I read in one of your replies that we need to connect a capacitor for the proper functioning of an analog to digital converter for a low pass filter..what does dat mean??
My friend GOOGLE says that rc low pass filter provides low frequency which prevents signal attentuation..I this the only use??
If yes why does this attentuation occur??

 

Um, who, what? It is typical to put a Low pass filter (LPF) on the input of a A/D so you do not sample frequencies beyond interest. The LPF acts as a high frequency attenuator above a set point. The response of the filter looks like a flat top with a sloping edge. Kinda like a house on a hill with the end of the yard sloping down to the ocean (Dream house). The house is like the frequency of interest, the ocean is ignored because it is so far down (Unless you surf) and is attenuated. In a nutshell, the LPF passes frequencies up to a point, anything beyond is attenuated and ignored. There are probably better explanations with better analogies, but thats all I got, I have a O chem exam to study for

 

thanks a lot...
Just one more doubt!!
Why do we give inputs to an ADC filtering it on frequency levels..I mean frequency greater than (say x) is not allowed and attentuated..but frequency less than (x) is allowed as input to an ADC..
Why do we do this?? If i plan to make a line follower How is it related to my bot functioning..??

Sorry for bodering..

 

Because, frequency X will have multiples of itself, like a mini-me, or more commonly known as a harmonic. These are undesired components and should not be sampled or your bits coming out will be wrong.

 

thanx a lot..

How do we get to know what frequency should X be set to??
What I mean to say is..how do calculate the value of a frequency above which I will attentuate my signal..??
is it equal to (1/rc)..??

 

You set X to maximum frequency you want to pass. For example; Audio 20Hz to 15KHz. You set your 3dB point for 15KHz, anything beyond this is attenuated. If your mini-me's need to be real small, you can use a multi pole filter for steeper edges, but this is getting beyond our scope I think. Pls do not ask me to design a filter for you, I have neither the time, nor the inclination to do so.

 

Hi,
The main reason of the LPF before ADC is to anti alias. Otherwise the aliased signal may fall into the passband and causes result to be affected which is known as noise.

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Yep you all right

You could also calculate accurate readings by over sampling & adding a software hysteresis.But a design with both hardware & software will be too great.

But for a line follower it doesn't need much more sensitive readings.

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My Website
http://gsmicro.blogspot.com/

 

hi,
What is the voltage/signal connected to the ADC input.??

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Eric
"Good enough is Perfect"

PIC tutorials:
Gramo's: www.digital-diy.net/
Bill's: www.blueroomelectronics.com/

 

What! you did not like the mini-me filter? Sure, anti alias sounds better.

 

Patwari i thing this link is useful for you (need java).


Signal Sampling and Reconstruction

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Thanks a lot people..

 

There are two reasons to filter an A/D input: reduce noise above the input frequencies of interest, and anti-alias.

Generally you want a filter to roll off the signal above the highest input frequency of interest to suppress noise above that frequency, which would otherwise add undesirable noise to the digitized signal.

At a minimum you want an anti-alias filter to suppress the input above 1/2 the A/D sample frequency, even if there are signal components above that. Any noise or signal frequencies above 1/2 the sample frequency are digitized and appear as low frequency (aliased) components in the digitized signal. This adds noise and difference signals to the digitized signal. For example, an input signal exactly equal to the sample frequency will appear as a dc shift in the digitized signal since the sample will always occur at the same point on the waveform.

If you took a 10KHz bandwidth music signal and fed it to a 10k sample/sec. A/D converter with no anti-alias filter, the digitized output would sound terrible when converted back to analog and played through a speaker. This is because all frequencies between 5kHz and 10kHz have been aliased back into the 0 to 5KHz band. (Nyquist's sample theory says the sample rate must be a least twice the highest frequency of interest. Thus a 10kHz sample rate can reproduce a maximum of 5KHz.)

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Carl