C / C++ / MFC

jeremie.delaitre19-Jun-08 23:22

19-Jun-08 23:22

I finally found a website which speaks the same langage than mine :

http://www.dsptutor.freeuk.com/dfilt1.htm[^]

Now that I am smarter, I will try to recapitulate Big Grin | :-D

The transfer function gives me the "filtered values". I must apply it on all my values to get the filtered ones.
To apply the transfer function, I need to compute the value of each coefficient which depend of the type of the filter (Bessel, Butterworth...) and the order.
Next I apply the function on my values using the coefficients in a simple algorithm.

The problem is that what I said here seems to work only for constant sampling. In my case, the elapsed time between two values is not always the same. How can I handle that ?

And another big question is : where the "low-pass thing" enter the game ? Does it mean that for low-pass Bessel filter I need to apply the Bessel transfer function on value which are to close ? or does it mean that the coefficient are different from a Bessel low-pass filter and a Bessel high-pass filter ?

Nelek20-Jun-08 0:29

Nelek

20-Jun-08 0:29

"The transfer function gives me the "filtered values". I must apply it on all my values to get the filtered ones."

Yes.

"To apply the transfer function, I need to compute the value of each coefficient which depend of the type of the filter (Bessel, Butterworth...) and the order."

correct

"Next I apply the function on my values using the coefficients in a simple algorithm."

You can do it at the same time you are getting the value as well. Under certain conditions (see below)

"The problem is that what I said here seems to work only for constant sampling."

This will most probably be because the constant sampling is given by the frecuency of the signal. but...

"In my case, the elapsed time between two values is not always the same. How can I handle that ?"

The moment you are taking the values (sampling) is not so important. The important thing is that you have to get the transfer function of the signal you are meassuring. Filters can only be used in periodical signals, that means time-repetitive working patrons. Once you have the function that define your signal... A simple transfer function if the signal is pure or a fourier-series function if your signal is a mix of different signals (this is not a must, but having it separated makes it easier). Then you just apply the filter for the value at that moment. Frecuency of a signal should be constant, so you can use "t" (time) or "s" (time in laplaced functions) parameters insteads of using "f".

"Does it mean that for low-pass Bessel filter I need to apply the Bessel transfer function on value which are to close?"

If your signal is not periodical... you must not apply filters. If your signal is periodical... you should not use it that way. Example: You have a pure sinus signal centered in the zero axis with amplitude top at 1 and bottom is -1 (don't remember if it is amplitude 1 or 2). With frecuency 1 Hz, ok? Then ONCE a second you will be in the same point of your graphic, (i.e. every full second at the top of the function value 1).

then your sampling will have always the same pattern, time 1s -> value 1; time 1,25s -> value 0; time 1,5s -> value -1; time 1,75s -> value 0; time 2s -> value 1

and so on with the intermediate values...

Then if you apply a filter which cutoff frecuency is near 1Hz you will have an attenuation of the amplitude but the frecuency will be the same, let's say that instead of 1 and -1 now you have 0,7 and -0,7.

Now if we make the signal a bit more difficult, let's say... a combination of that simple one, and another sinus with frecuency 10Hz and amplitude 0.25, -0.25 then you will have a little and fast sinus wave centered in the first one going up and down.

If we complicate it a bit more adding a slower signal with a big amplitude you will have something more strange as result. And this is the principle of Fouriers theoreme. As long as a signal periodical is, there is always the possibility to get that signal as a combination of different independant signals with different amplitudes and frecuencies, that are called (armónicos on spanish) ???armonics???. So when you apply a filter the result of the original signal will be a bit weird to be seen or imagined, but if you apply the filter to the fourier serie it will be easily recognised how every one of them change and the result will be the same.

The biggest problem you are going to have is to delimitate your scenario, find out the function (or addition of functions) of your signal and choose the right filter and its best order to fit your problem. Once made that, to apply the algorithm of the filter into your programm should be so easy as apply a mathematical formula to a parameter.

Regards.
--------
M.D.V. Wink | ;)

If something has a solution... Why do we have to worry about?. If it has no solution... For what reason do we have to worry about?
Help me to understand what I'm saying, and I'll explain it better to you
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Rating helpfull answers is nice, but saying thanks can be even nicer.

jeremie.delaitre20-Jun-08 2:49

20-Jun-08 2:49

Ok, I think I am close to get it.

If the user gives me :
- the data with their corresponding time stamps (corresponding to values picked from a periodical signal)
- the frequency of the input signal
- the wanted type of low pass filter (Bessel or Butterworth)
- the wanted order of the filter he wants
- the wanted cutoff frequency

I do :
- compute the coefficient of the lowpass filter according to the type of the filter, the order and the cutoff frequency. These coefficients represent the transfer function of the filter.
- decompose my signal into pure ones (Fourrier series) (I do not search informations on this for now)
- apply the transfert function on each pure signal
- build the filtered global signal from each filtered pure signal
- get the filtered value for each of my time stamp from the filtered global signal

Is that right ? (please tell me it is !!)

Nelek20-Jun-08 3:09

Nelek

20-Jun-08 3:09

More or less is right. The only problem is that if the user just gives you discrete meassures from the signal with the timestamps... it is going to be pretty difficult to imagine how the signal is. You will need A LOT of points to be able to imagine/extrapolate how is the function that fits that values in that time. Very hard, but needed (in my opinion).

If it is possible that the user gives the function that determine the curve of the original signal will be much much much easier.

Steps about the fourier serie can be omited, but in some scenarios will may make things easier.

More or less the main points are correct (or at least that would be my approach as well).

I am sorry that I don't have my notes from the university when I learned it, I had very good points and examples in them. 8 years and 2000km lapse are guilty. Sigh | :sigh:

Regards.
--------
M.D.V. Wink | ;)

jeremie.delaitre20-Jun-08 3:23

Re: http://www.musicdsp.org

20-Jun-08 3:23

Well, I can say now : it is pretty clear !

I understand the problem of the input signal but it is not possible to have the function. The data provides from experimental measurements.

As you say in your other post, I will ask a clear specification of what my client wants.

In any cases, I have learned a lot of things so it was not time loss.

I really want to thank you again...so..thank you !

http://www.musicdsp.org

div!sion20-Jun-08 0:24

div!sion

20-Jun-08 0:24

Have a look at http://www.musicdsp.org/archive.php?classid=3#237

jeremie.delaitre20-Jun-08 3:10

20-Jun-08 3:10

I will have a look, thanks.

cp987620-Jun-08 0:55

cp9876

20-Jun-08 0:55

jeremie.delaitre wrote:
The sampling rate of my data is not constant (for example, I get a value at 1s, then 1.2s, then 1.8s...).

In that case you have to think very carefully what your data actually represents, and it is not clear from your post. Applying any of the standard linear filtering techniques will cause the amplitude of the signal to increase in areas where there are more samples, this is a simple consequence of linearity. The standard filters, like the ones you mention, are defined in the frequency domain, so there is an inherent assumption of linearity.

It may help your thinking to consider what should happen if you add a second sample very close to one of your existing ones. If you would expect the effect of this to be almost identical to doubling the amplitude of the original sample, then proceeed with the linear filtering option.

If you expect the effect of a second sample to simply increase the certainty with which you know the sample value, then what you really want is some form of curve fitting, and for this numerical recipes would be a good starting point.

Peter
"Until the invention of the computer, the machine gun was the device that enabled humans to make the most mistakes in the smallest amount of time."

jeremie.delaitre20-Jun-08 3:01

20-Jun-08 3:01

Yes you are right, I must think about it.
I understand the problem now as I understand a little bit more linear filtering.
I do not know if this if the case of my client, so I will ask him Smile | :)

In fact, I think he wants really linear filtering, but I did not explain me the problem very precisely...

Thanks for pointing me that problem.

Terminal Server [modified]

Nelek20-Jun-08 3:12

Nelek

20-Jun-08 3:12

I strongly recommend you to speak with your client and make the guidelines as much clear as you can get them, with all the specifications that he wants, how the inputs are going to be and how the outputs should be.

If linear filtering is the point... you can pretty forget about my other posts :P Roll eyes | :rolleyes:

Regards.
--------
M.D.V. Wink | ;)

subramanyeswari19-Jun-08 19:53

19-Jun-08 19:53

Hi,

How do i check whether terminal server is enabled or not programmatically(VC++ or winSDK)
I am using GetSystemMetrics( SM_REMOTESESSION ) API to know whether terminal server is running or not. But on compiling i am getting 'SM_REMOTESESSION' : undeclared identifier compiler error. My environment is windows 2003 server VS6.0 with sp6.0

Thanks,

modified on Friday, June 20, 2008 2:59 AM

Naveen19-Jun-08 21:38

Naveen

19-Jun-08 21:38

subramanyeswari wrote:
But on compiling i am getting 'SM_REMOTESESSION' : undeclared identifier compiler error. My environment is windows 2003 server VS6.0 with sp6.0

You need to install the latest platform SDK to get the update header file. The value of SM_REMOTESESSION is defined as 0x1000. So if you dont want to download, just define SM_REMOTESESSION in your code.

#define SM_REMOTESESSION        0x1000

nave

[OpenedFileFinder]

subramanyeswari19-Jun-08 22:58

Re: Terminal Server [modified]

19-Jun-08 22:58

thank you

subramanyeswari20-Jun-08 0:18

20-Jun-08 0:18

I am reading the registry to check whether terminal server is enabled or not in the following path
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\Terminal Server
\TSEnabled.

How to disable Terminal Server. Is there any other way.
My working environment is windows 2003 server. I am using Manage your server>Add or remove role.

But still the value of TSEnabled is 1. Why it is not updating the registry

modified on Friday, June 20, 2008 7:01 AM

KarstenK20-Jun-08 3:20

KarstenK

20-Jun-08 3:20

You better look in the SDK where SM_REMOTESESSION is defined. There you see for what systems you got to compile. I guess NT >= 5, and SP4. Change in the stdafx.h.

Check error conditions carefully. The strangest is most to cut the connection by pluging out the network cable. (This belongs to our Release Tests)

If you should consider to use a never Visual Studio.

Greetings from Germany

subramanyeswari20-Jun-08 3:52