One-Third Octave Filtering of Low Frequency Data

I would like to analyze steady state, low frequency (0.1Hz to 80Hz) data that has been aquire over several minutes of acquisition at a rate not exceeding 1000Hz. I am using the Third Octave Filtering Tools in the Signal Processing Toolkit, but it is unclear to me that if I just change the sample rate input into the Third Octave Filtering.vi if that will give me the octave bands that correspond to the ranges that I am interested in. Is the any other way to calculate this? For example, if I wanted to find the band power of an individual octave band, say band 0, which has a center frequency of 1Hz, how would I do that?

There is a thorough explanation of the algorithms and details of the ANSI S1.11-1986 specification to which the toolkit is built around in the reference manual for the Signal Processing Toolkit. Here is a link to it:
http://www.ni.com/pdf/manuals/322142a.pdf
On page 24-4, it explains how to calculate the center frequencies of your 1/3 Octave filter bands.
Jack Arnold
Application Engineer
National Instruments

Similar Messages

One third octave in frequency domain

Hello,
I want to know dynamic stiffness of a machine. To do this I am using FRF function of Labview. But customer wants this FRF output (dynamic stiffness) to be plotted in 1/3 octave plot as end result. As FRF output is in form of magnitude Vs Frequency, I am not able to connect this output to one third octave. Is there any method where one can get 1/3rd octave in frequency domain? Thanks in advance.

There is a thorough explanation of the algorithms and details of the ANSI S1.11-1986 specification to which the toolkit is built around in the reference manual for the Signal Processing Toolkit. Here is a link to it:
http://www.ni.com/pdf/manuals/322142a.pdf
On page 24-4, it explains how to calculate the center frequencies of your 1/3 Octave filter bands.
Jack Arnold
Application Engineer
National Instruments

The specs for the NI model 4552 DSA card realtime octave addon are listed as max 10KHz band for one-third octave on 4 chan

nels (20KHz for 2 channels). Does this limitation apply only for real-time operation? In other words, if I was satisfied with non-realtime analysis of a steady-state input signal, could I get higher bandwidth, say, up to the 50 KHz one-third octave band?

nels (20KHz for 2 channels). Does this limitation apply only for real-time operation? In other words, if I was satisfied with non-realtime analysis of a steady-state input signal, could I get higher bandwidth, say, up to the 50 KHz one-third octave band?DonH,
This is a good question. The analog sampling is real-time and hardware controlled regardless of whether or not you are doing hardware analyzing. You are not at risk for missing any data points or affecting the sampling speed. If your computer is not fast enough to analyze the data as it comes in, you could stream the data to disk for analyzation once the acquisition is through. You will know the computer is not fast enough if you got a buffer error during the acquisition. Basically, you are not compromising the quality of data at all. It will just take longer to get results.

Best method for collecting low frequency data

Hello everyone,
I'm looking for suggestions on the best way to collect relatively low frequency data (about 1 Hz). I know there are a few different ways to do so in labview such as the DAQ assistant or making DAQ mx and making your own virtual channel. Also there are an abundence of different settings to choose from. I'm using an NI 9215 DAQ card and am collecting voltages. I would be interested to here any opinions on a method for doing so and maybe the settings that they would use.
The reason I'm asking is because I'm just using the DAQ assistant but I'm really not sure if that's what I want to be using. I feel like there is a better way.
Thank you all!

winterfresh11 wrote:
Is this different from triggering? Because this particular DAQ card can't be triggered.
There is a big difference between triggering and sample clock. The trigger tells the DAQ to start acquiring data. The sample clock tells the DAQ when to take a sample. You trigger once per acquisition. The sample clock just keeps on going until the acquisition is complete (either aborted or desired number of samples is acquired).
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
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One third octave alaysis

Hello everybody.
First, I'm sorry for my bad english. Hope everybody understands!
I'm studying vibration analysis, and I need to use an already measured data (acceleration in g/ time in seconds) to compare with the vibration criteria curves (picture below).
Note that the VC curves axis are Vrms (um/s) X 1/3 octave band Frequency (Hz). Well, I am able to do some integration of my raw data and get the values in velocity X time.
If I use a FFT, I can go to frequency domain. But I don´t know how to express my data using this 1/3 octave band.
I would appreciate a lot if someone could help me!
Thanks in advance!
Henrique

Hello Henrique,
To have a function to do 1/3 Octave analisys direct you need have installed LabVIEW Sound and Vibration Toolkit or Sound and Vibration Measurement Suite.
http://zone.ni.com/reference/en-XX/help/372416H-01/TOC27.htm
http://zone.ni.com/reference/en-XX/help/372416F-01/TOC20.htm
http://sine.ni.com/nips/cds/view/p/lang/pt/nid/209056
http://sine.ni.com/nips/cds/view/p/lang/pt/nid/209061
The following link have an example using Sound and Vibration Toolkit functions, but only open if you have this toolkit installed.
https://decibel.ni.com/content/docs/DOC-2085
Best regards,
Abel Souza
Engenheiro de Aplicações
National Instruments Brasil

Third octave analysis of a blast signal

Hi,
I am trying to analyze one blast signal in third octave band. I am taking a narrow band power spectra of the signal and it is giving me the correct values, but when I am trying to do third octave band power spectra it is not giving me the correct results. I tried using the SVT octave analysis from sound and vibrations toolkit but still it is not giving me right results.
The sampling frequency is 24000Hz.
I am attaching the Vi and the file which I am trying to analyze. I am trying to solve this problem from 4-5 hours and still not able to solve this. Any kind of help will be highly appreciated.
Thanks,
Nitin
Attachments:
Power spectra and third octave.zip ‏1225 KB

Hi again,
I figured that problem out. but still I am not getting the values from 10 Hz, even when I am setting the low band frequency to 10 Hz. It is giving me the value from 20 Hz. I am attaching the new VI and the file.
Thanks,
Nitin
Attachments:
Power spectra and third octave.zip ‏3388 KB

Third octave decomposition

Hi!
I’m working about third octave decomposition with a vibration signal.
I’m using the sound and vibration toolkit.
The sample frequency of my signal is 2400 Hz.
I use the vi “1/n octave IEC decomposition” and I would like to know if I can selected the frequency range “1 to 400 Hz” in third octave decomposition? I think the results are strange in a low frequency (1 to 4 Hz)
In fact, for example, the BW of the first 1/3 octave is less of 0.3 Hz!!!!
So it’s possible to begin the third octave decomposition at 1 Hz?
Thanks for your help.
Bastien

thank...
But I would like to analyse a short signal (5 to 10 second) in the frequency range [1Hz to 400Hz].
When you begin the third octave analyse at 1 Hz you must to have a vibration signal with 17.24 s minimal length in time.
this minimum time is say in the user manual of the toolkit "sound and vibration":
Filter Settling Time :
When starting or resetting the filtering operation of the fractional-octave
filters, a certain time is required before the measurements are valid.
This time is called the settling time and is related to the bandwidth of any
particular filter. The lowest frequency band has the smallest bandwidth and
defines the settling time required before you can consider the complete
fractional-octave measurement valid. In the Sound and Vibration Toolkit,.......
settling time is defined as five divided by the bandwidth of the filter used
for the lowest frequency band.
..........The filter settled indicator returns TRUE as soon as all the filters are settled.
When you begin analyse at 1 Hz, the first BW is 0.29. So, 5/0.29 = 17.24 second.
My vibrations signals are too short (5 to 10 second) and I can't have more long because this signals are due to train course and it’s very fast.
I don’t know how to catch information in a low frequency (1 Hz) with a short signal.
Thank for your help
bastien

Why am I getting very high values for the very low frequency region of a random signal?

I am tyring to produce a power spectrum graph for a Tachogram data, related to Heart Rate Variability Analysis. This data can be thought to be as a random signal, but has a frequency spectrum range of 0 - 1HZ.
The problem that I am facing is, I am getting very high values for very low frequency region, closer to DC value. Even the DC value is really high, in the range 10^8. It is suppose to be a low number. Any suggestions would be appreciated.
Thanks.

Here is what my work is all about. I am trying to develop a software for Heart Rate Variability analysis. I am not sure if you are aware of heart beat waveforms, they are bunch spikes, occuring at irregular intervals. We have to do analysis on this waveform. How ? First we have to create a plot called Tachogram. This is done by, for example, let's name the first spike R1, the second one R2, and the third one R3, and so on. This is how the co-ordinate points are created. (R1, R2-R1), (R2, R3-R2), and so on...
Here R1 is the time instant at which the spike R1 occurs, and R2-R1 is the time difference between these peak occurences. So, if we plot these values, then it gives us one like what you saw on your time domain plot. What we have to do with this signal is, we have to apply FFT technique to produce its power spectrum for analysis. Generally the frequency range of the spectrum goes from 0 - 0.5 Hz. They part this range into VLF, LF, and HF, and analyze how much power is distributed in these regions. The distribution would tell you if you were stressed out,or have worked out, things like that. Normally, you can view spikes around LF, HF region. If you had viewed the power spectrum for the data that I sent, then that's what it pretty much looks like. I think I have said a lot about it now. Hope you get it.
We were talking about the "other component", I can't really see where it lies, must be in the 60-80 sec. range in the graph. If you look at the spectrum, I am getting a huge value around 0.0033 Hz, how do I minimize or remove this ?
Also, I am using Unevenly Spaced Signal Spectrum.vi, I am not sure what excactly is the unit for the Power Spectrum output. Technical Papers denote, Periodogram algorithm used by this VI, gives Power Spectral Density ? If that's right, then the unit is V^2/HZ.
Please advise.
Again, I should thank you for your time and effort.
I have to meet a deadline this week in finishing this project, and your help is immense to me at this point.
Thanks & Best Regards,

How to make third-octave analysis program?

"I want to make a program which can do Third-Octave Analysis. But I haven't any DAQ device of NI company. I use a DAQ device of Nicolet company. That device can tranfer the acquired data as a file to a computer, and then I can use a software to transform the data into ASCII format. The DAQ device hasn't an anti-alias filter. How can I make the program with LabView? My version of LabView is 5.1."

The task that you are inquiring about could become quite complicated. for
example, if you are performing acoustical analysis then frequency and power
become your primary points of interest. However if you are performing
analysis of shock and vibration then additional processing will be required.
I would recommend that we take this off of the news group and work direct.
I can be reached @ [email protected]
remove no_spam_ for real address
Mike
"stewart342" wrote in message
news:[email protected]..
> The DAQ device of nicolet is called 'vision'. What I mean is how can I
> make a VI to get the analysis result from the data acquired. Because I
> am not good at LabView and signal
processing thoery, I hope you can
> give me a brief example. Though the device provides a filter, the
> cutoff frequency is not arbitrary. So another question is: If I don't
> use an analog anti-alias filter, can I use a digital filter instead?
> The raw data I've got is in ASCII format. The first column is the
> time, and the following columns are the signals acquired from each
> channel. Supposed the data haven't been filtered by the anti-alias
> filter.

Third octave operations with "sound and vibration toolkit"

Hi!
I’m working about third octave decomposition with a vibration signal.
I’m using the sound and vibration toolkit.
The sample frequency of my signal is 2400 Hz.
And I’m working with a 1 second vibration signal (so 2400 points).
Fs = 2400
N = 2400
I need to understand the different and successive operations that the “SVT Fractional-octave Analysis [IEC].vi”VI do.
In fact, I think that my frequency resolution (df = Fe/N=1Hz) is too low.
Do you have a solution?
Maybe I could turn into my 1 sec signal in 10 sec signal with 10 recopies and assemblage of my previous signal?
N’ = 10N so df = Fe/10N = 0.1Hz.
Thanks for your help.
Bastien of paris

Hello,
The first thing you have to do is to read the Chapter 7 of the user manual of the Sound and Vibration Toolset.
You can find it in : program -> National Instruments -> LabVIEW Sound and Vibration Toolset -> User Manual
This chapter discusses fractional-octave analysis, including fractional-octave analysis theory, averaging modes supported by the
Octave Analysis VIs, settling time, and ANSI and IEC standards.
regards,
Marc L.
NIF

Third octave analysis giving very close but in accurate results

I am having troubles with 1/3 octave analysis. I have recorded a calibration tone of 94 dB in B&K Pulse and exported a time series that is Pa values, an AC signal of 1.414 peaks over a time period of 15s.
When loading this CSV file and passing through the third octave module I get 94.6 dB at 1kHz and overall sound pressure of 94.6 dB. However the Labview output, especially in the lower bands, of the third octave anaysis are up to 10 dB out from the pulse measurement.
The settings, averaging type (linear), weighting (linear) are the same and yet the values of the thrid octave output are very slightly different. I can provide code and the Pulse output - i wonder if anyone else has has similar problems?
Any help is appreciated.
Thanks Greg

Hello Greg,
I cannot say that I have seen a similar problem. Which version of LabVIEW are you using? Which Sound and Vibration option have you installed?
There are several examples that ship with LabVIEW for Octave analysis. Did you start with one of the examples from Example Finder to build your application, or have you created one entirely on your own?
Providing the code would be helpful. Please post the smallest piece of code that can be used to replicate the issue.
Regards,
George T.
Applications Engineering Specialist
National Instruments UK and Ireland

"How to measure very low frequency?"

I'm using Labview 5.1 and a data acquisition card PC-LPM-16PNP. How can I measure a low frequency signal like the human pulse of 72 pulses/min or 1.2 Hz. Thanks.

Virgo,
When measuring very low frequencies, I recommend measuring the period between pulses and converting to frequency.
To measure the period, there are a few methods. For something like a heartbeat, I would probably use a peak detector to locate the peaks and measure the distance between the peaks. For other signals, it may be easier to locate zero crossings. Sometimes filtering or smoothing is required to remove noise and improve the results.
Bruce
Bruce Ammons
Ammons Engineering

How do I get the line-in to record low frequencies?

The iPhone with GarageBand is potentially one of the most powerful tools for mobile recording ever. Unfortunately there are several tragic yet avoidable drawbacks that reduce it to nothing more than a mere toy. 1. You can't use the camera usb adapter to bring in loss-less digital stereo input. 2. Even if you could, you can't split the stereo into 2 seperate mono tracks. And to take the cake... 3. You can't do much recording with the 3.5 jack's line-in because you get ABSOLUTELY NO LOWS! This is due to the iPhones roll-off low frequency limiter filter, which i'm told since ios6 can possibly be disabled, though it's not in the Settings and i have yet to find an app that will do it in the background while running Garageband. Sure, the Garageband app could do that, but then it would be a powerful incalculably valuable tool in the hands of everyone. No lows means no drums, no bass, and vocals that sound like...well...like they were recorded via telephone. Yes, it's painfully clear you recorded that demo on a phone, because when you play it back in your home or car it sounds like your listening to someone play it for you on the other end of a phone call. There is no adjustment to compensate for the absence of low frequencies. SOMEONE PLEASE TELL ME I AM WRONG and that you have a work-around or know of something I've missed. How do you get lows into Garageband???

Thanks for the info. I am not wanting flat stereo nor digital input from the 3.5. Only wanting flat mono. But before I go the apogee route (which is a $150-200 solution, not to mention a solution that will add more devices to the connection and thus more opportunities for noise), I will probably just resort to purchasing an ipad (can get a mini for $299) . My mixer has usb out, and I can go straight into the ipad digitally with the usb camera kit, albeit I still can't split 1 stereo channel into 2 mono channels. But I digress: my entire purpose for this post is that i would like to stay as mobile as possible by using the iphone, but with all the darn limitations imposed by no one other than Apple, what could be an easy way to record has been nullified. What's frustrating is that all the components are there and I have them, all the hardware is more than capable; but in both cases (the 3.5 mono or the usb stereo) there is a software feature (or lack thereof) blocking me.
Again if anyone knows of an app that will disable the software low-cut, please respond...

27" iMac screen low frequency humming + moving too freely on the base

Got a new 27 incher on tuesday after fedex lost the first one That's a different story.
The problems
1. The huge iMac moves a tad bit too freely on its base and when i try to tilt it to an angle, it doesn't stay at that angle it trys to go tilt back down ... maybe its gravity.. i don't know...
2. Very very low frequency humming noise from the iMac top left corner... its out when it goes to sleep. not the normal HDD/fan noise. I have a fan app, its not the fan... but its really annoying when working on it ...
i've seen a couple of posts about this. i'm taking this 27 incher to the store tomorrow... lets see how the geniuses deal with it. i hope they acknowledge this problem. does anyone else with the 27" have this problem?

Took it to the apple store, some genuises looked at it, confirmed it and sent me another one

Sound Distortion problems w/ lower frequency tones/instuments

I've been ending up with a lot of distortion in tracks when I bounce them over to Itunes. Primarily w/ lower frequency notes on bass and various pads - it's frustrating to have to adjust all the levels to lose the distortion. I obviously don't want to have to avoid using lower tones in tracks.
Am I stuck just filtering and adjusting eq's or is their an easier option?
Thanks in advance all.

What the best way is to avoid distortion w/ lower frequencies?
The likely source of your distortion is too hot of an output level. You might try using compression or limiting to keep those low-frequency levels under control. Other than that, you just have to adjust individual track levels so that you get a good overall level without distortion. EQ can also help if a certain frequency range is too hot compared to everything else.

One-Third Octave Filtering of Low Frequency Data

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