The frequency response of the FIR filter repeats at the sampling frequency Fs and its multiples. Im confused about how this affects the filter output, if the input signal has frequencies at Fs, 2 Fs, 3Fs?This doesn't seem right to me, that a low pass filter will pass higher frequenies too, since the frequency respone repeats at multiples of Fs.What am I not understanding correctly?
FIR filter frequency response
Started by ●October 23, 2008
Reply by ●October 23, 20082008-10-23
In frequency domain, all information is contained within 0 to Fs range. Now when we say that "somethings repeat at multiple of Fs" it essentially means that you are traversing the unit circle again and again and again. So whether you traverse once or you traverse N number of time's it is same. Hence all information is contained within 0 to Fs range. where Fs/2 is called mirror image symmetry point. Even fin = 0 hz and Fs point are same. For example if you were to sample 8000 hz signal at Fs = 8000 hz, then you would get only 1 point per cycle and if you position it to be peak or any other point other then 0, then you would get a DC output. Hope this clarifies some clutter. Rgds Bharat Pathak Arithos Designs www.Arithos.com DSP Design Consultancy and Training Company.>The frequency response of the FIR filter repeats at the samplingfrequency>Fs and its multiples. >Im confused about how this affects the filter output, if the inputsignal>has frequencies at Fs, 2 Fs, 3Fs?This doesn't seem right to me, that alow>pass filter will pass higher frequenies too, since the frequency respone >repeats at multiples of Fs.What am I not understanding correctly? >
Reply by ●October 23, 20082008-10-23
On Wed, 22 Oct 2008 22:54:53 -0500, Zeph80 wrote:> The frequency response of the FIR filter repeats at the sampling > frequency Fs and its multiples. > Im confused about how this affects the filter output, if the input > signal has frequencies at Fs, 2 Fs, 3Fs?This doesn't seem right to me, > that a low pass filter will pass higher frequenies too, since the > frequency respone repeats at multiples of Fs.What am I not understanding > correctly?This may shed some light: http://www.wescottdesign.com/articles/Sampling/ sampling.html. As Bharat said, in the sampled-time domain Fs "means" the same thing as 0Hz; this doesn't make a whole lot of difference in the sampled-time domain, but it makes a huge difference in going from continuous-time to sampled-time, or back. -- Tim Wescott Wescott Design Services http://www.wescottdesign.com Do you need to implement control loops in software? "Applied Control Theory for Embedded Systems" gives you just what it says. See details at http://www.wescottdesign.com/actfes/actfes.html
Reply by ●October 23, 20082008-10-23
>The frequency response of the FIR filter repeats at the samplingfrequency>Fs and its multiples. >Im confused about how this affects the filter output, if the inputsignal>has frequencies at Fs, 2 Fs, 3Fs?This doesn't seem right to me, that alow>pass filter will pass higher frequenies too, since the frequency respone >repeats at multiples of Fs.What am I not understanding correctly? >Im not sure Im following your replies. I guess what Im asking is, when you sample a signal for instance , the frequency response repeats at multiples of Fs, and those need to be filtered, or if you uponvert you see all those frequencies 2Fs, 3Fs etc. Those frequecies actually exist. If that actually exists, then doesnt this mean even the frequency response of a FIR filter, allows frequecies at 2Fs, 3 Fs, etc.since its response is repeating at those frequencies?
Reply by ●October 23, 20082008-10-23
>In frequency domain, all information is contained within 0 to Fs range. > >Now when we say that "somethings repeat at multiple of Fs" itessentially>means that you are traversing the unit circle again and again and again. > >So whether you traverse once or you traverse N number of time's it is >same. > >Hence all information is contained within 0 to Fs range. where Fs/2 is >called mirror image symmetry point. > >Even fin = 0 hz and Fs point are same. For example if you were to sample >8000 hz signal at Fs = 8000 hz, then you would get only 1 point percycle>and if you position it to be peak or any other point other then 0, then >you would get a DC output. > >Hope this clarifies some clutter. > >Rgds >Bharat Pathak > >Arithos Designs >www.Arithos.com > >DSP Design Consultancy and Training Company. >Im not sure Im following your replies. I guess what Im asking is, when you sample a signal for instance , the frequency response repeats at multiples of Fs, and those need to be filtered, or if you uponvert you see all those frequencies 2Fs, 3Fs etc. Those frequecies actually exist. If that actually exists, then doesnt this mean even the frequency response of a FIR filter, allows frequecies at 2Fs, 3 Fs, etc.since its response is repeating at those frequencies?> > > > >>The frequency response of the FIR filter repeats at the sampling >frequency >>Fs and its multiples. >>Im confused about how this affects the filter output, if the input >signal >>has frequencies at Fs, 2 Fs, 3Fs?This doesn't seem right to me, that a >low >>pass filter will pass higher frequenies too, since the frequencyrespone>>repeats at multiples of Fs.What am I not understanding correctly? >> >
Reply by ●October 23, 20082008-10-23
>On Wed, 22 Oct 2008 22:54:53 -0500, Zeph80 wrote: > >> The frequency response of the FIR filter repeats at the sampling >> frequency Fs and its multiples. >> Im confused about how this affects the filter output, if the input >> signal has frequencies at Fs, 2 Fs, 3Fs?This doesn't seem right to me, >> that a low pass filter will pass higher frequenies too, since the >> frequency respone repeats at multiples of Fs.What am I notunderstanding>> correctly? > >This may shed some light:http://www.wescottdesign.com/articles/Sampling/>sampling.html. > >As Bharat said, in the sampled-time domain Fs "means" the same thing as >0Hz; this doesn't make a whole lot of difference in the sampled-time >domain, but it makes a huge difference in going from continuous-time to >sampled-time, or back. > >-- > >Tim Wescott >Wescott Design Services >http://www.wescottdesign.com > >Do you need to implement control loops in software? >"Applied Control Theory for Embedded Systems" gives you just what itsays.>See details at http://www.wescottdesign.com/actfes/actfes.html >Im not sure Im following your replies. I guess what Im asking is, when you sample a signal for instance , the frequency response repeats at multiples of Fs, and those need to be filtered, or if you uponvert you see all those frequencies 2Fs, 3Fs etc. Those frequecies actually exist. If that actually exists, then doesnt this mean even the frequency response of a FIR filter, allows frequecies at 2Fs, 3 Fs, etc.since its response is repeating at those frequencies?
Reply by ●October 23, 20082008-10-23
On 23 Okt, 21:49, "Zeph80" <surabhi_tal...@hotmail.com> wrote:> >The frequency response of the FIR filter repeats at the sampling > frequency > >Fs and its multiples. > >Im confused about how this affects the filter output, if the input > signal > >has frequencies at Fs, 2 Fs, 3Fs?This doesn't seem right to me, that a > low > >pass filter will pass higher frequenies too, since the frequency respone > >repeats at multiples of Fs.What am I not understanding correctly? > > Im not sure Im following your replies. I guess what Im asking is, when you > sample a signal for instance , the frequency response repeats at multiples > of Fs, and those need to be filtered, or if you uponvert you see all those > frequencies 2Fs, 3Fs etc. Those frequecies actually exist. > If that actually exists, then doesnt this mean even the frequency response > of a FIR filter, allows frequecies at 2Fs, 3 Fs, etc.since its response is > repeating at those frequencies?You are correct, both in your observations and in the concern about the ambiguity. There is nothing you can do about such ambiguities in discrete-time domain. That's why one always should use an analog anti-alias filter prior to the ADC in sampled systems. The anti- alias filter should only allow one of the 'mirrors' through (it needs not be the baseband 'mirror') so that one always knows the 'true' bandwidth of the signal. Rune
Reply by ●October 23, 20082008-10-23
>The frequency response of the FIR filter repeats at the samplingfrequency>Fs and its multiples. >Im confused about how this affects the filter output, if the inputsignal>has frequencies at Fs, 2 Fs, 3Fs?This doesn't seem right to me, that alow>pass filter will pass higher frequenies too, since the frequency respone >repeats at multiples of Fs.What am I not understanding correctly? >I finaly found something which answers my question. In this link http://www.hunteng.co.uk/pdfs/tech/ddctheory.pdf on page 3 he clearly shows the frequency response of the LPF filter repeating at Fs,and he shows that if the input to the filter has frequencies at Fs and 0, both will be passed through.
Reply by ●October 23, 20082008-10-23
Zeph80 wrote: ...> Im not sure Im following your replies. I guess what Im asking is, when > you > sample a signal for instance , the frequency response repeats at > multiples > of Fs, and those need to be filtered, or if you uponvert you see all > those > frequencies 2Fs, 3Fs etc. Those frequecies actually exist. > If that actually exists, then doesnt this mean even the frequency > response > of a FIR filter, allows frequecies at 2Fs, 3 Fs, etc.since its response > is > repeating at those frequencies?You do understand that before sampling an analog signal at a sample rate fs, you must low-pass filter it to remove components at and above fs/2? (Bandpass sampling is an extension of this, not an exception to it.) Since the input to the filter can be no higher than fs/2, what is there to repeat? A more subtle way to say the same thing is that every component you feed the system that lies above fs/2 is translated by the physics of the situation to one below according to some simple rules. Keyword: alias. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●October 24, 20082008-10-24
surabhi_talwar@hotmail.com wrote: > visited DSPRelated.com and clicked on your name from this page: > http://www.dsprelated.com/showmessage/104558.php > to contact you. His message follows: > > I finaly found something which answers > my question. In this link > http://www.hunteng.co.uk/pdfs/tech/ddctheory.pdf on page 3 he clearly > shows the frequency response of the LPF filter repeating at Fs,and he > shows > that if the input to the filter has frequencies at Fs and 0, both will > be passed through. > > I guess I didnt frame my question well, but this is what I was asking. Chopping a signal by sampling it creates images. When those images overlap, we call them aliases. The important point to keep in mind is that for discrete-time signals, 0 and Fs are one and the same. The continuous-time frequency line becomes a frequency circle with discrete-time signals. On that circle, 0 and Fs overlap. Please don't bring newsgroup issues to me at home. Aside from any possible intrusion, you get the benefit of the group's collective wisdom by posting publicly there. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������






