Working of Digital filters...

[RRITESH KAKKAR]

I was just thinking how Digital filters work in case of analog we just connected
RC to the signal input to amplifiers, etc
but in case of Digital filter....what happen??

Thanks

[Papabravo]

In an FIR (Finite Impulse Response) filter the present output is a function of the current and previous inputs. If the input goes away then the output goes away. That is the 'F' in FIR.

In an IIR (Infinite Impulse Response) filter the present output is a function of the current and previous inputs AND the previous outputs. If the input goes away the out does not have to go away. That is the first 'I' in IIR.

Finite and Infinite Impusle Response refers to the output over time, NOT the amplitude of the response.

[crutschow]

For a digital filter you first convert the analog signal to digital format by sampling the signal at greater than the Nyquist sample-rate for the highest analog frequency using an A/D converter. A digital processor then takes these samples and performs a mathematical function of the values to generate the desired filter response for each sample. If desired the samples from the filter can be then converted back to analog with a D/A converter.

[RRITESH KAKKAR]

In case of analog low pass filter with connect R and C by using cut of frq. F=1/(2*PI*R*C)....and analyzing by Bode plot.
But this how really work in digital filters??
The input isin form of 010101010 if i want low freq. output of digital signal then what to do??

[crutschow]

Originally Posted by RRITESH KAKKAR

In case of analog low pass filter with connect R and C by using cut of frq. F=1/(2*PI*R*C)....and analyzing by Bode plot.
But this how really work in digital filters??
The input isin form of 010101010 if i want low freq. output of digital signal then what to do??

The digital filter algorithm is what determines the filter function. A simple low pass function can be performed by generating a running average of the digital values. Thus you would take the difference between the running average and the next sample and add a percentage of the difference to the running average. The cut-off frequency is determined by the digital sample rate and the percent value you add to the running average. It's not as simple a function as the analog circuit.

For more info look at one of these Google references.

[Papabravo]

It works the same way except there is no R and no C. What you do have control of is the sample rate and the coefficients of the impulse response. If you know the coefficients of the impulse response you can compute the output via convolution.

http://www.dspguide.com/

I think Prof. Smith's book is still a free download by chapters in PDF format

[RRITESH KAKKAR]

Guys can you just give an block example for better understanding of digital filters..??

[Papabravo]

Originally Posted by RRITESH KAKKAR

Guys can you just give an block example for better understanding of digital filters..??

Your best bet is to go to the free resources like Steve Smith's DSP book.

Here is another one

http://en.wikipedia.org/wiki/Finite_impulse_response

Boxcar filters have a low-pass characteristic similar to the multi-pole RC filter.

When you do some reading it will lead to more specific questions.

[RRITESH KAKKAR]

I think i am know getting how digital filters work as first we convert analog to Digital then uC/uP do calculation like FFT,etc then output is converted analog.......

But if we talk about convolution theorem it say multiply in freq. domain, vice versa...but what the use and how does it happen in CPU itself??

[Papabravo]

A digital filter is implemented in a CPU by a succession of multiply and accumulate (add) operations. In some processors there is even an instruction that does exactly that operation called "Multiply and Accumulate". A processor can also keep track of previous inputs and previous outputs which may also be used in the filter calculations. Was that what you were looking for?