Filter a waveforms

Hi everybody..
is there a simply way to filter a waveform (or an array of DBL) in the frequency domain if i have a transfer function?? Let's suppose my transfer functions is:
1/(1.2 + 0.5x + 0.7x^2... +...) that rappresent the frequency response of a system that i want to rapresent. If i want to filter any other signal like it pass through my system i have to implement the trasfer function. Have you got any idea?? I post i .vi i realized with the digital filter module, but when i insert high value for the filter denominator the response look a little bit strange... do you have any oder solution for this??
Thank You
Nigeltorque
Attachments:
Filter a waveform.vi ‏55 KB

Hello NigelTorque,
could you please take a look at this forum? I think you will find the information useful.
http://forums.ni.com/t5/Signal-Generators/Amplifier-Design/m-p/1000688
Antonios

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How do I use adaptive LMS filter with waveforms?

In the attached image file and VI, I am reading a .lvm file which contains abdominal ECG (AECG). AECG contains both maternal and fetal ECGs i.e. MECG and FECG respectively. I'm trying to input AECG and MECG to the LMS FIR filter so that I can get FECG as the error signal, e(n). In my VI, e(n) is coming out as a 16-Bit Double but I want it to come out as a waveform. How do I go about it?
NOTE: Since this forum doesn't allow uploading .lvm files, I have share in Google Drive here. You can use this file to run the attached VI.
Attachments:
AECG_NI.vi ‏38 KB

Step 1. Dont use the express VI. It makes assumptions about the data that don't appear to be valid.
Mike...
PS: The lvm is just a text file. Change the file extension to txt and upload away.
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Issues with signal input adaptive filtering, it won't filter the added noise from the input waveform

Hello,
I have an issue with my adaptive noise cancellation program. Essentially I want to input a custom wav add noise to it and then filter the noise away in order to gain the custom wav again.
While doing so I want to read the learning curve and read the adaptive coefficients. Unfortunately I have an issue when it comes to the filtering of the custom wav+noise. It won't filter the signal at all.
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Attachments:
Testing.vi ‏59 KB

Hey Jan,
Thanks for the reply. I am currently using the Adaptive Filter Toolkit in order to obtain those VIs. The VIs which are in use are the LMS Adaptive Filter ones.
I figured there might be an error with the input of the array. This VI requires a DBL Array but it seems like it can not process it.
The "Get Wavefirn Components" works better now but I still have an issue with a time. I put a 9s wav file in but it only computes it in a very short time and I can not play the file while it is computing.
I added the modified program to the attachments.
This program is ought to read a waveform file add noise to it and then use an adaptive filter in order to get the orginal waveform back again and if possible either store or play the final waveform.
Thanks for your help.
Attachments:
Testing.vi ‏62 KB

Looking for simple waveform filter

I am using E series board using single ended input to collect 16 channels of constant value data, (ie 10volts). The data has a small amount of noise and outlying data points which I need to remove. I am attempting to implement a simple low pass filter. I attempted the Buterworth and FIR filters. The code I have attempted based upon the help examples unbundles the waveforms, filters the data, and rebundles the waveforms. This produces a discontinuous waveform when ploted. Addtionally the filtered data looks like a sine wave with an amplitude ranging between zero and the bias of the constant input data dependent upon the input parameters fo the filter. I'm looking for an example code that filters a constant signal with outlying da
ta points and small amount of noise. I cannnot use a average because the outlers are all on one side of the average value and will bias my data.
Thanks for the assistnance
Brian Wells
Space Research Institute
Auburn University, AL

Brian,
Can you try a running average using shift registers? With this, if the latest point is over x% outside of the running average, then do not include it. Since your signal is "constant," then frequency filters will probably not help you. See the attached example in LabVIEW 6.0.2. The interesting part of this example is that you need to know what your expected value is. If this is unknown, then one must do a bit of statistics (probably through histogramming) to get an initial average value.
Randy Hoskin
Applications Engineer
National Instruments
http://www.ni.com/ask
Attachments:
Remove_Outliers.vi ‏62 KB

How to pass digital waveform to a raised cosine filter

so i have a digital waveform like this for example
and i want to pass it through a raised cosine filter (which i can generate its coeeficents by having the digital design toolkit)
below is the shape of the VI (it only outputs
and after passing the digital through the filter i want the analog signal to look like this
So my question is how can i pass the digital signal through that raised cosine VI ??

Coninued from here http://forums.ni.com/t5/LabVIEW/Please-any-help-would-be-appreciated-how-can-i-implement-a/m-p/30998...

Read & Filter Multiple .wav files, Export Filtered SPL

Hi
I'm searching for tips on how to use LabVIEW to read multiple .wav files, then filter them, then export just the calculated SPL values to a "database friendly format".
Basically I want a solution that allows me to point to several files, then press a "go button" that triggers LabVIEW to produce filtered SPL values as exported data.
I'll skim over the cal approach since I'm confident most readers know how to handle it. Basically, though, for each 30 s .wav file of broadband noise, an associated cal tone file is used to determine the dBFS value (dB below Full Scale) to be associated with 94 dB SPL, or 114 dB SPL, as the case may be. I'll leave it at that.
Detailed Requirements, (excluding cal steps, tho).
1) Operates on up to a dozen 30 s time records (44,1 kSa/s, mono .wav files);
2) Passes the signal from each desired .wav file through 18 of the 1/3 octave filters (IEC 1260 compliant, as in the S/V toolset) not necessarily in real time!;
3) Calculates the SPL for each 1/3 octave filtered time record; This makes (12 files) x (18 filters) = 216 filtered SPL values if end-user points to a dozen broadband noise files before pressing the "Go" button;
4) Exports SPL values in a suitable database format; an existing database does the rest of the mathematics and report generation;
Client FileMaker database functions include:
- import and store 1/3 octave filtered Leq(30) SPL values;
- average filtered SPL across the declared qty of mic positions of the present test case;
NOTES:
1) In the field, we do NOT use simultaneous / MULTI-mics. Instead, it is absolutely an "open and shut case" that the optimum process is to move one (1) single mic quickly from spot to spot. While holding the airborne noise at a constant level we position the mic at each spot for 30 s to feed the .wav recorder, then move on. We emerge from a typical day of field activity with, say, 60 to 100 .wav files to post-process.
2) (1), above, explains why we don't use any N.I. DAQ h/w.
3) LabVIEW, it seems, exports best to a spreadsheet, so if needed, I'm prepared to handle manually any ugly conversion from LV's format to a proper database format ("1st Normal Form, 2nd Normal Form, etc.).
4) Given a choice to calculate the averages in my database, versus in LV (say, to cut down on exported data qty), I'd likely opt to export the 216 SPL's and then do averaging in the database, (where the ASTM E336 reports are done).
TIPS Gathered To Date:
1) S/V toolset has IEC 1260 compliant filters, BUT...
2) If you use the S/V toolset VI's, then you CANNOT ACCESS the filtered time record NOR CAN YOU WRITE the filtered data back to a .wav file! All you can do is measure the filters' outputs. Fortunately for us here, the available measures include SPL.
Aside: Although not a show-stopper in this project, this constraint is surprising, nonetheless. To access & record the filtered sound, you'd have to either (a) ask N.I. to unprotect VI's, or (b) develop your own IEC compliant filters from N.I.'s published tap co-eff's etc.;
3) LabVIEW does not support drag'n'drop operations on files. Instead, the recommended technique is to use multiple "Browse to/ Navigation/ File Open" panels to allow LV to open the user's desired qty of .wav files simultaneously;
Thanks for any tips!

The filter VI is password protected in the Sound & Vibration Toolkit (SVT) for two reasons. First, since the filter is compliant to a standard, by locking the code we guarantee that no one has the chance to accidentally change it so that they are no longer compliant. Second, the method of filtering the data is considered intellectual property that we are protecting.
That said, from reading your post it does not appear that you need to have access to the filtered data since you are making an SPL measurement which is provided in the Sound & Vibration functionality. Is that correct?
Also, I wanted to point you to another subVI which comes with SVT but is not in the pallette that might be of use. There is an example for SVT which is SVXMPL_Wav Power Spectrum which opens up a wav file and computes the power spectrum of it. Inside the VI, there is a subVI which will open a Wav file and convert it into a waveform which you can use with your SVT functions.
Hope that helps,
Jack

IIR Filtering and response .vi: Butterworth filter magnitude response depends on sampling rate -why?

Hi folks,
I am not expert in filter design, only someone applying them, so please can someone help me with an explanation?
I need to filter very low-frequent signals using a buttherwoth filter 2. or 3. order as bandpass 0.1 to 10 Hz .
Very relevant amplitudes are BELOW 1 Hz, often below 0.5 Hz but there will be as well relevant amplitudes above 5 Hz to be observed.
This is fixed and prescribed for the application.
However, the sampling rate of the measurement system is not prescribed. It may be between say between 30 and 2000 Hz. This will depend on whether the same data set is used for analysing higher frequencies up to 1000 Hz of the same measurement or this is not done by the user and he chooses a lower sampling rate to reduce the file sizes, especially when measuring for longer periods of several weeks.
To compare the 2nd and 3rd order's magnitude response of the filter I used the example IIR Filtering and response .vi:
I was very astonished when I the found that the magnitude response is significantly influenced by the SAMPLING RATE I tell the signal generator in this example vi.
Can you please tell me why - and especially why the 3rd order filter will be worse for the low frequency parts below 1 Hz of the signal. I was told by people experienced with filters that the 3rd oder will distort less the amplitudes which is not at all true for my relevant frequencies below 1 Hz.
In the attached png you see 4 screenshots for 2 or 3 order and sampling rate 300 or 1000 Hz to show you the varying magnitude responses without opening labview.
THANK YOU for your ANSWERS!!!
chris
Solved!
Go to Solution.
Attachments:
butterworth-filter-differences.png ‏285 KB

Hello Lynn,
thanks for the answer. You are right that there are few points "behind" the curve in the graph, see png.
However, this is the filter response which Labview (2009) provides to me directly out of the "IIR Filter for 1 Channel. vi" in the "filter information" output cluster. Where up to now I do not know how to influence it - apart from adjusting the input parameters "IIR filter specifications". OK, I assume I have to gain more knowledge of this. The curve of the magnitude resonse dies not change when I change the number of samples of the input signal of the signal generator, only wehn I change the sampling rate.
I used directly the example vi from Labview with the name indicated in my first post "IIR Filtering and Response.vi".
So I assumed that everybody has it in his/her examples shipped with LV and it is not necessary to post it.
I just adjusted the size of the diagram of magnitude response to see the curves better as you see in the attached vi.
So I did no changes to the vital parts of signal generation and filter of the example. The screenshots are like they come from the example when using the option "one waveform" where I as user assume that this which is behind is quality-controlled by NI.
I was also astonished that the filter magnitude response is different to the one I copied out of graphs 1 year ago - but I unfortunately cannot reconstruct which example I used there...
Thanks for any further comments
chris
Attachments:
IIR Filtering and Response_CH.vi ‏55 KB
butterworth2nd_order_bandpass_0p1to10Hz_mag_response.PNG ‏18 KB

What am I doing wrong with this filter (counter/frequency issue), probably another newb question.

I extracted the part of my VI that applies here. I have a 6602 DAQ board reading a counter for frequency, using a Cherry Corp proximity sensor. Getting a lot of noise and errant ridiculously high readings. Speed of shaft which it's measuring is currently 2400rpm with one pulse per revolution so 40hz.
Trying to use the express filter VI to clean up the signal and ignore anything over, say, 45hz and under 35hz. No matter what setting I choose I continually get the 20020 error, Analysis: The cut-off frequency, fc, must meet: 0 <= fc <= fs/2. I know this relates to sample period somehow, but for the life of me I can't understand what I'm doing wrong.
I used this VI without filtering on bench tests with a hand-drill and got perfect output every time. Now it's on the machine and being erratic. Any help here will ease my stress level significantly, thanks.
VI attached
Still confused after 8 years.
Attachments:
RPM.vi ‏92 KB

Hello Ralph,
I'm not sure about mounting your sensor to your rig, but I can provide a couple ideas about the filtering. Depending on the type of noise, the digital filters on the PCI-6602 could help eliminate the behavior you are seeing. If the noise manifests as a "glitches" or a bouncing signal, you could use another counter with a minimum period to help eliminate the noise. This concept is discussed in greater detail in this KnowledgeBase. I noticed that you are using NI-DAQmx; the practical application of the digital filters on the PCI-6602 in NI-DAQmx is discussed in this KnowledgeBase. A more detailed description of the behavior of these filters is provided in the NI-DAQmx Help (Start>>All Programs>>National Instruments>>NI-DAQ) in the book entitled "Digital Filtering Considerations for TIO-Based Devices".
I also wanted to comment on your original post and explain why you were receiving error -20020. For convenience, I have copied the text of the error code below.
Error -20020 occurred at an unidentified location
Possible reason(s):
Analysis: The cut-off frequency, fc, must meet: 0 <= fc <= fs/2.
I think you may have misunderstood exactly what the Filter express VI does. The Filter express VI takes a series of values in a waveform and performs filtering on those signals. So, it will look at a waveform made up of X and Y values and apply the defined filter to this waveform. Specifically in your application, the cut-off frequency (fc) is the Upper Cut-Off level that you specified in the Filter express VI; any frequency components of the waveform outside of the range you have defined will be filtered. The fs is the sample rate based on the data that you wire to the Signal input of the Filter express VI. In your VI, this data is coming from the DAQ Assistant. So, fs will be the sample rate of the DAQ Assistant and is associated with the rate at which points are acquired. The sample rate does NOT relate to the bandwidth of the signal, except that the Nyquist theorem tells us that the sample rate must be at least twice the signal bandwidth in order to determine periodicity of the signal. So, in this case, the sample rate of the DAQ Assistant would need to be at least double the high cut-off frequency.
However, you are performing a frequency measurement using a counter. I think this is where the confusion comes in. For the frequency measurement using a counter, the DAQ Assistant returns a decimal value which represents the rate of the pulse train being measured by the counter. It does not return the actual waveform that is being read by the counter. It is this waveform that would be band-pass filtered to eliminate frequency content outside of the filter's bandwidth. Instead of the Filter express VI, I would recommend that you simply discard values that are outside the range you specify. I have modified the code you posted earlier to perform this operation. The image below shows the changes that I made; rather than using the Filter express VI, I simply compare the frequency to the "Low Threshold" and "High Threshold". I use a Case structure to display the value on if it is within the limits. Otherwise, I display a "NaN" value. I have also attached the modified VI to this post.
Message Edited by Matt A on 09-04-2007 07:58 PM
Matt Anderson
Hardware Services Marketing Manager
National Instruments
Attachments:
RPM (Modified).JPG ‏17 KB
RPM (modified).vi ‏72 KB

Problems with waveform integration

What I need to do is waveform integration from an acceleration vibration signal to a velocity or displacement signal.
To take the signal to velocity, I first filter the acceleration signal (HPFilter, 5Hz) to remove any offset and the perform the integration.
To probe the method, I use a signal composed by one sine WF with amplitude 1. The test consist in vary the frequency of the sine WF and evaluate the value obtained by the FFT of the integrated signal.
For a freq of 10 Hz, i obtained a value of 0,015931 which is nearly the expected 1/(2·pi·10)=0,015916
for a freq of 150 Hz, i obtained a value of 9,8130e-4 instead of the expected 1/(2·pi·150)=1,061e-3
for a freq of 450 Hz, i obtained a value of 7,91922e-5 instead of the expected 1/(2·pi·450)=3,537e-4
as you can see, higher the frequency, higher the deviation from the expected value. i am triyng to get this solve, but in this moment i don't know why this phenomenon occur.
any help will be well recibed.
CJMV

OK, that's a really good point, i am agree with it.
do you have made a study of the relationship between the sampling frequency and the maximum frequency of the signal?.
i am going to make it anyway, but to compare...
another point related to the waveform integration:
i rote in an application note that it is recommended to use a FIR filter instead of an IIR because it has less phase distorition of the signal. however, if you ask for the filter information and plot the "phase H(w)", you will se a big distortion in phase introduced by the filter.
in the other hand, if you do the same thing with an IIR filter, you will find that the phase distortion is less.
conclusion: i am using an IIR filter with a Fc of 5 Hz.
greetings
CJMV

Only first plot of waveform chart has run-time menu

Hello again all you helpful forum-goers!
My multi-plot waveform chart is working decently well, but now for some reason only the first plot in the plot legend displays a run-time customization menu when I click on it while the VI is running (allowing me to select the plot color, etc.). All the other plots do nothing when I click on them. I do not have any mouse-down events or filter events of any kind in my VI. I programmatically set the number of plots and the size of the plot legend, along with each plot's name, y index, and color. Any ideas as to what might be going on?
Thanks a bunch as always!
-Joe
Solved!
Go to Solution.

Okay, hopefully this will help get some more thoughts / answers flowing. I've copied my strip chart from my actual VI into a test VI, and it still exhibits the problem. On the other hand, I have created a new strip chart from scratch, and it works fine. I have attached both VIs here for you to see. My guess is that some property of the first chart is set incorrectly or something, but I cannot figure out which one (and there are so many!).
Any assistance you can give would be appreciated. The challenge is to get the chart in the first VI working (plots other than the first responding to a mouse click to change their color or other attributes), via changing some attribute of the chart (and being able to tell me which one you changed!).
Thank you again for anyone who can look into this.
-Joe
Attachments:
Test StripChart color selection not working.vi ‏17 KB
Test StripChart color selection working.vi ‏12 KB

How to design a good 60Hz notch filter?

I tried to design a 60Hz notch filter(59-61 as stop band,tried different kind filters/orders,cont.true)But I can't have a stable output. When test on 60 Hz input,the output signal's amplitude will change in a repeated pattern,sometime the ripples is very small, but sometimes it's big, especially at the waveform beginning. The Fs is 5000, data is sampled every few seconds. Are there any refined 60Hz notch filter can be used, or any better way to design a notch filter?
Averaging is one way to remove the ripples after filtering,but I hope the filter can give a steady output.Thanks!

sfhj wrote:
please if you have this example or know about it ,send it
thanks
@sfhj
Hijacking a old thread is never a good idea. I suggest you start your own. Also in order to get help you must explain better what you have done so far, why and how you struggle. The best way to do this, is to post your code, and some signals examples. I know a lot of tricks. And none of them need a fancy toolkit. But you must start your own thread in order to get help from me.
Edit: I noticed you have started a new thread, do you want to move the discussion to that thread?
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

Using DAQ-assist to input a waveform; need help building a counter to count voltage "spikes"

Hey all! I'm pretty new to labView and even newer to this forum, but its nice to meet you all...I hope that perhaps someone can help me with my problem.
Allow me to begin by detailing the specifications of the problem. I am an undergraduate student, and have a job doing research in a MEMS (micro/nanotech) lab. The graduate student I am making this program for is working on biomedical applications; eventually, the program will be connected to a microdevice that has a tiny channel in it, cut through a wee little capacitor, which blood will run through. As red blood cells pass this capacitor, the voltage will spike; meaning that for each voltage spike, we can (and are trying to) count the number of red blood cells.
However, I am still in the early developement of the program, so this above specific info is not that important. Basically, I am using a function generator to input a waveform to the DAQ assistant of, say 500 mV. I am trying to write a program that increments a counter every time I turn the voltage above say 550 mV (peak-to-peak), counting the number of simulated "spikes." I have tried quite a lot to write a working program, and although I have not gotten it to work yet, I will post a screenshot of my most recent attempt HERE:
I thank you in advance for any helpful tips, advice, or straight up assistance you may be able to give me. Please ask me any clarifying questions about the program I wrote or the application, or anything. Happy Friday!

Hey guys, it's been a while! A lot of stuff has been happening in my life and I have had virtually no time to work on my LabView project.
I did create a LabView program based off IanW's reccomendation. I am unsure of what exactly is going wrong, but when I run it, only a simple "snapshot" of a waveform from the DAQ shows up in the graph. Even when I put the DAQ assist in a seperate while loop, the same thing happens. I am including a screenshot of the project in case I am messing something entirely different up. If you happen to read this, I really appreciate your help and thank you Ian!
I am also having a random issue with the filter signal VI. So that background signals in the actual experiment do not read as "spikes" I have been instructed to include a high-pass filter in the VI. However, everytime I use the high pass filter VI, it botches my signal and turns it into a bunch of noise! I, nor my graduate mentor (who isn't too well-versed in LabView) have any idea why this is - we've tried using different types of filters to no avail.
Lastly, I would like to talk to Peter about a few questions I had abour LabView design. In case you're still around, I will write another post later today with more detail. In the meantime, I will try to find some of the example VIs about shift registers All who read this have a great day!
Attachments:
count spikes pic.png ‏29 KB

How to extract the signal out from the waveform by my designated power level?

Dear all,
How can I extract the signal from the waveform accroding to the power level? I read the Trigger&Gate.vi, but this vi extract signal according to the duration time. I want to extract signal according to power level.
As shown in the following figures, the signal I want to process is between 130000 to 140000, if I zoom in, I can see the the useful signal is between 135400 to 138200. The question is how to extract the signal in the zone?
I tried the sub_NoiseEst_And_Chop_Shell.vi in the Packet_based_link example too, but this subvi seemed to be a little slow. Can anybody give me better advice? Thanks in advance!
Solved!
Go to Solution.

I was working on something similar but haven't had time to fully develop it.
My idea was to use an envelope detector (low pass filter) and then use an energy detection VI on the envelope.
Here's where I left off
Anthony F.
Product Marketing Engineer
National Instruments
Attachments:
test.vi ‏331 KB

High pass filter don't work

i used the following setting for the testing;
input signal 1: 2khz sinewave, amplitude 1V
input signal 2: 50Hz sinewave, amplitude 10V
the above 2 signals are added together and fed into a filter. my objective is to remove signal 1(1V sinewave). i've tried some filter but it don't seem to work. for the attached vi, i used high pass filter. the filter don't work.
i tried to view the frequency component of the signal before and after the filter and found out that both signals pass through the filter.
1. why isn't the 50Hz signal being filtered off?
i had the following values:
before filter after filter
50Hz ==> 3Hz
2kHz ==> 120Hz
the filter seem to scaled down the frequency compon
ent instead of filtering off the 50Hz.
2. why is this so, how can i solve this problem?
3. what should i take note when setting the parameters of the filter?
Attachments:
time2freq_domain.vi ‏573 KB

You might try increasing the number of FIR filter taps. Also, would an IIR filter work for you? I recoded your example using the waveform filtering VIs that were developed for LabVIEW 6i. Please look at the number of taps for the FIR filter, and the PassBand and StopBand frequencies. Also, notice the IIR filter's superior suppression of the low frequency sinusoid.
Attachments:
time2freq_domain2.vi ‏107 KB

Savitzky filter: problem writing to file

Hi,
I have a very weird problem. I m using a Savitzky Golay filter to smooth some data and I can see the correct filtered data on the graph on the front panel. Now, I want to write this filtered data to a file. So I wired the output of the SG filter to a spreadsheet file. On replotting the file, I notice that the graph that I obtain is identical to the input that I gave to the filter. I have attached some screenshots of the front panels and the replotted graph.
Figure one: Is the input to the SG filter
Figure two: Is the output of the SG filter.
Figure three: Is the waveform that I obtained from replotting the output that I wrote to a file.
Figure four: Is the SG filter and write to file VI screenshot.
Can someone please help me out. Its driving me crazy.
Thanks and regards.
Attachments:
sgfilter_writetofile.doc ‏314 KB

You index the output-array of the filter VI before you save it to file. So on the frontpanel you see the whole waveform, but you save only the first value to the file. Another evidendence for this is the messed up x-Axis (time) in the data loaded from your file. The data you see in the file is neither the data before, nor the data after the filter function. You see only the first value of each measurement. YOu have to put a for-loop arround your save to spreadsheet, or better would be to not index the array at all and write the whole array to file. You can use probes to see what I mean.
Hope this helps!
Andre

Filter a waveforms

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