Similar Questions

• Loss of information on the edges of the sample of low-pass filter

  Hello

  I use a low-pass filter elliptical command to address 6 to remove trend of signal in a measure of the pressure of data sampled before, however given that the program must deal with and calculate an index in a given period of time, I can't filter and store then the data for the calculation later.

  The size of the sampe is this 1250 or 10secs data with sampling frequency value is 125 samples/s.

  In the effort to eliminate the distortion of the filtered signal, I use a technique described for tag/add a start and a value final ampitude on the beginning and end of the sample of the same length (1250), I have also run the data in the order opposite to eliminate phase effects.

  However I want to say is still a slight distortion that can be seen in the start menu of the sample; first of all, here with the loss of amplitude, but more often it's worse than that and distortion occurs at the end of time to time.

  

  

  I have some experience of the DSP (but since 20 years ago!) and remembers Windowing may be a way, but I still think that the filter should not be so difficult to implement more I don't want to lose any information amplitude.

  Any help would be appreciated.

  See you soon,.

  Kevyn

  johnsold wrote:

  ..., you may need to use different techniqoes that are not in the information stored on the previous behavior of the signal.

  Lynn

  In some cases special where information that happened before all data you have is implicit, but data yo u have, you can go out with reflecting data about initial/final data set and then run the data through the filter first forward, then backward. After that mix data accompanying your data "pretend" and just look at the part that iss associated with the actuall data set.

  Ben

• Low-pass filter before the NI 5112

  Hello

  I currently use a 5112 AND measure the signal of an infrared detector in an experience of ring to the bottom of the cavity. Below are three examples of signals. My main question is how I can implement a low pass filter, passive preference, before my 5112 OR undistorted extremely my signal due to the impedance mismatch. Now a few details:

  

  

  

  Some unique captures for each wavelength are shown in color, while average 25 pulses appears offset in black. The range and offset are chosen in each case in order to minimize the noise of "scanning". In the case of 3.14um, the noise that you see is about 25 times noise from scanning. They were taken without the limitation of BW and 100 ms/s mode.

  The detector (Vigo MIPDC-F-10) has a bandwidth of 10 MHz. I think it is a low impedance and is intended to be harnessed with 50Ohms, however its documentation confuses me, and I'm waiting for a definitive answer from the provider. 2.4 part of the manual says 50Ohms recommended, however the Datasheet and our map calibration (below) seem to suggest 1 MOhm is recommended!

  

  There are a few strange oscillations with a period of almost 180ns in our signal that I thought were due to the impedance mismatch existed in the system before I changed it:

  -Detector

  -1 metre 50 Ohm SMC Cable BNC (RG-174)

  -Inline BNC connector

  -meter 5, 75 Ohm BNC to BNC cable

  -Digitizer, DC, no BW limit, 100 ms/s, 1MOhm | 30pF

  When I saw this configuration, I knew something wasn't right and I even he modeled in LTSpice and he showed the same period of oscillations. But now, the Setup is:

  -Detector

  -1 metre 50 Ohm SMC Cable BNC (RG-174)

  -BW digitizer, DC, limit, 100 ms/s, 50 Ohm

  And we still have oscillations, even if the period seems to have changed to 320ns all about. These oscillations, which remain are 99%, probably due to our drop-down ring cavity experiment, however if anyone has recommendations on possible causes or ways I can confirm it is not because of my chain of detection they would be more than welcome!

  Now, the main question. Between the 50 Ohm 1 meter cable and the scanner I would insert a low pass filter. The BW limit has helped reduce the noise, but it can certainly still be further reduced without any lose to our signal. That's because we cut the beginning of the signal and then measure just the the decay time, which is relatively long and smooth (1 to 2 times 1/e US). Thus, in the future I may even want to try to eliminate the oscillations 320ns, but I'm afraid that this much filtering will distort the signal too. Therefore, for the immediate future I'm just looking to 'replace' the filter BW 20 MHz, with something like 1 or 5 MHz.

  Of course, I would disable the BW limit on the digitizer to avoid additional confusion, but nevertheless, I'm not sure how to approach the problem. Usually I do a lot of research and try different solutions. However, I don't have access to all components to this work, so everything should be ordered, and I don't have a lot of time to experiment. Ferrites seem like a possible solution, however not sure how effective they are at this low frequency or the way they work with coaxial cables. I know that the filter passes low RC base, but the 50Ohms (or 1MOhm | 30 pF if I change it) seem to make it impossible. I guess an op-amp based one might work, however the large input impedance is the impedance of coaxial cable... etc...

  All of the recommendations of the technique or red resources wort would be welcome. Thanks for your time.

  A possible way to separate your artifacts electric and the cavity is relatively simple.  You take the data at three wavelengths.  For each of them, make a simple exponential decay (for example exponential Fit.vi) adjustment to your data, then subtract this signal.  You should have something that oscillates on an average.  Compare the residual signals for all three wavelengths, either visually or with something like a power spectrum.  Anything in the three is probably the electronic (and you could possibly model and subtract it rather than trying to eliminate it).  This could break if the rise time of the signals are different, because that will include elements of different frequency.

  I am not convinced that you need to filter your signal before taking data.  As you said, any filter will distort your final signal.  My preference would be to take the raw signal and apply a filtering in the analysis.  LabVIEW has a rich filter, so you can experiment later.  If you apply a filter before the digitization of data, you take you will never receive data.  However, if you know that your data has no component of your proposed cut filter frequency, you should be good.  An analysis of the power on your current spectrum should tell you this.  Be careful.  Your form of rise time may have information you want later.  If you filter, you will probably slow it down.

  Good luck!  Let us know if you need more information.

• low pass filter in labview 7.1

  Hi all

  I would like to ask about the low pass filter.

  Is it possible to make a simple low-pass filter without any supplement on Labview 7.1.

  We strive to connect a micro-switch in a DAQmx device, but the thing is, because the switch is somehow Earth-connected to an engine step by step, each time the engine is running, it will have peaks and spikes were interpreted as logic 1 in the labview. Since we have no treatment signal Add ons in the labview, we try to do it ourselves.

  Thank you

  Although suggestions are significant

  But the solution has not been reached. So actually, we tried to change the analog to digital input in our DAQ hardware. I hope that the - top-of-10V-spike not to spoil our DAQ hardware. And it turns out OK. In the digital input, spikes has appeard not even once, and we think it does.

  @ t06afre: thanks for the material made up the suggestion, but since it is a testbox.foobar.com that we, his isn't going to be easy to put in engines and unlikely capacitors supposed to do. The cable twisted pair is not a bad idea though.

  We thought that the software solution filter would be the best (less time necessary and less messy) but is not as we have not thought of material assistance (R - C circuit, duuh) filter.

  And on the 'minimum pulse duration' setting, is not only applicable for some DAQ hardware? CMIIW

• I need a 50 Hz low pass filter for a 6 X 6 matrix

  I want my plate strength of the signals at 50 Hz to low-pass filter. I don't know how to apply a filter with my data types?

  Any help would be appreciated.

  See attached file

  I do not attach controls or subvi

  Thanks in advance

  Index on the channels you want filtered and run through 6 different copies of the VI filter.  You might be able to make the reentrant filters.  If so, make sure that all of the subVIs are reentrant also.  If your data acquisition is not continuous, look out for the transient filter.

  Although your speed needs are not too high, I move all the signal processing (zero, calibration, filter,...) and show the loop of consumer and have only the acquisition of raw data in the loop of the producer. If the treatment and the display can slow things down in your current configuration, you may eventually lose data.  With treatment in the consumer, the display or recording might be delayed, but you won't lose data.

  Lynn

• IMAQ Low pass filter failed with invalid image border.

  Hi Expert,

  I'm trying to use the imaqLowPass function. I expect that the input image would have been handled by the low pass filter.

  But it is a runtime error showed 'invalid image border.

  Can someone tell me what is happening?

  Problem solved.

  I add a border to the image.

  He must take care of the size of the border and the size of the filter.

• Lack of High-Low Pass filter/effect

  The high-low pass audio filter, Bass & treble and others are missing in my adobe after effects cs6. where are they? How can I get them?

  Reinstall the program. Do it with sufficient privileges to user and file permissions. Also to disable security tools. Even when running the program.

  Mylenium

• How to get data not filtered as o/p of the low-pass filter Group

  Hello

  IM new to Labview and exploring it.

  IM implementation of a system in which I need to use a LPF and a series of filter pass band.

  The filtered o/p of the LPF is as I / p for 1 GMP...

  Later, the o/p UNFILTERED 1st BPF is as I / p for 2nd GMP.

  The problem im facing is: I do not see any box in treatment that gives the data filtered or unfiltered together in the time domain of the signal.

  Could you please help me with this asap... desperately need your help.

  Thank you

  PSADAP

  There is not direct the tool to do this, but there is a solution to your needs.

  Say that if you use a LPF (with cutoff frequency, say 50 Hz), now to get blocked (UNFILTERED in your terminology) signal, set a HPF (with cutoff frequency of 50 Hz).

  See the block diagram

  

• low-pass filter modulation

  
  

  As I mentioned in other threads (and thank you for starting a new), you must post the real VI with classical control saved as default values. We cannot tell what is inside the screw Express of an image.

  Modulation is not made by adding two signals. Try the multiplication.

  It is difficult to get a clear filter. SSB transmitters often use Crystal filters.

  Lynn

• How to set the low-pass analog input (smoothing) on my DAQ configuration filter?

  I read the following of the PXI-6251 manaual:

  "On some devices, the cut-off is fixed. On other devices, this filter is programmable and can be switched to a lower frequency. For example, the hardware OR of the 628 x have a programmable filter with a cutoff frequency of 40 kHz that can be activated. »

  It does not explain how you know this value. So, I have the following questions:

  1. How will I know what the cutoff freq is on my device?

  2. How do I know if its programmable?

  3. how this program if it is programmable?

  I use the PXI-6251 in conjunction with the PXI-1031.

  Thank you very much.

  The PXI-6251 has no anti-aliasing/programmable filter. Bandwidth (-3 dB) small signal from the 6251 is 1.7 MHz.You will find these information in the specifications manual.

  If you need a filter, you must add the signal conditioning yourself.

  The 628 x M high precision tips are the only cards of the series M 'programmable' low pass filter. Check this KB for more details.

  I hope this helps.

  Luca

• Low pass FIR Filter on FPGA

  Hey guys,.

  I the DFD Toolbox and already built a few low pass FIR filter. I have a sampling rate of 50 kHz and I decimate it with a CIC filter with rate variable decimation. After the CIC filter, I need a high-grab for the low-pass filter FIR because I do want my DC signal. As I can change my rate of decimation, my FIR filter sampling rate changes also.

  The problem is that the coefficients of the filter cannot be changed during execution as the Butterworth IIR-filter function, including labview has implemented in the mathematics of the fpga function palette section.

  Are there examples how to build FIR filters where I can change the coefficients on the run?

  Think you can build a low pass FIR with the DFD toolkit and simply change the table with the coef. for a control? I could change them on the track...

  Greetz

  Slev1n

  Hi Slev1n,

  just found it in the community, maybe this might help you already:

  Polyphase Interpolation FIR Filter on FPGA with Diabaté and Coregen

  https://decibel.NI.com/content/docs/doc-16650

  Greetings

  Michael

• Filter low pass analog (anti-aliasing filter) external to the NI USB 6251 housing

  Hello everyone!

  -I m trying to acquire an analog signal of tension (high frequency content) using a connected to an edge NI USB-6251 BNC-2110. I learned in this Labview Forum that NI USB-6251 has no analog low-pass filter programmable (or anti-aliasing filter), so that I can't help but jitter when scanning my signals. For my application, the cutoff frequency of the analog low-pass filter must be equal to 100 kHz or MORE (maximum of 500 kHz). A possible solution to solve my problem, would be to work with an external analog low-pass filter before you scan the voltage signal. Based on this I'd like to know:

  (1) national Instruments develops analog external filters? I need a filter which also has one output, analog, so that I could send also the low-pass analog filter filtered signal to my NI USB-6251 box to scan correctly it!

  (2) what model of external low pass filter would be compatible with the NI USB-6251 housing?

  Any help would be much appreciated!

  Best regards!

  Hello

  all high resolution of the M series (628 x) cards are equipped with a filter low pass which can be enabled or disabled programmatically. For the anti-aliasing filter feature, examine the boards of National Instruments DSA (dynamic signals Acquisition) acoustic and vibration measurement
  currently the NOR 9221, 9225, 9227, 9229, 9233, 9234, 9235, 9236, 9239 and 9237 C Series modules feature anti-aliasing filters. These modules are intended for the high accuracy measures for which anti-aliasing filters are a necessity.

  Houssam Kassri

  OR Germany

• weak continuous signal low-pass filtering

  I get continuous signals to the NI USB-6259 of multifunction DAQ device is acquiring its signals to 1000 Hz. The rate of data acquisition can be changed through the GUI, but that

  is for later analysis. The main issue here is labwindows offers many options for filtering. I was wondering if someone could recommend the best option for a LP

  Filter on a frequency of 50 or 60 Hz. are there - it implemented an easy way to put this. Basically, I want to filter the data acquired and then store it in the file that is currently present.

  I was also wondering if material 6259 filtering, if it's a better road then should I use the filter material to clean noise signals?

  mdmorar,

  The reason why you get this error is the low-pass filter property is is not supported on the USB-6259. You will need to use a filter software for your application, because there is no low-pass filter in the material. If you look under the range of libraries in CVI and select Signal Processing > IIR digital filters > features of filtering in a single step, it will give you options for different low-pass filters. They have some low-pass filters here that should help you.

• decimation (decimation.vi there a pass filter down?)

  Hello

  I have a question about the decimation (decimation (continuous) .vi) function in labview. Has a low pass filter to take care of the aliasing, if so it's a zero-phase filter. Basically, I'm looking for something that can decimate 20 kHz signals sampled at 5 kHz.

  Thank you

  Kitenge

  Depends on your version of LabVIEW and installed toolboxes.

  The Digital Filter Design Toolkit has a decimation VI which includes an AA LP filter.  Open the help of LabVIEW and search "decimation" and "filter" to see what tools are available.

• I use a simple low-pass, but the output of the filter outputs at a different frequency, even if the input frequency is the bandwidth

  I'm simulating a sine wave at approximately 1000 Hz (I'm variable according to the frequency with a slider), I would like to pass this signal by a lowpass filter (butterworth) with a high frequency of 2200 Hz cuttoff and a low pass to 2900 Hz frequency. However, the output after the filter frequency seems to be lower in the order of a thousand. the output frequency is about 0.1 Hz.

  

  Y at - it someone who can guide me please to solve this problem, I tried different filters and I'm still having this problem, it would be incorrect sampling?

  I enclose the block diagram and the front panel

  Because you use express screws and the type of dynamic data...

  You convert the signal of DDT (which contains the clock information) in a table DBL to perform filtering. Take it a DBL array (which contains no data of timing) and converted it into a DDT (which now contains no data timing). That's why when you try to view and analyze it you have lost all the data timing (frequency).

  If you were to exit table DBL of your filter and build a wave form and provide the dt to the waveform of the sampling frequency control, then it will work.

  Better yet, ditch the DDT and use waveforms from the beginning