000? Thanks!

[@benyaminzakariah9543]

Hi there. I'm working on my project called Audio Compressor. How can I compress my .wav file to different factors and how to make a spectograph from it? Thank you :)

[@vineeshms9561]

superb....able to understand.thank you

[@rajaduraikanniappan5327]

Hi..Thank you so much for this video. I would like to design a butterworth bandpass filter with different orders (eg. low pass filter of order 2 and high pass filter of order 8). So could you please help me, how this could be done? Thanks in advance.

[@aminehaine3301]

good job :) you helped me a lot for my work :)

[@stupid8154]

a real help , cheers !!

[@ronaldinho711990]

thank you so much very useful explanation

[@thepyramid9777]

i need function and simulink in power system by useing kalman filter..... can you help please ?

[@unaihenry2964]

Just question ; how can i design BPF in matlab without using fir function

[@abhimgowda8168]

sir , i am doing a project on voice recognition , can i know how to use these filters to remove the noise...please help me sir

[@AbhilashaDAdak-iu7mw]

Do we need to enter four parameters in the 'butter' function for band pass and band reject filters?

[@ddorran]

For a bandpass filter you would use [b a] = butter(3, [0.2 0.4], 'bandpass') There are still three parameters passed to the function though. Each function parameter is separated by a comma and the second parameter is an array containing two values i.e. 0.2 and 0.4. I can see why this could be described as four parameters but in terms of "function parameters" there are only three.

[@AbhilashaDAdak-iu7mw]

David Dorran Thanks a ton.

[@aseramohammed4597]

Create_signal_flow(a,b) Can you give me the code sir please

[@ddorran]

Its available at dadorran.wordpress.com/2012/09/26/create_signal_flow/ or by searching create_signal _flow using the search engine of your choice

[@harunyldrm2624]

Hello Engine vibration time domain data has been taken. How do I covert from time domain into crank angular domain? May you help me?

[@UtkarshSoni2512]

Very good tutorial :)

[@lonesoldierx94]

Question: Hey, I'd like to use the freqz function and plot it simultaneously with pwelch or psd. Is that possible? I keep getting that they are not even values.

[@ddorran]

+lonesoldierx94 Here's some code that'll do the job. Basically I normalised the frequency axis (see kzbin.info/www/bejne/aHrOlYGpfLaae68) I was expecting the output of pwelch to be the same as H^2 but for some reason it was out by a factor of 3 (that's the reason for the plot in green). I have no idea why at the moment. [b a ] = cheby1(7, 1, 0.5); H = freqz(b,a); x = randn(1, 10000); y = filter(b,a,x); P = pwelch(y,200); plot([0:length(H)-1]/length(H),abs(H).^2); hold on plot([0:length(P)-1]/length(P),P,'r'); plot([0:length(P)-1]/length(P),P*3,'g'); legend('abs(H)^2 where H= freqz(b,a)', 'psd from pwelch','psd*3')

[@lonesoldierx94]

+David Dorran It works! Thank you!

[@saifalikhanpathan3422]

how i can remove noise from a .wav file without using filter functions?

[@ayaibdah5646]

hello can you help me in question in Image processing ?

[@KelvinLeUT]

good explanation!

[@jyothiguntamukkala9392]

sir , how to design pso for low pass lcl filter please tell

[@joeblow4938]

Is there some other tool besides MATLAB that plot these digital filter transfer functions easily?

[@ddorran]

+Joe Blow You can try using Octave, Scilab or Python

[@rohitkulkarni8873]

nice video sir...

[@aseramohammed4597]

Please can anyone help me .

[@francescovatrano8687]

Can I create an high_pass filter in the same way?

[@ddorran]

yes. just use [b a ] = butter(2,0.3,'high'); on line 28

[@francescovatrano8687]

Thank you ;)

[@kinjalmacwan4866]

Awesome! Thanks!

[@mdmonazirkhalifa3997]

Sir which version of matlab u r using

[@Rkalla251]

dis guy is great

[@mohamadamin8122]

how to make the block diagram?

[@tajammalnawaz5552]

how to set the cuttoff freq.?

[@Hyperbolicus86]

thank you, great instruction

[@MrUsamarao]

Thank You Sire.. =D

[@rohandhankani3962]

TOP thank u very much sir for your help

[@csmatyi]

how do you determine the cutoff??? of 0.3??

[@ddorran]

+Matthew Cserhati By examining the frequency content of the signal you can see that the noise occupies frequencies in the region of 0.5 to 0.8 (normalised frequency values). If you are using a low pass filter then the cutoff must be less than 0.5. By choosing a cutoff frequency less than 0.5 you will reduce higher frequencies by a greater amount. I selected 0.3 by trial an error - it reduced the noise by a visually pleasing amount that seemed to work well to illustrate the concept of filtering in the video. There are more formal ways to select the cutoff frequency and if you are interested then the following might help kzbin.info/www/bejne/enSmgn2If9KagtE

[@balotrune7499]

thank you so much.

[@hynesie11]

Thank you thank you

[@habibudinshams4414]

can you give me its code?

[@MyawesumMe]

and was is the "floor" for in 6:21

[@ddorran]

floor rounds down to the nearest integer. It's similar to round but will always round down. For example round(1.8) = 2, but floor(1.8) =1. If the decimal part of a number is less than 0.5 then floor and round will have the same result. for example floor(10.3) = 10 and round(10.3) = 10

[@MyawesumMe]

what is "num_bins" in 3:14 ?

[@ddorran]

it is the total number of frequency bins obtained from the fft function. See Using Matlab's fft function or How the Discrete Fourier Transform (DFT) works - an overview for an explanation of the term.

[@putianhe7692]

nice video

[@uzidesigns23]

thankyou!

[@azeemahamed3942]

how to perform for realtime bruvvvv

[@Captain_Rhodes]

why are the normalized units pi*rad/sample? does anyone actually understand that? ive yet to see anybody explain it!

[@ddorran]

There are 2 pi radians of a revolution in one cycle of a sinusoid. The maximum frequency that can be captured faithfully without aliasing will be half the sampling frequency. Therefore a sinusoid of 'maximum frequency' will have 2 samples per cycle. So over two samples the 'maximum frequency' sinusoid will have undergone 2 pi radians of a revolution, which equates to pi radians per sample.

[@Captain_Rhodes]

@@ddorran Thanks, that does make some sense. However what confuses me is that matlab says that the normalized frequency axis is defined as f/(fs/2): frequency divided by nyquist frequency. that should be units of cycles/sample right? Im starting to see what you mean, but in terms of the units I still dont get how that pi factors out

[@ddorran]

​@@Captain_Rhodes fnorm=f/(fs/2). f = w/(2*pi), where w is frequency in radians per second (rads/sec). fs is frequency in samples per second (samps/sec). fnorm=w/(pi*fs), therefore units can be expressed as rads/sec/(pi*samps/sec), which reduces to (rads/samps)/pi. Another way to think about this is that fnorm*pi = w/fs, so fnorm*pi has units of rads/samp. So if fnorm is 1 and we want to express this in radians per sample we'd have to multiply by fnorm by pi. The frequency axis should actually be labelled ( x pi radians per sample) to indicate that the normalised frequency should be multiplied by pi to get the frequency in radians per sample. I hadn't noticed this omission before.

[@Captain_Rhodes]

@@ddorran Thanks for your time. its much appreciated. All the best mate

[@protocolfree]

is this an analog filter ?! ( excuse my stupidity and ignorance )

[@ddorran]

No. It's a discrete filter.

[@alieng3002]

goooooooooooooooooooooood

[@الحمدللهاستغفرالله-ح1و]

Help me teacher

[@mdmonazirkhalifa3997]

Sir first define the signal 'x'

[@ddorran]

It's defined in dadorran.wordpress.com/2013/10/18/filter-design-using-matlab-demo/

[@chillingmeko]

@5:13 0.74 should be negative.

[@ddorran]

+Ohmeko Ocampo It is correct. It should be positive. The reason why can be explained as follows (kzbin.info/www/bejne/fnuok6ivaa2GqK8 is a related video): The b and a coefficients are associated with the general form of a difference equation which is written as follows: a0.y[n] + a1.y[n-1] + a2.y[n-2] + a3.y[n-3] + .. ... + ak.y[n-k] = b0.x[n] + b1.x[n-1] + b2.x[n-2] + b3.x[n-3] + .. ... + bm.x[n-m] when it is rewritten with the output y[n] on the left hand side the sign of the a terms change. Letting a0 =1 the general form becomes: y[n] = b0.x[n] + b1.x[n-1] + b2.x[n-2] + b3.x[n-3] + .. ... + bm.x[n-m] - a1.y[n-1] - a2.y[n-2] - a3.y[n-3] - .. ... - ak.y[n-k]

[@chillingmeko]

David Dorran Woah, thanks so much. I had a difficult with this. XD

[@mariamoosa4058]

I want to remove a 50Hz and 100Hz noise from a EEG signal. For that I need to design 2 notch filters. How do i do that?