Cut noise of waveform

Hi all

I read some TDMS files I created during data acquisition. The thing is that when there is no input, I pick up a low amplitude noise (low compared to what I am probing) which of course was recorded in my files.

I want to set a threshold of amplitude of the signal cut. So when the amplitude is less than this limit, the waveform will be all zeros and when she is at or above that threshold I will deal with my original wave normally.

I tried to do it with a case structure (REAL: output is a constant of waveform elements, of zeros and the initial stage of dt-FAKE: get the original waveform without any editing) decision but accidents of VI.

Could someone suggest an effective way to achieve this? If not with the decision-making process but the noise filters? My analysis VI already performs many calculations, so I would do something very simple and intuitive without compromising the resources of the processor.

Thank you very much

Panos

So that we can reproduce and fix the crash, it'll be great if you could post an example VI as well as some example data.

With respect to the threshold, it seems that you might be looking for something like this:

Herbert

Tags: NI Software

Similar Questions

  • Add noise and waveform

    I'm doing a VI that shows how certain types of noise on a signal. The result is not what I expected, comments or recommendations to improve? below is the vi

    Use the treatment-> screws of generation of waveforms for your generation of noise from the Signal.  They use the same sampling information according to your main signal.

  • Audio device plays boring loop forever, even if disabled.

    Hello

    I had a problem on multiple computers, Windows 7 & 8, where sometimes if a program crashes or is not closed correctly, the audio device that used the program will continuously play a short segment (maybe a tenth of a second) audio in an endless loop, until the computer is restarted.
    The only other way to cut noise is to transform the master up to 0 volume slider. We have tried to disable the audio device, but oddly the sound would still play so THAT the device has been DISABLED. Hibernation and then wake the computer also has not resolved the problem, even if it turns off virtually everything. This makes it appear as a driver issue, as RAM is the only thing preserved through hibernation.
    Is there a way to reset the driver somehow to make this problem go away? Or is there another solution here?
    Thank you!
    -Steve

    I found how to solve the problem on mine:

    1. go to: Control Panel > hardware and sound > manage Audio devices
    2. Select the annoying audio device, and then click Properties.
    3. click on properties under the controller information.
    4. click on change settings.
    5. Select the driver tab and then click on disable. Press Yes when asked for confirmation.
    6. then press Activate.
  • Problem of export of mixing multitrack hearing

    Hello

    It's my first time here and I hope I can get help. I have a multitrack session, I want to export to an AIF, but when I do, I find myself with a file where the sound cuts in parts, the absolute silence with clicks in others, or just a wall of static at some point in the annual final instructions. I have what may be by bus, bus, etc. I tried export some tracks of groups, hoping to reassemble them in a new session, but even then, I has the same problem of cutting/noise/static. Two other songs have exported just fine (granted, they don't have as many stalks). What could I do to fix this? Do I need a more powerful machine? I can't afford an and I need to get these urgent exported sessions. Help would be greatly appreciated.

    Thanks in advance,

    Richie

    Sorry, I missed this thread

    I know that Durin guys have but I am also happy to watch the sesx but I doubt very much if this is the question.  "Samples ignored" and "reverberation" in the same thread could look like insufficient processing power

    I might have missed, but I don't see all the details of your hardware or the number of tracks and files at the session

    Try another PC - what type of hardware configuration that was mentioned?

    I think you mentioned that you were running on a laptop - it this laptop has a SOLID-STATE drive or is this something like a 5400 RPM drive internal or the session runs from an external drive?

    The more information the better

  • Problem when cropping of the tracks

    Whenever I try to crop a track to use only a part of it, I get something like this image.

    Immagine.jpg

    As you can see, at the beginning of the track and at the end, I get these 'click areas,' as I called them. In fact, especially when the volume of the audio element is low as in this case, it results in a sound audible ' click' that really annoy me.

    I get the result even if I cut an audio clip in Premiere Pro. On each cut me I hear this click.

    What I am doing wrong?

    geordie89 wrote:

    As you can see, at the beginning of the track and at the end, I get these 'click areas,' as I called them. In fact, especially when the volume of the audio element is low as in this case, it results in a sound audible ' click' that really annoy me. I get the result even if I cut an audio clip in Premiere Pro. On each cut me I hear this click.

    What I am doing wrong?

    If you hear click, it isn't an anomaly of the FFT of all, but the fact that you cut through the waveform in a zero point. The solution is simple gloriously; Right-click on the timeline, select waspish and then put a checkmark in snap to zero crossings. When you select a mount point, this will move your cursor to the nearest point where the wave passes through zero (usually a distance very small, almost imperceptible), and this should stop the clicks and probably also stop the FFT display too. The reason for the click you just ask your audio device to move its production of a positive or negative value to zero - instantly - and he can't do that - it's rather disregard the laws of physics!

  • AE CC: Audio waveform on the timeline is cut in half

    I was looking at tutorials CS6 and when I look at after effects CC, the waveforms are cut in half makes it harder to find the time to a song. If anyone knows a solution, please tell me!

    It's a different way, and in my opinion, once you get used to watching, a rectified waveform is a better way to use a waveform as a guide for editing.

    In the Assembly, then to the right window the model name, you will see three small lines. Click here and uncheck rectified waveforms.

  • Construction of a waveform in a bit array and and noise sometida

    Hello world

    I have a binary table 1 & 0 and I want to build a wave of her form, each pulse with an interval of 500ms, adding little noise, how can I get it?

    LabVIEW version 2010

    Thank you


  • PXI_Add GAUSSIAN additive white noise to a carrier generated using NI RFSG mode of arbitrary waveforms

    Hello!

    Does anyone know how to add AWGN carrier generated using NI RFSG?
    I know how to generate a signal CW using NI RFSG arbitrary signals mode (see attached file), but I have not found out yet how to add AWGN to it.

    Thank you!

    Aina

    Hi Aina,

    I can always be understanding not completely exactly what you ask. So in my example I gave you, we add the additive white noise GAUSSIAN on both I and Q tables separately. What you are looking for is actually add noise to the signal after he was upconvertis. Does this sound right? With our equipment, it is not possible because the baseband I / data Q are mixed with the carrier in the material as the last step before that it is generated. To add the additive white GAUSSIAN noise after the signal has been on the carrier you would need a noise generator. The best you can do is add the additive white GAUSSIAN noise to the baseband I / Q, then upconvert it based on the carrier frequency, you specify in RFSG. Let me know if this has been helpful for you.

  • How to cut a pieces to each saw tooth waveform?

    I spent a lot of time trying to tackle this problem and hope someone can offer a useful overview.  I have attached the VI described here.

    I am using an NI USB-6212 data acquisition-related Labview to create a voltage ramp, which results in a laser through its analysis.  The laser is pointed at a detector, and at the same time, I read in a signal of the detector using the same DAQ. Because the detector is illuminated by the laser, it shows the scan pad.  I would like to repeat what voltage scan and read - in indefinitely.  A large part of this system works.  I used the "RampPattern.vi" to create a sawtooth wave, which is exit with success the acquisition of data.  As expected, the entry of the sensor shows the same trend sawtooth.  However, I am having a lot of trouble dealing and displaying the digitized seesaw of the sensor, which is currently read in as a waveform.

    The challenge that I can't move is manipulate or collection of the tooth of saw so that I can do two things:

    (1) view the entrants saw tooth a 'tooth' at a time.  Each saw tooth represents a laser scan and I would like to see the ramp to scan incoming as a single slope, not a series of sawtooths.  Ideally, as each analysis came, he would replace the previous analysis (as in the "Graphic" Labview), while the plot would show a stationary scan that would flicker to the entrance of the last analysis.

    (2) divide the entrants on each table ramp sawtooth never updated that can be followed with mathematics, stored in a file, functions etc.

    I have a feeling that it wouldn't even be a back of a donkey for an experienced user, but I was hooked on it.  Could I collect the analog signal entering the detector into pieces that are the length of each ramp?  I could an event at each ramp that would split the waveform in a table?

    I am open to any suggestions!  I thank you for the input and help,

    Sam

    Problem solved!

    The solution was to wire the parameter 'number of samples per channel' in VI of reading DAQMx.  I put the 'number of samples per channel"number of points to each scan sawtooth (this is the same value as 'number of waveform samples", used to generate sawtooth in BasicFunctionGenerator.vi).

    The analog playback VI DAQMx now reads a sawtooth, both.  The waveform graph shows each saw tooth, at the same time as they appear.  Also the output waveform has the appropriate dimensions (the duration of the analysis), and I can perform mathematical operations on each single sweep.

    Works great! I posted the working code.

  • Cut the background noise but keeping the narrative

    Hi people,

    I know I can create an audio mute with Actions button but I can put up an act conditional that will cut only the background sound (music, for example) but keep the blade and the sound object play?

    See you soon

    Steve

    You can not exclusively mute background music, but you can stop the background sound on a slide with a narrative (audio slide) or audio of the object.

  • 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!

  • Mono recording has 2 waveforms, 1 very low Amplitude

    Screen Shot 2016-06-27 at 11.56.53 AM.pngScreen Shot 2016-06-27 at 12.12.48 PM.pngHello

    I am changing a mono recording. However, I have 2 waveforms. It is almost flat and short along the mark dB infinity and the other from the voice that I have saved. The registration of a person speaking was made in a room untreated. The (gun) microphone was farther from their mouths that ideal. I guess the low amplitude waveform is probably echo chamber.

    I was initially concerned about the implications on the Assembly. Now I think that given the very low amplitude of the waveform "2nd" (which will appear as a flat line unless I Zoom in so close I can see the sampling points) I can ignore and just edit to higher amplitude waveform.

    I would like your comments on these thoughts please.

    Thanks a lot for any assistance.

    Bryce

    Slacks247 wrote:

    I am changing a mono recording. However, I have 2 waveforms. It is almost flat and short along the mark dB infinity and the other from the voice that I have saved. The registration of a person speaking was made in a room untreated. The (gun) microphone was farther from their mouths that ideal. I guess the low amplitude waveform is probably echo chamber.

    Yes, as mentioned you are in multichannel mode. But, one microphone will produce only a single output, so at best this other waveform will only be noise. So, if you simply convert the channel with the real signal in a mono track (which is all you need), then the ambiguity apparent will disappear.

    So, you cut the track with noise (click the L or R button until the track you want to cut gray goes), go to "save under" and change the type of the sample in Mono in the Panel that opens, give the new file a name appropriate and save it. Gives you another advantage - don't you change the original track more, so you'll always come back if you change your mind about what it is. And we see this as a good practice anyway!

  • Buzz of noise in the tempo switch

    My song has 4 different tempos. Every time that happens, occurs a humming noise. If I get all tracks FX behind it disappears. How can I keep the delay, but not the buzz?

    Thank you y´all.

    Use automation for on/off switch of the delays to and after each change of tempo.

    If that kills too late in the face of that time, and then introduce a Gain ahead of every delay, automate the Gain to cut admission to the delay when changing the tempo. Who will cut the whole track, however.

  • How can I remove annoying clicking noise when opening new tab?

    I have goggled it and it says something of non-script (I'm not running) and nothing else that sounded remotely close to my problem. Basically, every time I open a new tab (which I do tons of) I hear the most odious type click hard and he's pushing me to the top of the walls. I simply cut the whole browser then many media players and the game as the audio directly instead of adobe flash or other browser. I just learned that he also plays more noise in the world when I use the spell check feature...

    Any help on deleting this file its terrible of Firefox would be greatly appreciated... I'd hate to start using chrome comes out of this sound.

    Only the Add-ons I use are Adblock Plus 2.6.3 Advanced SystemCare surf Protection 1.0 (I guess it installed itself with the CSA program) and Leechblock 0.6.6.

    All my plugins are disabled by default, unless I manually turn their add-on Manager window when I start firefox and the front sound pieces I never turn them on.

    Hmm...

    What exactly is the sound? To see if it is a built-in Windows sound, could you review the sounds assigned to different actions in the control panel and see if the noise you hear is one of these:

    Start menu > Control Panel > hardware and sound category > change system sounds > (in the dialog box) sounds tab

    Also, use the parameter which indicates each tab as his own overview in the Windows task bar? If so, could test showing you the real only windows on the bar?

    "3-bar" menu button (or tools) > Options > tabs

    clear the check box "Show glimpses of tab in the Windows task bar.

  • Little noises between almost every Cup

    There are separate and unwanted audio short pieces between most of my cuts on my current project, how can I get rid of them and why are they here?

    What type of audio do you mean? Is it clicks, real audio, noise, like the music, voice, etc.? Have you played the clips separately to ensure that the audio is not associated with the clip? What are the specifications of the clips? The audio is there when you add a transition? What version of iMovie and Mac OS are you running? As you can see on my questions more information is needed to find a solution.

Maybe you are looking for