Similar Questions

• 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

• CAN´t set up a low-pass filter properly

  Hello everyone,

  First of all, sorry for my bad English!

  Before asking this question, I ve tried to seek answers in the forum and couldn t find a useful for my case.

  I m new to LabView and I m test for the analysis of the signals.  I m using an Agilent signal generator and a NI USB 4431 to acquire the signal.

  OK, here´s my problem. I can´t use of a Butterworth or a Chebyshev filter (or any type) to create a low-pass filter filter. I Don t know if I didn t understand it s parameters correctly or if I m set something wrong. When I use the ExpressVI filter, I get the result I want to, but when I use the function of Butterworth, it doesn´t work.

  Can someone help me please?

  I m sending the project I ve designed, so that you guys can see what I ve done.

  Here some explanations: the "puro" labeled graph is the pure I m signal generation.

  The labeled graph "Spec. PURO"is the power spectrum of the signal current

  The graph "filtrado" is the signal after going through a low pass filter in the express VI (which works fine) and the graphic "filtrado Spec" is the power of it s spectrum.

  In the upper part of the loop is the function of Butterworth filter. I ve wired the pure data to its input signal and expect one out everything as the express VI creates, but he's not even conspire anything in the chart.

  

  The windowed FIR filter VI generates the error-20023, which constitutes a violation of Nyquist. Because this VI returns only an error code and not the cluster of standard error, you must connect explicitly an indicator or manipulation to the error output.

  The cause is that you have the frequency to zero.  OR use a somewhat confusing nomenclature for the inputs of the filter frequency screw these detailed help says:

  high cut-off frequency: fh is the high frequency in Hz. The default value is 0.45 Hz. The VI ignores this parameter when the type of filter (low pass) 0 or 1 (high-pass). When the filter type is 2 (bandpass) or 3 (Bandstop), high cut-off frequency: fh must be superior to low cut-off frequency: fl and respect theNyquist criterion.

  Thus, for the high pass filters and both low-pass cut-off frequency is the value wired to low cut-off frequency: fl. I regularly get this error. When I get strange results, I read the help and fix it. As soon as I wired 5000 to fl, the output looks like this:

  

  The differences in amplitude and transitory initial are likely due to different specifications of filter.

  The way I start it is to convert the flag to a Pure control, do default to the current value, and then put all the DAQmx screws in schema structures disable. I have disable placing the pure control (or a copy of it) in a case to permit the schema structure which has the DAQmx Read. Since you have only one data channel I added the Index table to get a unique waveform of the table. Then all the code signal analysis works.

  

  Lynn

• 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.

• 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 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

• 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

  

• Weak WIFI signal

  Hi, am currently using an HP Pavilion 2307tx g6 but get really weak wifi signal in my room. Only able to get about 2-3 bars every time while my other laptop can go up to full bars.

  Any help/solution would be greatly appreciated!

  Welcome to the HP Forum, Rascal.

  Discussion on the theme of the power of the Signal

  Buffaloes by themselves may be a little empty of meaning, UNLESS you can interpret the definition of what each 'bar' says.

  Signals are affected by the type of router you have - routers are NOT equal.

  • The cheapest router can not not the support required for the performance of your computer's wireless card.

  Example: An older router that only supports 802.11 g will not provide faster than 54Mbps - even if you have the best card you can buy in your computer.

  • The opposite can be true as well - a good router cannot provide a better service than the computer's wireless can support.

  Example: If the card in your computer is 802.11 g, so even if you have a router capable of speeds of more than 300 Mbps, you can get only 54 Mbps.

  Speed and autonomy are limited by the link in the chain.

  • 802.11n is faster than the 'g', but it can also be slowed by the walls, the furniture in metal and other obstructions.
  • Move your equipment, even a few inches, could make the difference to a router 802.11 g

  AND

  • The same might be true for a 802.11n router that has a small antenna (low power)
  • Antennas in routers are NOT equal.  Small antennas in routers fantasies can still match a whimpy router.  You want a router with a 5db antenna 7db (or better if they make these by the time you read this).

  Assuming you bought the biggest, baddest, best wireless available for your computer, then you may need to upgrade your router.

  Strength vs. Signal bars

  Bars can indicate how things are moving along, or they can cloud the issue - depends on the "bars": which generates the bars, the sensitivity are the bars and you can do a lot in a bar anyway?

  You can use another measure: the traction measured in percentages, detection and Signal type:

  WirelessNetView v1.50 - wireless network monitoring tool

  WirelessNetView works on Windows 7 and Windows 8 - there are a few problems with XP: read reviews of the author.

  ---------------------------------------------------------------------------------------------------------------------------

  You can try to update / reinstall your Wireless driver.  Help - can be a waste of time.  If you have already done, this new fact will not help - so I don't recommend it.

  Reading:

  Laptops - troubleshooting Internet or email connection on a wireless network with Windows 7

  Windows 8 stops responding or becomes very slow after you connect to the Internet

  Reference:

  Description 802.11n 802. 11-802.11 g

  Date approved	June 2010	2001	2003
  Frequency band	2.4 GHz	5.0 GHz	2.4 GHz
  Speed	100 - 200 Mbit/s	Up to 54 Mbps *.	Up to 54 Mbps
  Estimate 128Mbps	Estimate 128Mbps	27 Mbps	20 to 25 Mbps
  Range 1	range of 230 ft	range of 60 feet	100 ft range
  Modulation technique	Multiple input multiple output (MIMO multiple) and channel-bonding/40 MHz	Orthogonal Frequency Division Multiplexing (OFDM)	Spread Spectrum (DSSS/PBCC)
  Interference problems		Less interference than 802. 11 b or g	Increased interference (mobile phones, microwave ovens, ovens, etc.).
  Maturity of the product	Official specification will be finalized in June 2010		Official, final specification in June 2003
  Distance between Access Points		About every 50 ft	Depends on configuration
  Biggest advantage	More high speed and battery life	Higher speed	Top speed coupled with 802. 11B compatibility

  Help, Suggestions and examples:

  Parameters to improve wireless connectivity

  Articles, various, including:

  • The power of WiFi management
  • Laptop wireless hesitates or don't stay connected when executing a stack

  When you see a post that will help you,

  Who inspires you, gives a cool idea,

  Or you learn something new.

  Click the 'Thumbs Up' on this post.

  Fixed / responded? Click this post accept as Solution to help others find answers.

• 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

• 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

• 6733 continuous signal output generation.

  Hello

  I want to use an NI PCI-6733 map for generation of continuous signal output.

  I downloaded the modules OR measure small DDK OR 671 x & 673 x examples (PCI & PXI) and run the samples.

  Our goal is to put a few samples in the FIFO buffer and use the external update (on PFI5) to send the samples to the exit.

  Essentially a combination of AOEX4.cpp and AOEX6.cpp example should work.

  However, example 6 uses board-> DACDirectData0.writeRegister (.) to set an output value.

  To update the output, I use the following code to generate a wave of block.

  Sub
  NIPCI6733Card:nPoll_Test_T6() / / function called at 5 kHz
  {
  t67xx * Council const = & GetNI();
  Support * const theSTC = & GetSTC();

  ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
  theSTC-> Joint_Reset.writeAO_Configuration_Start (1);
  theSTC-> Joint_Reset.setAO_Configuration_Start (0);
  theSTC-> Joint_Reset.setAO_Configuration_End (1);
  theSTC-> Joint_Reset.flush ();
  ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////

  Board-> DACDirectData0.writeRegister ((OutputHigh). 0400:0 x 0 x 0);
  Board-> DACDirectData1.writeRegister ((OutputHigh). 0400:0 x 0 x 0);
  Board-> DACDirectData3.writeRegister ((OutputHigh)? 0100:0 x 0 x 0);

  OutputHigh =! OutputHigh;

  }

  My frequency of update of application (the election) is 5 kHz. The output should run on 15, which means that each update I need to resolve before the 3 samples. I want to use FiFo for this.

  (Note: the output is triggered by PFI5 at 15 kHz)

  My idea is to 3 samples in the FiFo buffer each update of the application and leave the external trigger update of the output.

  Example 4 uses the FiFo. However, this example stops after 5 running running the same fifo data.

  How can I set up FiFo continuous data?

  370735e.PDF manual describes the following:

  With non-regeneration, old data will not be repeated. New data should be
  continuously written to the buffer. If the program don't write new data to
  the buffer at a speed fast enough to deal with the generation, the buffer
  will be negative and cause an error.

  This is what I want. Is there an example?

  Kind regards

  Peter

  Dennis,

  Thanks for your reply.

  I have whil post-its on the correct Board.

  I implement for IN-Time. It is a time real OS for Windows.

  Kind regards

  Peter

• Amplitude measurement of a continuous signal in a given time window

  I'm working on an acquisition system that acquires a continuous signal of 250 kHz. My goal is to measure the amplitude peak-peak of the first reaction of signal, the problem with my setup, this is the first part of the signal is always higher than the part of the signal that I'm interested. If I try to use the measure of max from Ridge to ridge of signal VI then responds with the measure of Ridge Crest of the initial part of the signal. See the attachment for a better understanding, I would still like to view the raw signal as is, but I would like to measure the peak voltage at peak of the signal between the yellow sliders.

  Thanks in advance...

  If transient initial always occurs in the first 12 microseconds, you can use any subset of table or similar wave function to retrieve the last part of the wave. Then use the measurement from Ridge to Ridge on this subset.

  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