For example, for a given 10s length audio signal. I want to perform LP filter from 3s to 5s time window. At the same time, the total sample number should remain the same with the un-processed one. Who can tell me how to do it? Thx very much. Kevin _____________________________________ Do you know a company who employs DSP engineers? Is it already listed at http://dsprelated.com/employers.php ?
How to perform filtering on a time window for an audio signal?
Started by ●April 11, 2007
Reply by ●April 11, 20072007-04-11
On Apr 11, 10:29 am, "dingke1980" <dingke1...@hotmail.com> wrote:> For example, for a given 10s length audio signal. I want to perform LP > filter from 3s to 5s time window. At the same time, the total sample > number should remain the same with the un-processed one. Who can tell me > how to do it? > > Thx very much. > Kevin > > _____________________________________ > Do you know a company who employs DSP engineers? > Is it already listed athttp://dsprelated.com/employers.php?I can. Anyone else? Dirk
Reply by ●April 11, 20072007-04-11
Lets call your signal x. Extract the signal from 3s to 5s (call it y). Padd y with zeros and subtract x from y: z= x - ypad (where ypad is y padded to be equal in length with x). Filter y (call it yf) and add it to z to get the final result: Signal= z + yf On Apr 11, 9:29 am, "dingke1980" <dingke1...@hotmail.com> wrote:> For example, for a given 10s length audio signal. I want to perform LP > filter from 3s to 5s time window. At the same time, the total sample > number should remain the same with the un-processed one. Who can tell me > how to do it? > > Thx very much. > Kevin > > _____________________________________ > Do you know a company who employs DSP engineers? > Is it already listed athttp://dsprelated.com/employers.php?
Reply by ●April 11, 20072007-04-11
On Apr 11, 10:54 am, "Ikaro" <ikarosi...@hotmail.com> wrote:> Lets call your signal x. > > Extract the signal from 3s to 5s (call it y). > Padd y with zeros and subtract x from y: > > z= x - ypad (where ypad is y padded to be equal in length with x). > > Filter y (call it yf) and add it to z to get the final result: > > Signal= z + yf > > On Apr 11, 9:29 am, "dingke1980" <dingke1...@hotmail.com> wrote: > > > > > For example, for a given 10s length audio signal. I want to perform LP > > filter from 3s to 5s time window. At the same time, the total sample > > number should remain the same with the un-processed one. Who can tell me > > how to do it? > > > Thx very much. > > Kevin > > > _____________________________________ > > Do you know a company who employs DSP engineers? > > Is it already listed athttp://dsprelated.com/employers.php?- Hide quoted text - > > - Show quoted text -It appears that 'z' is 'x' with data from 3-5 seconds zeroed out, and 'y' is 'x' with data outside 3-5 seconds zeroed out. Then you filter 'z' and add it back to 'y' to get 'Signal'. Questions: 1) Wouldn't you want to be able to use data from outside 3 to 5 seconds to generate the filtered version of the original signal corresponding to 3 to 5 seconds? 2) Wouldn't you want to remove any filter delay so the results that you add are time aligned? Dirk
Reply by ●April 11, 20072007-04-11
Thx very much for you reply. From your method, the filtered signal will have frequency discontinuity at the 3s and 5s boundary. This is not the stuff I really want. I want to make the boundary smooth between filtered area and un-filtered area. If you are using cooledit, you can select the time window signal and use the tool tp perform filtering from menu: "Transform->Filters->Quick Filters...". It won't add new samples during filtering. At the same time the samples will not change out of the desired filtered area. Do you have any comments and method? Thanks Kevin> >Lets call your signal x. > >Extract the signal from 3s to 5s (call it y). >Padd y with zeros and subtract x from y: > >z= x - ypad (where ypad is y padded to be equal in length with x). > >Filter y (call it yf) and add it to z to get the final result: > > >Signal= z + yf > > > >On Apr 11, 9:29 am, "dingke1980" <dingke1...@hotmail.com> wrote: >> For example, for a given 10s length audio signal. I want to perform LP >> filter from 3s to 5s time window. At the same time, the total sample >> number should remain the same with the un-processed one. Who can tellme>> how to do it? >> >> Thx very much. >> Kevin >> >> _____________________________________ >> Do you know a company who employs DSP engineers? >> Is it already listed athttp://dsprelated.com/employers.php? > > >_____________________________________ Do you know a company who employs DSP engineers? Is it already listed at http://dsprelated.com/employers.php ?
Reply by ●April 11, 20072007-04-11
On Apr 11, 11:46 am, "dbell" <bellda2...@cox.net> wrote:> On Apr 11, 10:54 am, "Ikaro" <ikarosi...@hotmail.com> wrote: > > > > > > > Lets call your signal x. > > > Extract the signal from 3s to 5s (call it y). > > Padd y with zeros and subtract x from y: > > > z= x - ypad (where ypad is y padded to be equal in length with x). > > > Filter y (call it yf) and add it to z to get the final result: > > > Signal= z + yf > > > On Apr 11, 9:29 am, "dingke1980" <dingke1...@hotmail.com> wrote: > > > > For example, for a given 10s length audio signal. I want to perform LP > > > filter from 3s to 5s time window. At the same time, the total sample > > > number should remain the same with the un-processed one. Who can tell me > > > how to do it? > > > > Thx very much. > > > Kevin > > > > _____________________________________ > > > Do you know a company who employs DSP engineers? > > > Is it already listed athttp://dsprelated.com/employers.php?-Hide quoted text - > > > - Show quoted text - > > It appears that > 'z' is 'x' with data from 3-5 seconds zeroed out, and > 'y' is 'x' with data outside 3-5 seconds zeroed out. > > Then you filter 'z' and add it back to 'y' to get 'Signal'. > > Questions: > 1) Wouldn't you want to be able to use data from outside 3 to 5 > seconds to generate the filtered version of the original signal > corresponding to 3 to 5 seconds? > 2) Wouldn't you want to remove any filter delay so the results that > you add are time aligned? > > Dirk- Hide quoted text - > > - Show quoted text -Sorry, typos (god I hate that), I'll try again: It appears that 'z' is 'x' with data from 3-5 seconds zeroed out, and 'y' is 'x' with data outside 3-5 seconds zeroed out. Then you filter 'y' and add it back to 'z' to get 'Signal'. (TYPOS WERE HERE) Questions: 1) Wouldn't you want to be able to use data from outside 3 to 5 seconds to generate the filtered version of the original signal corresponding to 3 to 5 seconds? 2) Wouldn't you want to remove any filter delay so the results that you add are time aligned? Dirk
Reply by ●April 11, 20072007-04-11
>On Apr 11, 11:46 am, "dbell" <bellda2...@cox.net> wrote: >> On Apr 11, 10:54 am, "Ikaro" <ikarosi...@hotmail.com> wrote: >> >> >> >> >> >> > Lets call your signal x. >> >> > Extract the signal from 3s to 5s (call it y). >> > Padd y with zeros and subtract x from y: >> >> > z= x - ypad (where ypad is y padded to be equal in length with x). >> >> > Filter y (call it yf) and add it to z to get the final result: >> >> > Signal= z + yf >> >> > On Apr 11, 9:29 am, "dingke1980" <dingke1...@hotmail.com> wrote: >> >> > > For example, for a given 10s length audio signal. I want to performLP>> > > filter from 3s to 5s time window. At the same time, the totalsample>> > > number should remain the same with the un-processed one. Who cantell me>> > > how to do it? >> >> > > Thx very much. >> > > Kevin >> >> > > _____________________________________ >> > > Do you know a company who employs DSP engineers? >> > > Is it already listed athttp://dsprelated.com/employers.php?-Hidequoted text ->> >> > - Show quoted text - >> >> It appears that >> 'z' is 'x' with data from 3-5 seconds zeroed out, and >> 'y' is 'x' with data outside 3-5 seconds zeroed out. >> >> Then you filter 'z' and add it back to 'y' to get 'Signal'. >> >> Questions: >> 1) Wouldn't you want to be able to use data from outside 3 to 5 >> seconds to generate the filtered version of the original signal >> corresponding to 3 to 5 seconds? >> 2) Wouldn't you want to remove any filter delay so the results that >> you add are time aligned? >> >> Dirk- Hide quoted text - >> >> - Show quoted text - > >Sorry, typos (god I hate that), I'll try again: > >It appears that >'z' is 'x' with data from 3-5 seconds zeroed out, and >'y' is 'x' with data outside 3-5 seconds zeroed out. > >Then you filter 'y' and add it back to 'z' to get 'Signal'. (TYPOS >WERE HERE) > >Questions: >1) Wouldn't you want to be able to use data from outside 3 to 5 >seconds to generate the filtered version of the original signal >corresponding to 3 to 5 seconds? >2) Wouldn't you want to remove any filter delay so the results that >you add are time aligned? > >Dirk > >As I said in the previous reply, I just want to implement the function like cooledit does to perform filtering on part of the audio signal. The requirement is: (1)Not generate new sample due to the filtering. (2) Should frequency smoothing b/t the boundary. (3) Don't change the value outside the time window. Could you help me? Thanks Ke _____________________________________ Do you know a company who employs DSP engineers? Is it already listed at http://dsprelated.com/employers.php ?
Reply by ●April 11, 20072007-04-11
On Apr 11, 12:02 pm, "dingke1980" <dingke1...@hotmail.com> wrote:> >On Apr 11, 11:46 am, "dbell" <bellda2...@cox.net> wrote: > >> On Apr 11, 10:54 am, "Ikaro" <ikarosi...@hotmail.com> wrote: > > >> > Lets call your signal x. > > >> > Extract the signal from 3s to 5s (call it y). > >> > Padd y with zeros and subtract x from y: > > >> > z= x - ypad (where ypad is y padded to be equal in length with x). > > >> > Filter y (call it yf) and add it to z to get the final result: > > >> > Signal= z + yf > > >> > On Apr 11, 9:29 am, "dingke1980" <dingke1...@hotmail.com> wrote: > > >> > > For example, for a given 10s length audio signal. I want to perform > LP > >> > > filter from 3s to 5s time window. At the same time, the total > sample > >> > > number should remain the same with the un-processed one. Who can > tell me > >> > > how to do it? > > >> > > Thx very much. > >> > > Kevin > > >> > > _____________________________________ > >> > > Do you know a company who employs DSP engineers? > >> > > Is it already listed athttp://dsprelated.com/employers.php?-Hide > quoted text - > > >> > - Show quoted text - > > >> It appears that > >> 'z' is 'x' with data from 3-5 seconds zeroed out, and > >> 'y' is 'x' with data outside 3-5 seconds zeroed out. > > >> Then you filter 'z' and add it back to 'y' to get 'Signal'. > > >> Questions: > >> 1) Wouldn't you want to be able to use data from outside 3 to 5 > >> seconds to generate the filtered version of the original signal > >> corresponding to 3 to 5 seconds? > >> 2) Wouldn't you want to remove any filter delay so the results that > >> you add are time aligned? > > >> Dirk- Hide quoted text - > > >> - Show quoted text - > > >Sorry, typos (god I hate that), I'll try again: > > >It appears that > >'z' is 'x' with data from 3-5 seconds zeroed out, and > >'y' is 'x' with data outside 3-5 seconds zeroed out. > > >Then you filter 'y' and add it back to 'z' to get 'Signal'. (TYPOS > >WERE HERE) > > >Questions: > >1) Wouldn't you want to be able to use data from outside 3 to 5 > >seconds to generate the filtered version of the original signal > >corresponding to 3 to 5 seconds? > >2) Wouldn't you want to remove any filter delay so the results that > >you add are time aligned? > > >Dirk > > As I said in the previous reply, I just want to implement the function > like cooledit does to perform filtering on part of the audio signal. The > requirement is: (1)Not generate new sample due to the filtering. (2) > Should frequency smoothing b/t the boundary. (3) Don't change the value > outside the time window. > > Could you help me? >are clicks a problem? you might want to consider some graceful ramping at the beginning and end of your selection (i would not call that a "window", not yet). r b-j
Reply by ●April 11, 20072007-04-11
> Questions: > 1) Wouldn't you want to be able to use data from outside 3 to 5 > seconds to generate the filtered version of the original signal > corresponding to 3 to 5 seconds?Sure that's another way to do it. But you would have to make sure that the output of the filter is cropped to the same length of the 3-5 s signal. Alternatively you could also set the initial conditions of the filter.> 2) Wouldn't you want to remove any filter delay so the results that > you add are time aligned?Yes that is correct
Reply by ●April 11, 20072007-04-11
On Apr 11, 12:02 pm, "dingke1980" <dingke1...@hotmail.com> wrote:> >On Apr 11, 11:46 am, "dbell" <bellda2...@cox.net> wrote: > >> On Apr 11, 10:54 am, "Ikaro" <ikarosi...@hotmail.com> wrote: > > >> > Lets call your signal x. > > >> > Extract the signal from 3s to 5s (call it y). > >> > Padd y with zeros and subtract x from y: > > >> > z= x - ypad (where ypad is y padded to be equal in length with x). > > >> > Filter y (call it yf) and add it to z to get the final result: > > >> > Signal= z + yf > > >> > On Apr 11, 9:29 am, "dingke1980" <dingke1...@hotmail.com> wrote: > > >> > > For example, for a given 10s length audio signal. I want to perform > LP > >> > > filter from 3s to 5s time window. At the same time, the total > sample > >> > > number should remain the same with the un-processed one. Who can > tell me > >> > > how to do it? > > >> > > Thx very much. > >> > > Kevin > > >> > > _____________________________________ > >> > > Do you know a company who employs DSP engineers? > >> > > Is it already listed athttp://dsprelated.com/employers.php?-Hide > quoted text - > > >> > - Show quoted text - > > >> It appears that > >> 'z' is 'x' with data from 3-5 seconds zeroed out, and > >> 'y' is 'x' with data outside 3-5 seconds zeroed out. > > >> Then you filter 'z' and add it back to 'y' to get 'Signal'. > > >> Questions: > >> 1) Wouldn't you want to be able to use data from outside 3 to 5 > >> seconds to generate the filtered version of the original signal > >> corresponding to 3 to 5 seconds? > >> 2) Wouldn't you want to remove any filter delay so the results that > >> you add are time aligned? > > >> Dirk- Hide quoted text - > > >> - Show quoted text - > > >Sorry, typos (god I hate that), I'll try again: > > >It appears that > >'z' is 'x' with data from 3-5 seconds zeroed out, and > >'y' is 'x' with data outside 3-5 seconds zeroed out. > > >Then you filter 'y' and add it back to 'z' to get 'Signal'. (TYPOS > >WERE HERE) > > >Questions: > >1) Wouldn't you want to be able to use data from outside 3 to 5 > >seconds to generate the filtered version of the original signal > >corresponding to 3 to 5 seconds? > >2) Wouldn't you want to remove any filter delay so the results that > >you add are time aligned? > > >Dirk > > As I said in the previous reply, I just want to implement the function > like cooledit does to perform filtering on part of the audio signal. The > requirement is: (1)Not generate new sample due to the filtering. (2) > Should frequency smoothing b/t the boundary. (3) Don't change the value > outside the time window. > > Could you help me? > > Thanks > Ke > > _____________________________________ > Do you know a company who employs DSP engineers? > Is it already listed athttp://dsprelated.com/employers.php?- Hide quoted text - > > - Show quoted text -Ke, If you are really happy with what CoolEdit is doing, you should develop a few simple experiments to find out EXACTLY what CoolEdit is doing, and do the same. It might save you from reinventing the wheel. Decide what questions you need answered and design CoolEdit input test signals appropriately to get the results you need. Dirk
