Xincrement is not agree with sampling frequency

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• High speed continuous measurement of encoder with sampling frequency of 1 kHz

  I am able at all times the position of a linear encoder using a PCI-6602 counter card, and I need to know how to set up so that the counter rotating at high speed, but the data is inserted into the buffer at a frequency of 1 kHz.  I am able suddenly to a hydraulic cylinder, and I am not concerned about the event recording to high frequency except to the extent where they throw off the number considerably if the equipment does not run fast enough to detect all the impulses of the encoder.

  Now, I think is that the external sample clock signal control (routed internal pulse output counter) time rate whereby the equipment detects the impulses of the encoder and the rate at which it inserts data into the buffer.  With a pulse 100 per inch encoder and a sampling frequency of 1 kHz, the extended final position of the cylinder is turned off by +/-0.15 inches, which is unacceptable.

  I need calculate a speed of this information, so I prefer not to use software timed sampling to control this (it's more difficult programming for other reasons as well - several asynchronous measures).  Any ideas on how to configure the hardware to count faster than the speed at which she inserts counties in the buffer?

  OK, you're clearly on the right track here, so I will focus on some details.

  1. How do you know that the +/-0.15 "differences are * measurement error rather than * error of movement?  Why wouldn't be an accurate measure and a proposal which can vary slightly from the nominal value?

  2. I wonder some all electric noise and defects that may produce false edges.  The fact that the behavior was better by using a sampling rate limited (200 kHz) in the digital inputs may be that some of these flaws were so short that they were never captured.

  I did a ton of work with the Commission to 6602 encoder and I can certainly confirm that count equipment is sensitive to the edges in a few tens of MHz.  (I know its 80 MHz for edge counting, but I think I remember that it can be of the order of 20 to 40 MHz to accommodate the time of signal propagation extra of the quadrature decoding circuit).

  A small point of clarification.  You're talking about the speed at which the meter "works to.  The value of count is a register whose value is changed completely by the circuit, * independent * of the sampling frequency.  If you enjoy with material-clocked County in memory buffer or interrogation of software without buffer not a bit for circuits that increments / decrements the value of the counter register.  (In other words, I am completely convinced that you would get commensurate with position end even if you took only 1 sample software-polled after the end of the move instead of sampling at 1 kHz all the way through.)

  So, if the value of the counter is disabled, it is because the circuit detects producers of County of the edges that shouldn't be there.  Something you can try is to set up digital debounce filter for input lines of the PFI corresponding to the encoder Source inputs and to the.

  -Kevin P.

• DAQmx task Read DAQmx with sampling frequency of 10 Hz produced much too much data

  I have a simple configuration with a strain of channel 4 OR-9237 amp holds a carrier of series C of WLS - 9163 (wired ethernet mode) - Details probably does not matter.

  I used MAX to create a DAQmx task associated with which all four gauges samples.  The calendar setting is "Scan Loads" is continuous sampling, 2 k buffer (read samples) and 10 Hz rate.  I guess that this task would generate 40 data values per second - 10 for each channel.

  I have a simple loop of reading using DAQmx Read.vi that works always (without any stimulation time).  Playback is set to read all available data and then pump it into a table.
  In the attached example, I also added a few words of debugging to stop the loop after N iterations.

  As the loop is programmed with a 0.2 second period, I expect each pass of the loop to read about 8 samples or 2 samples per sensor.  Instead, I get hundreds each passage.  It's like reading has substituted the sampling frequency specified in the task of the unit.  I absolutely need data to be material to the rhythm.

  Where have I lost?

  Thanks Adnan,

  I changed your example I selected 'Strain gage' entry analog and then lowered the minimum and maximum thresholds to +-1-2.  What happens is that each other in the loop, I 2048 samples or zero samples.  The display flashes a whole line and then it clears any other past.

  In response to your second post, I understand that the loop cannot run quite right that I select.  I think that, but at a sampling frequency of 10 Hz, I have to sleep on the software side for nearly a minute before I built 2 K samples.

  I played with the frequency of sampling, assigning to various values from 0.1 to 10000Hz.  The behavior is the same until I approach the high rates where available samples remains to 2048-4096 sometimes, the display becomes continuous.

  Ahhh, Darn.  Yet another search was this link that points to the root of my confusion.  The 9237 can taste arbitrary rates using its internal clock.  Duoh!  I wish that the pilots are smart enough to warn you if there is a discrepancy between the selected sampling rate and capabilities of the device

• Time of the locale with sample local, not agree with personal computer server?

  I just installed xampp and followed the instructions located at the following site:

  Set up a local test in Dreamweaver CS5 Server | Adobe Developer Connection

  Everything works as it should, except for one thing that is confusing to me.  The time on the page timetest.php indicates a time which is 3 hours before the current time.

  Is there a way to fix this? I checked all the regional settings on my computer but I don't know what I can do in Dreamweaver or in the Apache server to fix this problem.

  Y at - there someone who can help me with this?

  My operating system is Windows 7 and I use Dreamweaver CS5.

  Thanks a lot guys! I solved the problem! Woohoo!

  I went into control panel xampp and clicked on the "config" button and a menu pop up.  From there on, I chose the choice of PHP (php.ini).

  Then I scrolled down to the settings module and changed settings date.time in my area and the zone time and saved the file.

  Finally, I closed the programs and their back again and time.date parameters were I put, whether they are.

  Thank you very much for your advice and help!

• NEITHER 9212 - issues with sampling frequency, number of samples and size of buffer or slowness comes together with thermocouple module

  To begin with, I am very new to labview and unfortunately on that my first task is to build something rather complicated for a project that my company works. Fortunately it is especially followed rather than control critical processes.

  Currently, I have a cDAQ with modules, DI, AO, and TC.

  cDAQ-9133

  NEITHER 9472

  NEITHER 9421

  NEITHER 9263

  NEITHER 9205

  NEITHER 9212

  I'm currently running read-write for all 5 of these modules in parallel while loops and analog and digital inputs/outputs modules that all work as expected. Each loop requires 1 sample per channel through the wizard DAQmx etc for each module and I got can interact with all the asyncronosly inputs and outputs which at the moment is the goal. I can switch power switch for each module save memory, or if the need arrives later.

  My problem is that the module TC (NI 9212) when tilt on which allows the acquisition of sample 1 loop seems to take much, much longer to collect samples. I have played with different acquisition parameters and can seem to get a continuous flow of data when you use streaming samples, or even when specifying N-samples but I come in questions where the "application cannot catch up with the hardware.

  I tried to find an optimal number of samples/sampling, rate, etc I can get an update of temperature more than every 1 to 2 seconds, but if I set the number of samples that are high enough to apparently get a constant flow, what ends up happening waveform graphs seem to lag behind the actual data, and that's where I get errors. I only want 1 sample per channel per cycle anyway. Yet once, all the other modules in the cycle with less than 100ms delay between acquisitions but the module thermocouple ends up being 1-2 seconds. It's okay because in our application temperature should not rise/drop very quickly but its boring nonetheless.

  I have the acquisition set up for the same (NI 9212) thermocouple module was like my modules of analog and digital inputs (NI 9205 and NI 9421) take 1 sample every time the task is called so I'm having a hard time understand why do the same for the TC module introduced such a delay. Max sampling rate is supposed to be 95 samples/s/ch, so I guess that the delay must come from call the task over and over again in a loop.

  Synchronization and how you have configured the 9212?

  In the case of high resolution, it can reach 1.8 samples/s. see page 7 of the plug technique here for rates for the supported modes:

  http://www.NI.com/PDF/manuals/374389a.PDF

  -AK2DM

• Due to my inability to find Chrome continued to stop and the reason why they are changing some things, I do not agree with. With Firefox, am I looking through google? If so, there is no reason for me to download the program (s) of Firefox.

  If your service uses Google, I have your service. How a search without using Google?

  You can use any search engine of your choice.

  • See-> https://support.mozilla.org/en-US/kb/Search%20bar
  • See-> http://kb.mozillazine.org/Search_Bar
  • The search engines that you can add-> http://mycroft.mozdev.org/

  The recent change in Google's policy only affect users if they sign in with a Google account or use a Google product that requires a sign-in (i.e. Gmail)

  • http://www.LATimes.com/business/technology/la-fi-TN-Google-Privacy-Policy-March-1-20120229, 0, 2129072.story
  • and Google is not not alone I look at-> http://www.theatlantic.com/technology/archive/12/02/im-being-followed-how-google-and-104-other-companies-are-tracking-me-on-the-web/253758/

  If this answer solved your problem, please click 'Solved It' next to this response when connected to the forum.

• Windows 7 screen resolution is not agree with me after I did a fresh install of Windows!

  So after doing a fresh install of Windows 7 Home Premium on my laptop two days ago, I have tried many unsuccessfully to try to resolve a problem of resolution resulting from the update. My resolution is automatically set to a resolution of 1024 x 768 very blurry and I cannot recover my original resolution. When I go to the next option and the last, on the resolution list, 1280 x 1024, the screen is completely distorted and stretched which is hard on the eyes and is certainly not the original resolution for my laptop before the update. It's like installing new Windows completely wiped out my original graphics drivers.

  I also noticed with update that my graphics driver and the driver were set at "NVIDIA Standard VGA Graphics Adapter" which I am absolutely positive is not what I was using before, that I don't have a video card installed, even if I use not this laptop for games and is no longer for the school and other work. I'm 99% sure that I have a card NVIDIA GeForce FX 5200 installed, that even though he is old, always worked well, but even in this case, I'm still not sure if it's the card I have, which could be the reason why I can't find the right drivers for it.

  For reference (I'm running in Mode safe):

  

  I tried to update this standard graphics driver, but he claims, there is no update. I run all my virus programs, I have all my updates installed Windows, and I used the boost driver to try to see if there is any other updates, but nothing has worked. I did a "sfc/scannow". I ran Microsoft FixIt, tried to look in the "list all Modes" under the control panel screen resolution and I currently can't 'uncheck' Modes 'hide' because it is all grayed out. Honestly, I think I have tried everything to solve this problem, but I think he came to where I think I need to ask someone who can find the exact video card I have in my computer so I can possible find the driver for it, and maybe that would solve the problem.

  For reference, I also have a laptop Toshiba Satellite P25-S507. Running Windows 7 Home Premium 32-bit. If any other type of information is necessary, I can give him, let me know how to get around to provide for you. I will continue to try to see what I can do but for the moment, help! D:

  You do not use / install "driver booster" or other programs of type 'FixMyPC '.

  You need to visit your computer makers portable, specific to your model support site and install all the available drivers (Chipset/graphics/lan, etc.)

• AS the class is not agree with MovieLoader

  I've never posted here before, so if I missed something essential in what most people post then a request for assistance, do not hesitate to mention.
  
  Basically I have a movie (the "mini game"), which uses two different classes, "Eye.as" and "Target.as". Then I have another movie ("hub"), that loads the first, that of the mini-game, via a MovieClipLoader. When I run the mini-game in itself, everything is fine. However, when I scoured the mini-game the hub, just target class may as well not exist. Constructor, methods, everything disappeared. But the class Eye works perfectly well and she should.
  
  Both classes extend MovieClip and are on the scene in other clips without class, not through placement of code, but the design in the editor WYSIWYG (Flash 8 Pro). The mini-game with a beginning frame nothing but "stop();" and on the hub event MovieClipLoader onLoadComplete, the mini-game is sent to its initialization frame, which then sends it to the part 'instructions' (just a bunch of text and graphics for the player). On which, the player can click on a button and be moved to part real ' game' of the mini-game, which is where the bodies of the eye and the target exist.
  
  Anyone encountered this before? I tried to have the target on the scene at the beginning, like the eye, endearing, or duplicate a targets already on the scene. Nothing seems to have no effect on the problem, and the problem only when executed by the hub.
  
  Any help, comments, critical, flames or anything else is welcome. I hit a brick wall completely stumped on this one. Thanks in advance for anything that anyone can offer.

  -kglad:
  Target initially used _root for some items, and I was using _lockroot (since a number of games potential that it could load, not everything I write, can use _root). However, I deleted _root and found other ways to get what I was trying to understand this.

  -DMennenoh
  Just pass it to onLoadInit (which _worlds_ more logic, thank you), and unfortunately it does not solve the problem.

  However, I found that I be blind to what it was to me. The film of "hub" were two class paths together (prepare forehead smack now), and in one of these folders, another class named "Target.as" has found its place. So while I thought "Target.as" was used by the fla "mini game" to its swf, "finalize" the target in this swf class, load the SWF mini-game in the hub has replaced the target class in that the hub already implemented...

  Thus, after having solved this problem of naming ignorant enough, the problem. Sorry for the trouble with something that I should have seen in the first place. -_-

• How to acquire with NiScope at different sampling frequencies and lengths Records?

  I need to acquire the data of 2 channels of the NI PXI-5114 map two different sampling frequencies high, at the same time. Also, I put 2 different record length. Is this possible?

  I understand that 'Vertical' settings can be configured for individual chains because the function 'Vertical niScope Configure' has 'channels of entry with which we can assign the desired channel. But for horizontal settings such as "min sampling rate" and the record min length, I could not find such an option to specify the channel. Would it not common to both channels?

  I hope that the device is capable of simultaneous sampling and therefore channels can be configured individually to different sampling rate.

  Hi AJ_CS,

  Why do you have to be distinct from sampling frequencies on channels separated from the digitizer even?  What different sampling rate do you want?

  But for horizontal settings such as "min sampling rate" and the record min length, I could not find such an option to specify the channel. Would it not common to both channels?

  You do not have an option to configure the settings of hoirizontal on a channel by channel basis because this concept does not exist in the traditional sense of the use of a scope.  Compatible with the concept of IVI, an oscilloscope traditional benchtop will have only a button or a set of buttons for setting the parameters of synchronization of the unit.  There is therefore no horizontal configuration to separate channels on the scanners NOR.

  I hope that the device is capable of simultaneous sampling and therefore channels can be configured individually to different sampling rate.

  Similar to a traditional benchtop oscilloscpe, the device is capable of simultaneous sampling.  But as mentioned above, the channels can not be configured for different sampling frequencies high.

  However, you can ignore data that you think is not relevant.  For example, if you assign 100MS/s CH0 and CH1 to 50 MS/s, then you throw all other samples.

  Alternatively, you can use separate scanners (a channel on each digitizer) and configure them to taste at different rates.  You can set frequencies of sampling on scanners NOR separated and even synchronize them with TClk.

  -Andrew

• Synchronization of the inputs and outputs with different sampling frequencies

  I'm relatively new to LabView. I have a NOR-myDAQ, and I am trying to accomplish the following:

  Square wave output 10 kHz, duty cycle 50%.

  Input sampling frequency of 200 kHz, synchronized with the output that I get 20 analog input samples by square wave, and I know what samples align with the high and low output of my square wave.

  So far, I used a counter to create the square wave of 10 kHz, display on a digital output line. I tried to pull the document according to (http://www.ni.com/white-paper/4322/en), but I'm not sure how sample at a different rate than my clock pulse. It seems that this example is intended rather to taste one entry by analog clock pulse. There may be a way to create a faster clock (200 kHz) in the software and use that to synchronize the analog input collection as well as a slower 10 kHz output generation square wave?

  I eventually have to use the analog inputs to obtain data and an analog output to write the data channel, so I need the impetus of the square wave at the exit on a digital PIN.

  How could anyone do this in LabView?

  Hi Eric,.

  All subsystems (, AO, CTR) derive from the STC3 clocks so they don't drift, but in order to align your sample clock HAVE with pulse train that you generate on the counter, you'll want to trigger a task out of the other. I would like to start by a few examples taken from the example Finder > Input and Output material > DAQmx. You can trigger GOT off the train of impulses, start by Gen digital Pulse Train-keep -you probably already use a VI like this to generate 10 k pulse train. AI, start with an example like Acq Cont & chart voltage-Ext Clk - Dig Start.vi-you'll want to use the internal clock so just remove the control of the "Source of the clock" and it uses the internal clock. From there, simply set the "Source of the command" either be the PFI line generates the meter, or ' //Ctr0InternalOutput '-assuming that you are using the counter 0. You'll want to make sure that the start of the task HAVE faced the task of counter I is ready to trigger off the first impulse. They should be aligned at this point.

  For debugging, you can use DAQmx export Signal to export the sample clock - you can then brought the train line and the PFI pulse to make sure that they are aligned.

  Hope this helps,

  Andrew S

• What is the relationship with the sampling frequency and number of samples per channel?

  In my world, if I wanted to taste 10 seconds 10 Hz (100 s/ch), specify a rate of 10 and a number of samples of 100.  This would take 10 seconds to return data.  The task does not appear to behave this way.  No matter what rate and the number of samples, I chose, I spammed with data at 1 Hz or more.

  What I am doing wrong?

  This problem is resolved by making a request for telephone assistance.  It turns out that the minimum sampling frequency of the NI 9239 is 14xx s/s.  I don't know why there is a minimum sampling frequency, but now I have to go to the next question discussed at this link:

  http://forums.NI.com/T5/Multifunction-DAQ/how-can-I-read-gt-10x-samples-at-a-sample-rate-of-X/TD-p/2...

• DT of a waveform is not equal to the sampling frequency

  I wrote the attached VI to take data from two different channels, 1 a cell, and the other being an encoder. I have my calendar synchronized with the entry load cell. The question I have is that at different sampling rates high for the program, does not change my dt in my waveform data.

  If I understand correctly the dt of the waveform should coincide with the sampling frequency. Am I missing something?

  Thanks in advance for your help

  
  

• Sampling frequency do not match

  I'm incredibly confused and a little frustrated, I hope someone can help me.

  • 10.9.5 OSX
  • Hearing 2014.2 CC
  • Blue Yeti Pro USB Mic
  • Apple Midi Setup: output: Hz 192000,0, entry: Hz 192000,0
  • Its Apple: Blue Audio USB 2.0
  • Audio hardware preferences hearing
  • New Audio file to 192000 Hz
  • Sample rate error

  Somehow, I got it work before. 192000 Hz at the hearing. No amount of fiddling with settings, reboot, or others can get me higher than 96000. 96000 works that if I go down the Apple Midi configuration to 96000, which makes sense, as suggests the error message it must be the same as Apple Midi. How is it, I can't Audition to save to 192000 when I put in configuration 192000 noon Apple?

  If I uncheck "attempt to force material to document sampling frequency" "Sampling rate" menu does not go higher than 96000, same Apple Midi Setup is set to 1920000. For fun here's a screenshot of the Audio channel mapping.

  No problem.  Let's take a step back and go through some basics.

  digital recording of sampling rate determines the highest frequency that can be reproduced.  A 44 100 sampling rate corresponds to a higher frequency of 22 050 Hz which is comfortably above the highest frequency even the best human ear can capture.  (Theorem of Nyquist Google if you are interested to know why.  This frequency is also higher than what most of the microphones can pick up and most of the speakers can reproduce.

  There are some who claim they can hear the difference in the higher sampling rates but certainly, I don't pretend to be one of them - and I have yet to see a blind man good test that could confirm.  In addition, some say that some special effects would be preferable to a larger "resolution."  Again, I'm skeptical, but maybe there could be some 3rd party plug-ins designed to work that way.

  That said, if it's for the video, you can work at 48, 000Hz, not for the quality but because it's a video standard.  FYI, it's to do with pair it with some video frequencies,

  Then, little depth.  This determines the available dynamic range.  24-bit has become somewhat of a standard record in common in the distribution of 16 bits.  For simple voice, you don't need 24 bits of dynamic range, but if you can't set to 16 bits is not a problem.  Hearing (and many other editors to work internally to the point 32, but floating because it protects you from distortion if you leave the creep too high levels when editing.  However, hearing handles it for you, and you only need to set your inputs and outputs according to the needs. In your case, that this is starting to sound like 48 000 / 24, but as you will be video and.what your default mic.

• Sampling frequency of input/output does not match.

  Hi all

  I recently moved to audacity to audition and I have a small question: I have a Zoom h2n who worked well with Audacity, but when I tried my first recording with audition, I was told that my input and output sample rates do not match. I've already looked around and discovered that I need to change the format of my speakers to match the entrance to 44.1 I my h2n zoom fixed to however I was wondering if there was when even just ignore it. What the output is important in any case, my fault for my speakers is quality studio 24 bit 48000. Did I miss an option in the settings of the hearing? What I really need to drop out of the enclosure to 16 bit, 44100?

  Thanks in advance for any help.

  Peace.

  If you use the Zoom as an interface to audition then you really must match your session, the sampling frequency of the aufdition otherwise interface is going to be on-the-fly resampling

  So when you create a new session you assign at the same rate as the Zoom or vice versa

  You can also try to edit > Preferences > Audio hardware and check or uncheck the "attempt to force material to document sampling frequency.

• Hello, I agree with Adobe. I reconfigured my Apple and reinstalled Adobe. But I can't not find my applications (photoshop/lightroom). I wonder for my series of numbers, I can't find. Please help me...

  Hello, I agree with Adobe. I reconfigured my Apple and reinstalled Adobe. But I can't not find my applications (photoshop/lightroom). I wonder for my series of numbers, I can't find. Please help me...

  If you have CC desktop application installed and then try to connect and insert it again.

  Otherwise, install as you did initially.  Download/install the desktop application, connect and install applications subscription.

  Creative cloud to desktop
  https://helpx.Adobe.com/creative-cloud/help/creative-cloud-desktop.html

  Sign out, sign in | Creative office cloud app
  http://helpx.Adobe.com/creative-cloud/KB/sign-in-out-creative-cloud-desktop-app.html

  Install, update, or uninstall applications
  http://helpx.Adobe.com/creative-cloud/help/install-apps.html