specification of IIR filter

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• 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:

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• High-pass filter question

  Hi all

  I have a question about the high pass filtering in LabVIEW simply based on observation, have never been taught directly on the filters.

  A signal of high pass filtering, with a Butterworth IIR filter (2nd order) for example, around the time Fortran, = 0 there is a transition from a high-intensity "false" at a rate expected, as the officer although this is achieved thanks to a program as I have nothing else to hand at the moment.  In order to minimize the importance, for the rest of the filtered signal, you can increase the sampling time, change the filter settings or even add time of additional levy in order to truncate it.

  Is there an established method to get around this?  The reason why I ask is because I'm trying to distinguish accurately Laminar and turbulent parts of a signal of speed but on time = 0, in a completely laminar signal, our method returns false positives for the presence of turbulent structures.  I assume that this will always be the case, but it is better to ask first and concluded later!

  Thank you.

  All filters have a transient initial.  You must wait until it has settled on or try to compensate for the spike in some way.  Transitional measure also depends on the input (including any noise) signal, it is generally not possible to compensate exactly.

  Lynn

• How to initialize the State of the filter?

  Hello

  In a VI I have continually acquire and filter several different signals (with for example the Butterworth filter or the Butterworth filter point by point). The problem is that whenever I call the filter vi ti eithers begins from a State 0 or state that he had when parsing the previous signal (depending on the Boolean control).

  What I need to do is to read the final State of the filter when it finishes a piece of 'signal' A filtering. Then, the next time I need to filter a piece of the signal, I could feed these values as the initial state to the filter.

  In fact, I opened the VI of Butterworth filter, then inside, I opened the cascade IIR filter inside, and there is shift registers giving access to initial and final of the filter States. But since it's in the Labview libraries, I can't edit them.

  I have also noticed that if I in independent bodies VI of VI for each signal, then each keeps track of its own States. But I have more than 25 different signals (and the number changes all the time) so it's not a viable solution.

  Was the answer, but post here in case someone else has a similar question.

  VI's like filter Cascade IIR with I.C. (initial condition instead of the initial state, that is why I did not before), allow you to initialize the filter and save its State for later. The only thing is you need to do your own coefficients with something like Butterworth Coefficients VI or more. But it is also an advantage, since you do not need to use each iteration time recalculate the.

• Filter the data to WAV butterworth - problem

  Hello!!

  I have LabVIEW 2013

  I m new in matlab and I have the following problem:

  I want to filter voice and data to WAV file with block Butterworth Filter.vi. I don't know how to load the signal with his reading file Simple.vi, I know how to get the info on file with the Info.vi file, and I know that I can display data of waveform with waveform chart/graph. But I do not know how to properly set up butterworth filter in my diet, or better said how to properly bring data on this subject. When I m try to display data filtered with waveform graph I get only flat line.

  I already checked OR examples (IIR Filter Design, IIR filter using option Specs, IIF assigned and response, IIR filter design). But after that, I felt even more stupid and lost.

  In fact, depends on number of channels (mono/stereo) - should I separate?

  Do I have to use a loop?

  Second question - how many changes I have to do when I want to have a filtering in real time with filter.vi to butterworth (for example - to obtain data from the microphone)

  Thank you very much!

  Hi TOM 5 times.

  you set a low pass of the 1st order, so he dampens high frequencies with only 20 dB/decade. It will NOT erase anything greater than 10 Hz, it will be just wet it!

  You must use an order higher for your low-pass filter to effectively cancel higher frequencies!

• samples cdaq-9172

  The following vi allows the user to choose 1-8 global data channels and record is a record for a certain time or number of samples. A file is written to the location selected users. Any channel can be separately filtered through an IIR filter, and the user can select the rate and sample size. There is a graphical display of each channel and a digital display for each. The problem is that I get a time-out of samples. If I put the high sample rate (say 2500) I start missing samples. I use 9172 cDAQ chassis. The error is 200279, tried to read samples is no longer available. In the recording of samples or when you save a timed record, the number of samples are not correct at higher rates. Can I have a problem with my logic of vi?

  The time base of 80 MHz can be used for the counter / timer source contributions.  Generally, the 20 MHz or 100 kHz is used to generate the synchronization signals HAVE and AO.  This information and more, are in NEITHER cDAQ-9172 User Guide and Specifications.  Therefore, the question with the x-axis is probably because he is trying to use a basis of time which is too high.  In order to read faster, we can increase the number of samples per channel to read with the lu DAQmx vi, as suggested by this article in the knowledge base .  In addition, you may want to move the part of the acquisition of data from your application to a producer / architecture of consumer, who may also increase the rate of reading.  There is an example of the producer / consumer at the bottom of the knowledge base article linked above architecture.  In addition, this Tutorial DeveloperZone has a good explanation of the design model.

  David

  Technical sales engineer

  National Instruments

• DAQmx read 1 d or 2d

  Attached, I have a 4 channel daqmx read that's nchan 2d - nsample. I can't get the IIR filter works with 2d data. I can operate with 1 d, but this isn't what I wan't. The IIR filter is possible to work with 2d data? The vi works now because I just left the unwired filter.

  When I like you suggested, created the constant of waveform and connected it to the queue, everything was fine. It had been difficult for me, but you nailed. Thank you very much.

• filtering of noise over 200 kHz

  Hello!

  I'm recording ultrasonic vocalizations of rat at 1 ms/s using an acquisition of data PCI-6132.  One of my colleagues is ultrasound analysis of data from the records and asked me to apply a filter on the signal, as it is registered.  He asked the specifications was to "filter noise on 200kHs, with a steep slope.  I have no training in hearing analysis, so I have no idea what he's talking about, or how to impliment it.  Can anyone interpret this and help me understand this?  Thank you!

  Matt

  When I had to filter signals in LabVIEW, I had success with the express VI, called "filter". You can find it in the following range: "treatment-> conditioning of waveform of the Signal-> Filter.VI.

  Before we begin, I noticed that your condition is to filter "noise more of 200 kHz. This means filter any more than 200 kHz or just 'noise '?

  If you filter any more than 200 kHz, there are several parameters that you need to pay to reach your goal. Double-click the Filter.VI once you have he fell on the block diagram. The first thing you need to set is the same type. If you want to filter all above 200 kHz and spend everything below 200 kHz, you want a low-pass filter.

  Since this is a low-pass filter, you will have only a cut-off frequency. It's 200 kHz is your case, since you want to spend everything below 200 kHz.

  For a slope steep, I like to use the elliptical topology. You can start with an order of 3 to see if it produces the answer you need.

  I would recommend search filters on a site like Wikipedia, if this does not produce the results you need. Using this VI will take you where you need to go, but you may need to adjust the settings based on your signal. However, I think this will at least get you started in the right direction!

• multicast performance test?

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  We use Dell N4032 switch, the current configuration of the network's MTU 9000, with active igmp snooping and quierer address configured as the IP of the interface VLAN switch, seems she works as assumed:

  show ip igmp snooping groups

  Group VLAN journalist IIR filter address Source

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

  95 224.1.2.3 192.168.51.11 exclude Te1/0/2

  95 224.2.3.4 192.168.51.11 exclude Te1/0/2

  95 239.255.255.253 192.168.51.11 exclude Te1/0/2

  95 224.2.3.4 192.168.51.12 exclude Te1/0/4

  95 239.255.255.253 192.168.51.12 exclude Te1/0/4

  95 224.1.2.3 192.168.51.13 exclude Te1/0/6

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  95 239.255.255.253 192.168.51.13 exclude Te1/0/6

  95 224.2.3.4 192.168.51.14 exclude Te1/0/8

  95 239.255.255.253 192.168.51.14 exclude Te1/0/8

  I also tried to disable igmp monitoring on this vlan, makes no difference on the performance of multicat test.

  It's true. In fact, we put the limitation/performance target of network multicast as 1000 M/bits during the test, the desired line speed is 1000 * 1024 * 2/8 = 125 M/B. As long as bandwidth real is greater than 50 M/B, we consider the NECK.

• Restrict a smart collection

  I have a search that pulls the two members of a family. I do a search for a member of the family. I have the key words being uniform #-name. I put that I'm looking for, but am coming up with two members of the family. Adding to the challenge is we got the number 20, one I'm eager. But the other has #2 as their uniform number. Any thoughts on how to restrict this search?

  TIA.

  Hi jim,

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  Or, if you're looking for this image with the specific keyword through filter, then turn on the filter bar, by going to the view menu

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  2 choose the keyword in the first menu drop-down

  

  3. Select the desired keyword in this list of keywords and which must display the refined result.

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• . VI filtering IIR and response: response of Butterworth filter size depends on sampling rate - why?

  Hi people,

  I'm not an expert in the design of the filter, only a person in applying them, so please can someone help me with an explanation?

  I need to filter signals very infrequent using a buttherwoth filter 2. or 3. order of the bandpass 0.1 to 10 Hz.

  Very relevant amplitudes are BELOW 1 Hz, often less than 0.5 Hz, but there is as well the amplitudes beyond 5 Hz to observe.

  It's fixed and prescribed for the application.

  However, the sampling rate of the measuring system is not prescribed. It may be between say between 30 and 2000 Hz. Depends on the question of whether the same set of data is used for analysis of the higher up to 1000 Hz frequencies on the same measure or this is not done by the user and he chooses a lower sampling rate to reduce the size of files, especially when measuring for longer periods of several weeks.

  To compare the response amplitude of 2nd and 3rd order filter, I used the example of IIR filtering .vi and response:

  I was very surprised when I found that the response of greatness is considerably influenced by the SAMPLING RATE I say the signal generator in this example vi.

  Can you please tell me why - and especially why the filter of order 3 will be worse for the parts of low frequency below 1 Hz signal. Told me of people experienced with filters that the 3rd oder will less distort the amplitudes which does nothing for my the frequencies below 1 Hz.

  In the attached png you see 4 screenshots for 2 or 3 command and sampling rate of 300 or 1000 Hz to show you the answers of variable magnitude without opening labview.

  THANK YOU very much for your ANSWERS!

  Chris

  Hello Cameron and thanks for my lenses of compensation.

  I can now proudly present the solution of my problem.

  It seems to be purely a problem of the visualistion information filters through the cluster of the scale.

  After looking in the front panel of the IIR, I suddenly noticed that the "df" of the pole size is changing with the Fs of the input signal.

  For a Fs to 30 Hz, the "df" is 0.03 Hz so you see the curve of the filter with more points, see png.

  For a Fs 300 Hz "df" is 0.3 Hz, so the curve is larger with only 3 points between 0 and 1 Hz.

  For a 1 kHz Fs the df is 0,976 Hz, so there is no point in the graph between 0 and 1 Hz.

  It's strange that for constant Fs, df of this cluster NOT reduced with the increase in the number of samples, as it does in an FFT.

  However, I hope now the filter used now for the curves obtained with the proposed Lynn way and the response of greatness from the filter information fit together.

  Thank you for your support.

  Merry Christmas and a happy new year to all.

  Chris

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  Have you tried right-click (ctrl-click on Mac) and choose 'Open link in a new tab' instead? With all the dynamic html today browser back and before options are often dead still. If you open links in a new tab or window, you can simply close the window instead of using the back button. It may take awhile to get used to it, but there are some benefits, such as the loading of the pages in the background, and you do not lose the focus of the page where you. You can disable "when I open a link in a new tab, swtich it immediately." If I understand your problem, maybe it works better.

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  Hello

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