Converting square signal into sinusiod avoiding Fourier Transform

I have a clean digitized square wave signal with a frequency around 50 kHz and I would like to convert it into a sinusoid with the same frequency and phase (don't care about amplitude). I would like to do so avoiding the stantard approach that would be doing the Fourier tranform to measure the frequency and phase and then generating a new sinusoid. Are you aware of any other procedure?
Thank you!

What's up with the FFT-phobia? Is this some kind of homework with silly restrictions?
(Even the zero phase filter mentioned by Darin might use FFT like functions internally. )
Keeping up the facade, here's one way using "extract single tone". (Which also uses FFT internally).
Of course you could get a good initial estimate on the frequency and phase by measuring the zero crossings, then fit to a sine function with adjustable amplitude, phase and frequency using levenberg-marquardt. Suddenly, FFT seems like a better idea .
LabVIEW Champion . Do more with less code and in less time .
Attachments:
SquareToSine.png ‏20 KB

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Converting analog signals into digital signals

I have been transferring analog video cassettes to iMovie for further editing by using a digital camcorder (Canon MV960) between the 8mm video camcorder (analog camcorder) and iMac. I have always done this without any cassette in the digital camcorder, i.e. I have been using the digital camcorder as an analog/digital converter only.
I recently acquired iMovie HD 6 but this most recent version of iMovie refuses to import any signal from the analog camcorder via the digital camcorder. The window in iMovie is empty with an error message saying that "no tape". Well, it's true, there is no tape, there never was any tape when I used the previous version of iMovie!
Does this mean that I have to convert the analog video tape into digital video tape and, once recorded in digital format, the digital tape can be imported in iMovie HD 6?
All cable connections are OK:
- the analog film could be seen in the screen of the digital camcorder:
- digital video tape could be imported to iMovie HD 6.

Hi 5480:
try running a permissions repair (which should actually be done before and after upgrading software). You will find it in applications / utility / disk utility.
Also, this is a copy of how Matti Haveri (iMovie Guru) sets up analoge to DV:
My Analog to DV passthrough settings memo:
1. Take the tape out of the camcorder
2. Connect the analog device's audio and video to the camcorder's minijack, use the S-Video jack for video if the output device supports S-Video
3. Put the camcorder into VTR-mode
4. Select A/V->DV out: ON, in VTR mode's menu choices
5. Disable the info display on the the camcorder by toggling its DISPLAY button
6. Put iMovie (or other DV application) to the Camera Mode
The analog image should now appear on the camcorder's screen as well as on the iMovie's screen.
Note that if you have a tape in the camcorder, you have to put the camcorder's VTR into REC mode (you may pause the tape) but this is not advisable because it produces wear to the tape.
* Note that the SCART which comes with the European nEUtered TRV320E camcorder is Out-only.
Sue

Convert analog signals into I2C

hi all,
I am using NI-6351 DAQ unit.I want to capture two analog signal and I2S data from the hardware.
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Depends upon what computer you have. You'll need a player with an output line of some sort, then a converter to go from analog to digital (e.g. http://store.griffintechnology.com/catalogsearch/result/?q=imic). then recording software (free Audacity). Time.
Those are the basics. For higher quality you might want a soundcard that has RCA input and various software packages for cleaning up the audio.
How may cassettes? it may be cheaper to just buy the CD. There are also commercial services that will transfer cassettes. If you're just talking one or two I would go that route because you can spend a lot of time doing this and that will increase sharply if you haven't done this kind of thing before.

How to convert ethernet signal into wifi?

I am staying in an apartment this month that only has internet via direct-line ethernet. I need to get a signal to my ipad and iphone. Is there a way to use my macbook to create a signal that my iOS devices can detect?

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Converting square wave to double

I am trying to pass to square wave signal into math script. I am assumeing that i have to convert the signal into a double. Is this possible? If so how could I convert this signal so it can be passed into math script?
Thanks,
Leif

Which data type does your square wave signal has? If it is a waveform you can get the value as an array of double with the Get Waveform Components primitve.
Waldemar
Using 7.1.1, 8.5.1, 8.6.1, 2009 on XP and RT
Don't forget to give Kudos to good answers and/or questions

After fourier transform

Hi all, A simple question do ask. I have this input signal which is being generated by Signal Generator by Duration.vi. I got another output signal generated after applying the Fourier transform to the signal above.
I believe the signal generated by the first vi is to be a signal in the time domain, and the signal after applying the fourier transform, I get a waveform in the frequency domain. Is there any way where I can view the waveform under the same waveform graph?
Thanks a lot in advance,
Gareth

Hello Gareth,
Have you taken a look at this thread yet? It is about showing two waveforms on the same waveform graph. Briefly, you will need to use the Build Array function to combine the two data signals together (e.g. if you are using ExpressVIs, combine the two Dynamic Data Type outputs) using the Build Array function, and feed it into the graph. Some properties of the waveform graph needs to be modified to show separate axes for each signal.
While this is technically possible, it might not be a very feasible idea as the x-axis will be common to both waveforms. Thus, you might not be able to view both waveforms clearly at one go. I have attached an example in this post. "Bothsignals.jpg" show both the time domain and FFT signals together in the same graph. However, if we zoom into the green area, we can see the actual time domain signal ("timedomain.jpg"). The time duration is about 100ms. But for the FFT, the x-axis should represent frequency (Hz), and is centered at 50Hz, which is much bigger than 100milli.
Thus, I think it might be a better idea to separate the two. But this is a personal view.
Best regards,
Victor
AE, NI ASEAN
Attachments:
bothsignals.jpg ‏148 KB
timedomain.jpg ‏163 KB

How to convert a waveform signal into array of complex numbers

How to convert a waveform signal into array of complex numbers.

Hi Chaks,
try this:
(Get Waveform components, convert to complex DBL)
Message Edited by GerdW on 01-28-2008 09:23 AM
Best regards,
GerdW
CLAD, using 2009SP1 + LV2011SP1 + LV2014SP1 on WinXP+Win7+cRIO
Kudos are welcome
Attachments:
ToCDB.png ‏1 KB

Directly convert the analog signals into digital without passing through FPGA

To whom it may concern,
I am using a SCB-68 board R-series and I have a couple of technical questions:
In my system at the moment 8 analog channels are connected to the 8 input analog ports and then data are sent to a FPGA (PXI system 7852R). I am planning now to add other 4 channels to the system. Since over the total 12 channels I just need a very fast acquisition mode (using the FPGA) for 6 of them, while the other 6 can be sampled in a ms scale, I was wondering if you can suggest me something that I can use to directly convert the analog signals into digital and in this way used many unused digital signals provided with this model.
Thanks in advance.

yes it is correct, what do you need is:
software + (pc) + PXI + cable + chassis + C-series module
With embedded controllers, there’s no need for an external PC
With PXI remote control kits, you can control PXI systems directly from desktops, laptops, or server computers.
What Is PXI?
To configure a complete PXI System please go to: Build Your PXI System

Problems when converting square into button symbol

Hey, I got a question about converting square that has linear gradient no border and hard edge into a button symbol each time I do this the size of the symbol is always about 1px wider or taller or both, this doesnt happen when i try to do the same with square bitmap with filled color. (FW CS4)
Please explain if anyone got any clue whats going on with it.

Fireworks CS4, WinXP Pro:
I create a rectangle 100 x 100, linear fill, hard edge. Select > F8 > Convert to Symbol - button. Symbol is exactly 100 x 100 px after conversion to a symbol (button type) -- I cannot reproduce what you see...
Maybe someone else can help you better...

Fourier Transform of an Exponential Function

ok. this is something which i had tried earlier. I'm supposed to do a fourier transform of this signal.... exp(-alpha*t) where alpha is greater or less than 0.. the upper limit is inf and lower limit is 0. the waveform i got is a straight line, and there's this error -20003 i.e number of samples > 0.
I thought I had already defined the number of samples? Is that anything which I had missed out? Please see attached.
Attachments:
FT for real decaying exponential (eg 3-24) test v02.vi ‏25 KB

I cannot find your function exp(-alpha*t) anywhere in your VI. -- Where should I define this?
On the block diagram!
Where do you enter alpha? -- As above.
Maybe you want to add another control to the front panel.
Quote: "... where alpha is greater or less than 0...". Is that the same as "where alpha is not equal 0"? - Nope. Sorry poor definition there. The equation applies only when alpha is Smaller or equals to 0
Are you sure? So why would the function have negative alpha in the formula? A more reasonable assumption would be that alpha is positive and thus "-alpha" negative.
I assume that "t" is probably related to the array index. However, you are taking the plain exponential of each array value instead. Why? To reflect the original waveform into the waveform chart.
You talk about an exponential function, but you actually generate a sine function, which is not even mentioned in the problem description. Could it be you want a sine function with an exponentially decaying amplitude?
Why do you integrate? As the question goes, I need to find a decaying exponential over time. That's why I integrate the function.
Integrating a function does not generate a decaying exponential. Makes no sense!
What is the purpose if the derivative without signal input? Don't get your question
In the lower right of your diagram you have a second graph terminal connected to the output f the derivative function. There is no input wired to that function.
Why don't you put indicators on the error outputs? Alright, will do that.
What's up with the blue diagram constants. Make them orange to avoid coercions. - May I know how I can do this?
Easiest would be to just delete the constant, then right-click in the desired function input and select "create constant". Now you have the correct representation, guaranteed.
If you later want to change it, right-click the diagram constant and change the representation.
LabVIEW Champion . Do more with less code and in less time .

Fourier transform of rectangle function

We know that the fourier transform of a rectangle function is a sinc function.Now I want to use Labview to simulate
this process.How to do it? I input a rectangle function to FFT vi ,but the result is not a sinc function,why?

With the FFT you are attempting to approximate the FT of a continuous function using discrete sampling for a finite time. The approximation is "perfect" when the continuous function is bandwidth limited. The Rectangle function is not, you require an infinite number of terms in its Fourier series approximation.
A simple way to imagine the relationship between the finite sample on which you are going to perform the FFT and the continuous function you are approximating is to periodically extend the finite sample in positive and negative time. When you put in a simple rectangle pulse into the FFT you are seeing the approximation to the FT of a square wave with some duty factor, not of a single pulse. This probably explains most of the confusion.
The details of the relationship between the FT of a continuous function and the approximation via a discrete FT is detailed and subtle and you should probably study a signal processing or mathematics textbook.

Why not use Fourier transforms for filtering?

Are there any resources for constructing filters using discrete Fourier transforms (DFTs)? Or is it as easy as it seems. For example, for lowpass filtering I obtain the DFT of my signal, zero out components above the selected cutoff frequency and then transform back to get my signal minus the high frequency (noise) components. It seems to behave correctly for the signal I've applied it to, but am I missing something. Upsides would appear to be (1) no time delay between the input and output and (2) the filter is sharp with no transition band. Downside is that this method is slower than using impulse filters, but are there other upsides, downsides, or considerations I'm missing?

The technique of filtering using frequency domain truncation and IFFT is used in specific areas where computation time is essential (to avoid the time domain convolution with a long impulse response), like image processing. However the technique also has some downsides you should be aware of.
The frequency domain filtering performed on a data record is totally equivalent to a circular convolution made on your time signal with the impulse response of your ideal low-pass filter (typically a sin(x)/x function). The fact what we are doing a circular convolution will result in unwanted artifacts and you'll see some "parts" of the beginning of your signal at the end of the filtered signal and vice-versa.
Try for example to create a short signal (like a sharp pul
se) and extend it with zeros. After filtering you will see a sort of "pre-shoot" of your original signal at the very end of your signal. If you then rotate your time signal you'll discover that the signal is actually continuous (in a circular way).
You can reduce the problem by, for example, adding "some" zeros at the beginning and at the end of your signal (zero-pad) before your filtering operation (before the FFT) and then remove these additional samples again from you filtered signal (after the IFFT).

Real Fourier Transformation: confusing phase spectrum

When setting sample rate and block size both to 1024, the Real Fourier Transformation of a sin-wave with Frequency=1 and PhaseAngel=0 calculates the phase spectrum as displayed in the left part of the image.
Setting sample rate and block size both to 512, results in the phase spectrum as displayed in the right part of the image.
There are a few things I don't understand:
*) Why is the phase of frequency 1 equals -90, when I set the phase angel of the input signal to 0?
My explanation is that the phases are calculated relative to a cos-signal. Is that true?
*) Why does the phase spectrum contain more than one value, when the input signal just contains one single frequency?
*) Why do the phase spectra (is this the plural of "spectrum"?) look different for different sample rates / block sizes?
Maybe someone is able to answer these questions
Thanks!
Roman

This is not my area of expertise, so, I went to one of the experts, and he gave me this. I hope that it helps.
*) Why is the phase of frequency 1 equals -90, when I set the phase angel
of the input signal to 0?
My explanation is that the phases are calculated relative to a cos-signal.
Is that true?
The important information of the
phase spectrum is how the different sine waves forming a signal are shifted
in relation to each other.
In a typical signal analysis you
cannot force your analysis interval to be in phase with a specific sine
wave, so the offset of the phase spectrum
is arbitrary.
If you try to analyse the composite
signal of several sine waves with different frequencies, you will see that
they all have the same offset.
I would also have expected an
phase offset of 0 in this ananlysis, but it does not make a difference.
*) Why does the phase spectrum contain more than one value, when the input
signal just contains one single frequency
At all other frequencies the amplitude
value (and also the complex frequency value) should be 0. But 0 does not
have a defined phase.
The values come from rounding
errors and it cannot be determined by software whether there is a meaningful
information.
Typically you should only take
those phase values into account where the corresponding amplitude value
is larger than 0.
The software cannot determine
whether this is inherent to the signal ar an artifact of the rounding errors.
*) Why do the phase spectra (is this the plural of "spectrum"?)
look different for different sample rates / block sizes?
Same reason as above, different
settings yield different rounding errors.
- cj
Measurement Computing (MCC) has free technical support. Visit www.mccdaq.com and click on the "Support" tab for all support options, including DASYLab.

Digital (square) signal error

Hello
I have a problem with MultiSIM 9.0.155.
I have built a circuit shown on this picture http://img237.imageshack.us/img237/9764/digitaltoanalogcircutzw1.png.It is supposed to convert a digital (4bit) signal into analog signal (by it's value) so that when the digital signal is 0000 the analog value is 0V, and when it's for ex. digital sig is 1111 the analog value is of 2V... (this circuit uses a two 8-LED bars to indicate which "switch is on", and there isnt an analog signal output yet...)
This circut should work fine, but: When I tried to simulate a 4bit value with a four function generators (with frequencyes set to 1Hz, 2Hz, 4Hz, and 8Hz) the result is something like this:
http://img528.imageshack.us/img528/4268/digitaltoanalogcircutpoig1.png. In this picture the 4 keys are unpressed, and even then i still have a signal like that on the picture...??
I replaced four function gens with 4 batteries as a source but still the signal is the same??...-> an square alternating current with rising and falling edges...
Here is a closer picture with the batteries as a sourcehttp://img300.imageshack.us/img300/9616/digitaltoanalogsourcera4.png.
I don't know if something is not set properly (but it shouldn't get an square signal with the batteries as a source and the 4 keys unpressed), so if anyone sees what's the problem post an answer please..
I have tried to restart MultiSIM but still with the same results...

Hi there,
I'm not sure how you created this schematic, but somehow, the 4 nets (a.k.a wires) that you are measuring have the same name! When Multisim starts simulating, it treats all wires by the same name as a single electrically connected wire. That is why you saw the same signal on all 4 channels of your oscilloscope. There is a way to fix this:
Right click on one of the wires, lets say the one you labeled bit 0
Click properties
Change the net name to b0
Click OK
Multisim is going to tell you that you have other wires that have the same name but are not connected. Tell Multisim that you do not want to change the names of the other wires.
Now repeat this for the remaining 3 bits, but name them b1, b2 and b3
I bet there may be other wires in your design that have the same problem. Unfortunately, I think you will have to manually go through your design and find them all.
If you ever find out what you did to make this happen, please let us know. We are always interested to know how to make Multisim easier to use so that mistakes like these don't happen.
Hope that helps!
Message Edited by yyao on 02-05-2009 05:26 PM
Yi
Software Developer
National Instruments - Electronics Workbench Group

Control a 3 phase motor and convert input voltages into pressure values.

Hi,
I'm in my 4th year at university and I have been given a project that involves the experimental investigation of the interaction between fishing gears and the seabed. The work will consist dragging full-size or scale trawl components along a channel containing typical seabed sediments. I aim to create a LabView program that will allow me to operate the motor and analyze the 6 transducers I will be using.
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1 - Start the motor.
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I have version 8.5 if anyone has any information or advice i would greatly appreciate it .
Thanks,
Grant.
Message Edited by grantstephen on 03-14-2008 11:28 AM

You should try in the LabVIEW forum. This forum is for Employers with LabVIEW job openings.
http://forums.ni.com/ni/board?board.id=170
=============
XP SP2, LV 8.2
CLAD

Converting square signal into sinusiod avoiding Fourier Transform

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