Reconstruction of Signals

[@karlm9584]

It all makes sense now, thank you.

[@karthikeyann1808]

Really crystal clear examination sir, salute

[@holly1190]

Amazing lectures sir ❤️💞thank you sir for such teachings...!!

[@krishnakumarsingh4386]

Explanation is awesome!!!!!!!!!!!😎😎

[@muhammedalikumasoglu7221]

Greatings from Turkey Sir I. learned from you a lot. This channel will always have a place in my heart.

[@QuizletQ]

thanks a lot sir🥰

[@manishadas7434]

sir, please upload a lecture series of full digital communication

[@JituSingh-uu2im]

you are superb bro . thanks lord

[@vakhariyajay2224]

Thank you very much. You are a genius. 👍👍👌👌🙏🙏🔝🔝

[@freeandreliablejeeprep820]

thank you srila prabhupad , krishna and sir for explaining

[@chahlabedroh4703]

u r the best ...

[@sultanayeasmin3444]

very very helpful video indeed! thanks a lot sir!!

[@raj_kumar08]

Nice video
I got the concept easily.
Thanks

[@AhamedKabeer-wn1jb]

Well explained...

[@Thatipelli_Harishankar]

Hoo nice explaination
40rad/sec is answer for homework problem

[@awesomearchit111]

Sir will you upload lectures on PSD,ESD and correlation??

[@WaqasAhmed-hg5rd]

is it uploaded?

thank you sir

[@shubhamkhandelwal8833]

Thanks

[@rr8055]

Thanku very much
It is very much cleared sir

[@poonammantry6958]

Thank you ☺️

[@sarushanjegatheswaran3935]

just amazing!! thx

[@yashsengupta6116]

Congratulations for 1million

[@sagnikmukherjee6604]

Sir why is H(w)=Mr(w)/s(w)
Also why S(w) = 1/Ts for w=0
How is gain of low pass filter = Ts

[@amalenduaman7239]

Low pass filter is an LTI system,so its output will be convolution of input and impulse response in time domain...but in frequency domain we do multiplication instead of convolution....I mean we will multiply sampled signal in frequency domain with transfer function H omega

[@mudassarraza6448]

Great lecture.

[@brentameszimmerhanzel2468]

Your lectures are getting me through this course!!

[@egehanusluer489]

By the property can we also say mr(w)=s(t)*h(t) where h(t) is inverse Fourier transform of H(w)?

[@akilaava2490]

Thank you sir.

[@abhishekrankawat4332]

Great work

[@raj-nz4bj]

i want to donate but I am student hope in future i could do so. thank you.

[@phymath-jisanislam9639]

Why the value of S(w) is 1/Ts , when w = 0 ?

[@sreegopal4673]

I'm also having same doubt pls give a clarification on this.

[@alphasatari]

Sree Gopal If you’ve followed lecture on “Sampling Theorem” then there was a relation for Fourier Transform of s(t).
S(ω) = (1/Ts)[...+M(ω+ωs)+M(ω)+ M(ω-ωs)+...]
When, n = 0
M(ω) = 1
S(ω) = (1/Ts)[M(ω)]
S(ω) = (1/Ts)

[@subhambhuyan3323]

@@alphasatari but he said w=0 not n=0

[@shahkrunal6820]

Thank you very much sir for the best explanation

[@manoahjoseph5283]

Sir what if the spectrum had only the right part of the triangle I mean just from 0 to W what will it be its sampled spectrum?

[@anantmishra4893]

sir how that slope becomes 1/Ts and that mag of h(t) becomes Ts, could you please tell me that sir

[@sohamroy7668]

Same doubt

[@rahulkothapalli6629]

@@sohamroy7668 watch the previous lecture M(w) contains 1/Ts as weight so we are designing to get our m(t) back, whose amplitude is 1. we are taking a filter which has the amplitude after multiplying with the 1/Ts and the result becomes 1 so we're taking Ts

[@i_m_hk]

No confusion sir🙏

[@naushadturja6547]

You are great

[@ranjansrivastava4455]

Excellent

[@darrendoang]

so what is omega c?

[@balaviknesh8074]

How oversampling is like interpolation

[@sreegopal4673]

Why the value of s(w) =1/Ts at w=0 ???

[@Editechno]

Sir ur voice is very cute♥️

[@saranv8173]

sir please make video series on analog and digital communication

[@-A-Ruchitha]

Much respect to you sir 🙌🙌

[@lorenzosvenmontero168]

Could you explain how the amplitude of H(W) is T_s?

[@huihui666]

Ts is just an arbitrary value

[@huihui666]

^for the gain of the lowpass filter

[@nithinmanepalli2772]

@@huihui666 no dude it's not arbitrary value Ts here is the sampling period and that lowpass filter gain should be Ts because only then we can recover our original signal back . In sampled signal's frequency response the peak of that triangle is at (1/Ts). but our original peak is at 1 so to recover back our signal lowpass filter gain should be Ts

[@notimepass9456]

​​@@nithinmanepalli2772How did you determine that the amplitude of sample signal spectrum is 1/Ts and that of the Original message signal spectrum is 1?

[@gamar1226]

Best teacher!

[@chethankumar1541]

Amazing explanation sir
Tqsm💖❤️‍🔥

[@navyamalla7362]

🙏🙏🙏👏👏

[@satyaweeryadav3972]

sir z transform ka lecture dalo

[@dhruvvarshney1040]

#bestteacher

[@anuragkumar3285]

40pi rad per sec.

[@editztown5262]

40π

[@CurdiOnTheTrack]

Hbb u

[@josephcherian506]

mwuthmani :-*

[@successisreal151]

946424

[@Avinash_Kumar-n9g]

S

[@abderrahmeneeng8444]

Thank you so much

[@AIVidya]

#bestteacher