Digital filter design toolkit

Me and my teacher would like to use the Labview Digital filter design toolkit. He has lot's of different CDs from Labview in his Lab. He said maybe find out if it's maybe on one of the Labview CDs we already have here The question is : Is the Labview digital filter design toolkit only on one CD, or can we find it somewhere on an other Labview CD ?

Hello,
The DFDT is sold sperately but it could also be bundled in a Developer's Suite. Most of the time its a seperate CD. So should check to see if you have a CD for DFDT or perhaps a LabVIEW Toolkits CD.
Regards,
Chetan K
AE-NIC

Similar Messages

Digital Filter Design Toolkit for LabVIEW 8.0

I need Digital Filter Design Toolkit for LabVIEW 8.0 and I cannot find it on the site.Where can I get it from?

Contact your local NI representative. He might just sent you one.

Install Digital filter design toolkit on labview 8

I get an error that says:
You must install labview FDS or PDS 7.1 (or later) before
running Digital Filter Design Toolkit 7.5 Setup.
I have only Labview 8.0 installed.

Hi storeytime,
Are you running LabVIEW in evaluation mode or as Base Version, than this could be the reason why it doesn't work.
You have to activate LabVIEW as Full Development System (FDS) or Proffessional Development System (PDS) to install the DFD Toolkit.
If you have allready activated LabVIEW as FDS or PDS you need to start it once to get the registry updatet, then the installation of the Toolkit should be possible.
Regards, Christian

How to interpret the Coefficient Report from the FXP Digital Filter Design

Hi, I tried to design a low pass IIR filter in cascaded form II. I then pass the filter into the filter structure conversion VI and then pass the filter into the FXP conversion and get the Coefficient Report.
However, I got problem understanding the report. The index of the coefficient does not correspond to the description of the Filter structures in the Labview help:
In the labview help where they describe the filter structures, the coefficients have two indices, one for the stage, and the other for intra stage.
In contrast, the report only gives out coefficients with one indicices.
My question is: how to map the coefficients to the structure.
Attached is the digital filter structures appeared in the Labview help.
The following is the report:
            Reference Value               Quantized Value              Note
Forward Coefficients:
b[0]       +1.0105197822678145E-6       +1.0104849934577942E-6
b[1]       +2.0210395645356291E-6       +2.0210864022374153E-6
b[2]       +1.0105197822678145E-6       +1.0104849934577942E-6
b[3]       +1.0105197822678145E-6       +1.0104849934577942E-6
b[4]       +2.0210395645356291E-6       +2.0210864022374153E-6
b[5]       +1.0105197822678145E-6       +1.0104849934577942E-6
number of Overflows:    0
number of Underflows:   0
number of Zeros:        0
Reverse Coefficients:
a[0]       -1.9962855130669792E+0       -1.9962768554687500E+0
a[1]       +9.9628955294911659E-1       +9.9627685546875000E-1
a[2]       -1.9984573661467793E+0       -1.9984741210937500E+0
a[3]       +9.9846141042409486E-1       +9.9847412109375000E-1
number of Overflows:    0
number of Underflows:   0
number of Zeros:        0
gain       +1.0000000000000000E+0       +1.0000000000000000E+0       Floating-point
* Gain processing is on the host
Attachments:
IIRcascaded.GIF ‏7 KB

Hi David,
The following Developer Zone article discusses important concepts involved in utilizing the digital filter design toolsets. It specifically address the form in which the design coefficients are expressed.
I hope this helps.
Cheers,
Emilie Kopp | National Instruments

Simulating a Digital Filter using Z-Transform

Good evening!
I have some sampled data logged in text (.lvm) format which I would like to perform off-line digital filtering to.
I have a Z-transform of the form ( 1 + a * z^-1 )/( 1 + b * z^-1 ) to describe the behaviour for the given sampling rate.
I will use the "Read from Measurement File" to import to LabVIEW as dynamic data, I can then arrange that in an array if need be. I am just unsure of how to implement the z-transform; are there any examples performing a similar function?
I apologise for the "can you help me?" type post but I have struggled to find information on building this.
Any help/references appreciated!
Peter
Regards,
Peter D

I used the "NI LabVIEW Digital Filter Design Toolkit" and the difference equation method to create my filter. But I am getting some odd results when I apply my input data to the filter.
I have made some notes on the VI to what I am trying to do.
Basically trying to reduce the response time of a first-order system using filtering; the filter will cancel out the previous time constant and replace it with a new one. I have pasted the bode plot of the system into my VI as there is no Flash on my computer to upload on here.
I am still trying to get my head around what is happening - the noise content of my real input signal could be having an adverse effect.
[ edit : attaching the VI ]
Peter
Message Edited by PjayD on 04-14-2010 09:38 AM
Regards,
Peter D
Attachments:
Convert_LVM_Array.vi ‏223 KB

CIC filter design and implementation

I need a CIC (cascaded integrator-comb) filter - see here
How can I design and implement it in CVI?
Thanks
Vix
In claris non fit interpretatio
Using LV 2013 SP1 on Win 7 64bit
Using LV 8.2.1 on WinXP SP3
Using CVI 2012 SP1 on Win 7 64bit, WinXP and WinXP Embedded
Using CVI 6.0 on Win2k, WinXP and WinXP Embedded

Hi,
Although there is a specific toolkit named Digital Filter Design Toolkit in LabVIEW, there isn't a specific toolkit for CVI.
You can see these document for more info:
http://www.ni.com/white-paper/52336/en/
http://www.ni.com/white-paper/14784/en/
http://www.ni.com/white-paper/2859/en/
http://zone.ni.com/reference/en-XX/help/371988F-01/lvdfdtconcepts/cic_filters/

Designing digital filter for analog output

Hi all,
Iam working on pci-6723 NIDAQMX 7.3 LABVIEW 7.1 ENVIRONMENT.I have designed 16 channel data acquisition system.in order to supress glitches in the analog output signal ,which filter i have to adopt to the analog output generation. since i dont want to design hardware low pass filter , if i adopted digital filter will it improve the output signal performance ,kindly suggest me.with some example.
Regards
labview boy

Hello,
If you are talking about a software digital filter with the program that is producing the analog output, that will have no affect on an analog output signal. The glitching comes from the digital to analog converter (DAC). Check out this knowledge base article:
http://digital.ni.com/public.nsf/allkb/3BB2D5D837F6C9FE86256EAC007C123B
Really, your only solution is hardware.
If you are acquiring this signal on another data acquisition system and you want to remove the glitches from what you are seeing, you could use software filtering there.
-Alan A.

Problem using PID toolkit and control design toolkit to design the closed loop system. (for DC motor closed loop)

Hi, i have facing some problem during i upgrade a project. This project is an opened loop system, it only achieved up to controlling the speed of motor, detecting the speed of motor, and switching the direction of motor while controlling the motor (This motor is a 12 VDC motor).
In order to improve the project, closed-loop dc motor control system will be implemented to correct the error of the motor's speed and maintaining the speed for the initial setting. I would like to use PID control method to do the closed loop system.
I already installed the PID toolkit V8.2 and control design toolkit V2.1.2, and i'm using DAQ usb 6221, LabVIEW 8.5
Below is my problem.
http://img177.imageshack.us/my.php?image=howtocreatethesetpointnx2.jpg
question: 1. How to generate a icon for set point? my setpoint is duty cycle...
2. How to feedback the output duty cycle to summing junction?
below this is my basic concept
http://img237.imageshack.us/my.php?image=closedloopbz5.jpg
Problem 2:
http://img357.imageshack.us/my.php?image=problem1yk2.jpg
question: Why PID toolkit icon cannot wire to cd series.vi?
Below is my original program...
http://rapidshare.com/files/140538836/pwm_generate-final_PSMII.vi.html

Hi Cyrus
Have you had the opportunity to see our
developer zone site on the PID toolkit? This article also has sample
code at the bottom that may help you in developing your application. I
have also linked below knowledge bases regarding setting a point
profile and generating a PWM from a digital output line.
PID toolkit
http://zone.ni.com/devzone/cda/tut/p/id/6440
How to generate a set point profile:
http://digital.ni.com/public.nsf/allkb/125F27AC143B6AFD86256C2B0004A4DC?OpenDocument
How to generate a PWM on a digital output line:
http://digital.ni.com/public.nsf/allkb/1561D31534F07D608625727900391114?OpenDocument
Thank You
Eric Reid
Thank You
Eric Reid
National Instruments
Motion R&D

Quadrature encoder on PCI 6220 DAQ custom digital filter problem

I am using both counters on my 6220 DAQ for quadrature encoders. What I am trying to do is filter some bounce that I am getting on the axes. I want to be able to custom set the minimum pulse width for each encoder (or both the same should be fine). The problem is that the digital filter using DAQmx channel property for each line seems to work fine (I haven't proven that this actually works) but one has to choose from preset values (the smallest being 2.55ms). I want to reject anything that is less the 10ms. This document mentions that you can set up a programmable filter based on an external filter clock. How do you actually do this? Is there no way to use one of the internal timebases for this? If so how? The "Digital Filtering Conciderations for TIO-Based Devices" entry in the NI-DAQmx help states that you can choose one of four different values for a debouncing filter on a PFI line (different values than the previously linked document) and a custom filter value.
OK here is the short question .... is there a way for me to digitally filter the lines coming into my counters to reject anything less then 10ms through software only? If so, how? If using an external timebase is required, how do I do this?
Thanks in advance for any help
Greycat

Hi Greycat,
Although the documentation does mention that you can have one programmable filter setting, this is only true for TIO-Based devices. Unfortunately, the NI-6220 is a M-Series, which means that the only allowable minimum pulse widths settings are 125 ns, 6.425 µs and 2.55 ms. More information on this can be found in your NI-DAQmx help under NI-DAQmx Device Considerations » Digital Filtering » C and M Series.
I believe for your application requirement, we would have to look into some oour Counter/Timer devices.
S_Hong
National Instruments
Applications Engineer

Cannot use DFD IIR Comb Filter Design in my application

Hi everybody,
I am filtering power line noise 60 Hz and its harmonics (120Hz, 180Hz,..) using Comb Filter Design. I do the same as the example in LabVIEW, but it does not work.
Anyone help me how to put that Comb filter into my application?
Thank you.
Attachments:
Comb Filter.vi ‏83 KB

Hello mipro,
When you say you do the same as the example do you mean the one at this link?
https://decibel.ni.com/content/docs/DOC-14292
And when you say it does not work, can you please be more descriptive. Are you not filtering at all? Can you post pictures showing your results in comparison to results you expect from using the example that you were talking about?
Regards,
-Travis E
National Instruments
Applications Engineer

Filter Design Problems - Labview 8.2

I am trying to designe a filter to replicate the frequency weighting curves in ISO-2631-4.
These curves are specified (in third octace steps) between 0.1 and 400 hz
I have based my design on the "DFD least Pth Norm Design.vi"
I have created the band specification array but in my vi I do not appear to be getting anthing out of the "DFD Get Freq responce.vi"
Any thoughts
I quite new to filter design, Am I using the right technique? If not how should I be implementing this filter?
Attached is my vi
Thanks
Attachments:
RQ Filter.vi ‏40 KB

dehmann wrote:
Wow, thank you for the diagram tbob. The program outputs correctly. I've been searching for help for almost 3 weeks now. I have a few remaining questions. 1) Why did you choose to use shift registers instead of indexing nodes? 2) What exactly is writing the data inside the loop? 3) How do I make the program write every 100 ms instead of once every cycle? 4) Do you have any other tips for programming? Thank you so much for your help.
1) You are using a For Loop which can run 0 times if the input to N is 0. With regular terminals, if this happens, the output terminals will be at default value and you will get an error on your Close function after the loop, and the file will not close. Causes memory problems. Also if there was an error before the loop, the default error (no error) will occur after the loop and you lose your error. With shift registers, the values going in (error and file reference) will be carried to the output if the loop does not execute and the file will close normally and the errors will be reported. This was a suggestions given to me by NI in one of their training classes.
2) The Format Into File function will format the data according to the input format string and then write the formated string to a file.
3) Make the cycle 100ms and then write once every cycle. This is the easiest way to do it. Make the delay 100ms to do this. You gather data every loop cycle so just write once every cycle. Why would you want to write the same data more than once?
- tbob
Inventor of the WORM Global

Fir notch filter design.(using DFD tool kit)

Hi
We used standard narrow band lowpass FIR filter design vis to desin a filter which can filter out any signal between 0.1 hz to 5hz. This does generate a filter design but is of a very high order, hence takes a very long time to process the raw data. Our raw data is of about 5000 samples and probably need to have higher samples when acqiring in the lower band frequencies,(say 0.1 hz or 0.5 hz).
Presently the need is to design a filter with folllowing characteristics:
1. Phase linearity to be maintained in the pass band.
2. sharp cut off freq band (can be the best possible to optimize speed of computation).
3. speed should be optimized.(require fast)
4. signal is mostly sign waves with noise riding on it.
5. High sampling rate is necessary to maintain the shape of the sign wave since we are more concerned with the phase of the input signal.
I have attached the snap of our design, (we used "DFD Narrowband Filter Design.vi"), which already uses the cascading technique.
thanks and regards,
sunil
Apna Technologies and Solutions
Bangalore,
Karnataka,
India
Attachments:
snap of FIR narrow LPF.JPG ‏63 KB

Hi,
Thanks for the reply.
The vi is designed but we are having an issue of speed of processing.
This filter takes a long time to process.
We are looking for the following:
Since our signal can range from 0.1Hz to 5Hz our sampling freq needs to be higher, about 8000 samples/s, this is necessary since the shape needs to be maintained for accurate phase measurements (the signal is always sinewave and with noise on it).However you may suggest this sampling rate based on your experience.
Since the phase in the pass band needs to be linear we have to select FIR and since the filter roll off needed is very sharp as provided in the attachment, the resulting design has a large number of taps(coefficients), this makes the processing slow.
--What would be the appropriate sampling frequency for such a filter design.
--Is it that such filters are slow, if so how can they be put in the front end filtering of signals acquired from a daq (PCI-6229)analog channel in a continuous acquisition mode.
we would be thankful for a quick response.
PLS refer to the attachment for filter characteristics.
thanks & regards,
sunil
Apna Technologies and Solutions
Bangalore,
Karnataka,
India

Configure digital filter for external clock

I am trying to use a digital filter on my PXI-6602 card and I can't get it to configure for an external clock. It keeps telling me the numbers don't match up but I don't see the logic to the numbers it supports. Here is the error I got for a time base of 100 Hz and a min pulse width of 1/(100 / 2) based on the period of 2 clock cycles.
Desired Minimum Pulse Width could not be produced.
Minimum Pulse Width is affected by the Digital Filter Timebase Source and the Digital Filter Timebase Rate. To see how these two property settings can affect the Minimum Pulse Width, refer to product documentation for more details.
Property: CI.CountEdges.DigFltr.TimebaseSrc
Requested Value: /PXI1Slot2/PFI36
Property: CI.CountEdges.DigFltr.TimebaseRate
Requested Value: 100.000000
Property: CI.CountEdges.DigFltr.MinPulseWidth
Requested Value: 20.0e-3
Supported Values: 80.0e-3 to 171.798692e6
Task Name: _unnamedTask<49>
I know my math is off since the filter uses the leading edge of the pulse but 0.08 seconds is 12.5 Hz and I don't get it. Other frequencies produce different but also odd (to me) numbers.
Attached is a copy of my VI
Attachments:
External Clock for Filter.jpg ‏131 KB

Digital filtering ensures that a high pulse is high for at least a certain time (minimum pulse width) in microseconds. This is to ensure that a fluke noise signal does not count as a high pulse. It also ensures that a voltage overshoot to the high value does not register more than one high pulse as it settles in to the value.
The specifications of the digital filtering is outline on page 3-1 to 3-3 of the 660x User Manual, found here. It specifies on page 3-3 that there are five different settings for the digital filter minimum pulse width:
5 µs
1 µs
500 ns
100 ns
Or, programmable with a custom tfltrclk (period in seconds of Filter Clock). However, when using tfltrclk, minimum pulse width needs to equal to 2*tfltrclk. Your current setup has the minimum pulse width set up for minimum pulse width = 1/(tfltrclck/2) = 2/tfltrclck rather that 2*tflrclk.
I think you already knew all of this. However, the Filter Clock does not equal the Filter Clock Timebase.
If we look on page 3-3 in the manual we seed that Filter clock is actually 1/4th the speed of the Filter Clock Timebase.
Therefore, in your setup:
Filter Timebase Rate = 100Hz
Filter Clock Rate= 1/4th *100 Hz = 25Hz
Filter Clock Period = 1/25Hz = 0.04 seconds
2*Filter Clock Period = Minimum Pulse Width = 2*0.04=0.08 seconds.
Which is the minimum value it was suggesting. This will always be four times as large as what you were guessing before.
Eric S.
AE Specialist | Global Support
National Instruments

I wanted to use an online website as my digital printing/design service

Please recommend me a digital printing / design web site. Or is it really worth doing this? Or I have to seek some other alternatives.
I wanted to start my own digital/printing business like signage and banners and posters. Any advic will help me out.
Thank you very much.

So, you want to be the middle-man in this venture? Customer comes to you and wants a trade show banner and a large format poster, you then give an online service all the info and they a.) design the piece for you, and b.) print the pieces for you? I'm not sure that's a very good idea. My advice would be look for a brick and mortar location where you can setup your own operation ( i.e., design and printing ). I know, this is no small task, so start out small with one workstation and one large format printer. If you, further on down the road, wanted to get adventuresome, then you could build an eCommerce site where customers can go to your site for design and printing. The problem with using online design and print services are you lose control over the process and if the customer wants to make changes and still keep within the deadline, then things get shaky. Some online services design but, then they ship the file to Taiwan for printing. Where's the proofing process in all this? Let's say everything goes along swimmingly, but then you take delivery of the goods and nothing is the way it should be? My point is you want to be in control of every step in the process. Even if you do not want to start up your own physical plant, focus on design and sub the printing to a local source where you can build a relationship and maintain consistency. You in no way want to wait and see until you get delivery that everything went as planned. I could be wrong, but what if I'm not?

Adding Proportional control using NI LabView Control Design Toolkit

I was just wondering is it possible to add a proportional control to your program using the NI Control design toolkit? There is a PID control fuction available, but i just want the P
Thanks

LC87,
A proprotional controller is equivalent to a multiply, so you could just create a transfer function with Numerator=[Kp] and Denominator=[1]. You could also use the CD Create PID Model (Parallel) with zero wired to the Integral Gain (Ki) and Derivative Gain (Kd). If you are going to implement this in a Simulation Loop, then all you need to do is use a multiply primitive.
Chris M

Digital filter design toolkit

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