Digital Signal Processing

Hi all.
Recently, I've been busy with a project on signal processing. I have no experience in this field so i am clueless about how to do it.
I've been searching the internet about his topic and i keep getting the same results. So i would just like to ask you some questions to clear things out.
I've been able to create a program which records sound and stores the samples in a buffer. I want to analyze this sound and see which frequencies are present in it. I keep getting posts on Fourier analysis, but i don't understand why exactly i need it, or more precisely, what to do with it.
As i understand, the sequence of samples which i've gathered are hard to analyze, and so, i need to convert it to another series with FFT or DFT. I have found a peace of code which takes as an input the series, and spits out an output which i have no clue how to read or what it represents. The code is basically this: http://people.csail.mit.edu/hammond/teaching/hide/fft/fftcode.html
I get two columns of numbers, A's and B's, but i don't know what they are supposed to represent. What do i do with this output to check which frequencies are present in the signal?
I hope someone can guide me to the right direction.
Thanks.

Thanks KlausJJ, my program can now identify frequency magnitudes of a sound. I struggled a bit with the FFT, then i changed the FFT code i had mentioned earlier, and used another one, and it's working fine.
I now have another important question. I want to identify a clap sound. I'm not sure how to do this. I'm not sure if i should use FFT or not. If i want to compare a sound patterns produced by a clap, do i compare the frequencies or simply the clap waveforms. I'm thinking i can compare the frequency magnitudes of the claps for the frequency ranges. What i wanna do is get the 20 highest frequencies of a clap, and see if another sound is a clap by looking at its 15 highest frequencies. If those 15 freq ranges are present in the original clap's 20 highest frequency ranges, then i confirm that it's a clap. Or maybe i should compare the 15 highest and lowest?
Is this method effective? Or is there a better way of comparing patterns?
Thanks.

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Is there any tutorial for using ARM cortex-A processors of Zynq for digital signal processing ?

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Hi
Check below links
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Digital Signal Processing Technique

Dear Members
The problem I have got in my term paper is to input a sound sample by a sound card(microphone)and acquire the digital sampled frequencies of that sample.In order to obtain DFT of that signal for recognition.
Which API or method should I use to develop in JAVA.

javax.sound might be what you want.

Help with digital signal processing app

I have been programming in CVI for nearly 7 years now, using general purpose
National Instruments DAQ boards.
I have an application where I need to mimic an analog processing stage digitally,
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Thanks!

Thanks KlausJJ, my program can now identify frequency magnitudes of a sound. I struggled a bit with the FFT, then i changed the FFT code i had mentioned earlier, and used another one, and it's working fine.
I now have another important question. I want to identify a clap sound. I'm not sure how to do this. I'm not sure if i should use FFT or not. If i want to compare a sound patterns produced by a clap, do i compare the frequencies or simply the clap waveforms. I'm thinking i can compare the frequency magnitudes of the claps for the frequency ranges. What i wanna do is get the 20 highest frequencies of a clap, and see if another sound is a clap by looking at its 15 highest frequencies. If those 15 freq ranges are present in the original clap's 20 highest frequency ranges, then i confirm that it's a clap. Or maybe i should compare the 15 highest and lowest?
Is this method effective? Or is there a better way of comparing patterns?
Thanks.

I am using the Order Analysis Toolkit and want to get more information about the compensation for "Reference Signal Processing", which is scarce in the manuals, the website and the examples installed with the toolkit.

I am using the Order Analysis Toolkit and want to get more information about the compensation for "Reference Signal Processing", which is scarce in the manuals, the website and the examples installed with the toolkit.
In particular, I am analyzing the example "Even Angle Reference Signal Processing (Digital Tacho, DAQmx).vi", whose documentation I am reproducing in the following:
DESCRIPTIONS:
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INSTRUCTIONS:
1. Run the VI.
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1. SVL Scale Voltage to EU.vi
2. OAT Digital Tacho Process.vi
3. OAT Get Even Angle Reference.vi
4. OAT Convert to Even Angle Signal.vi
5. OAT Compensate Even Angle Signal.vi
My question is: How is the synchronization produced at the time of the compensation ? How is it possible to eliminate the errors in a synchronized fashion with respect to the surface of the shaft bearing in mind that I am acquired data at a low rotation speed in order to get the "even angle reference" and then I use it to remove the errors in the vibration signal acquired at normal operation. In this application both operations are made in different acquisitions, therefore the reference of the correction signal is lost. Is it simply compensated without synchronizing ?
Our application is based on FPGA and we need to clarity those aspects before implementing the procedure.
Solved!
Go to Solution.

Hi CracKatoA.
Take a look at the link bellow:
http://forums.ni.com/ni/board/message?board.id=170&message.id=255126&requireLogin=False
Regards,
Filipe Silva

Signal Processing and LabVIEW Expert Full Time Position – London, UK

1x Signal Processing and LabVIEW Expert - Full Time Position – London, UK
1.0      Description:
As a Signal Processing/Software Engineer, you'll be part of a team that is responsible for developing applications based on large high speed data analysis for a large green energy project. Your skills must include a knowledge of Waveform analysis and understand the concepts of cross correlation, Fourier Analysis, Spectral analysis and other similar items.
The position ideally requires someone of CLA level of LabVIEW program to act as the project lead on the programming and technical levels of the system.
2.0      Duties and Responsibilities:
The duties and responsibilities of this position include, but are not limited to, interacting directly with customers to program, train, inform, advise and consult in the design of their applications using the best test and measurement hardware and software by:
◦     Programming memory and speed efficient applications using LabVIEW and Object Oriented Programming and Event-based architectures.
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4.0      Location:
This role is located in London, UK
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Signal Processing Engineers will report to the Technical Director.
6.0      Salary:
Competitive salary based on level of experience.
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Hi,
I am from india.I can able to complete the projects through the internet itself.
Please let me know if any demos required to show my expertation.

Measure Frequency of Digital Signal from DAQ-Board

I'm using a National Instruments DAQ board and I need to measure the
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I know the measure frequency.vi (In diagram - data acquisition -
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Thanks in advance,
[email protected]

Hi,
Attached is a vi which models your signal, and provides a way to measure the average frequency over a window of time. It converts the signal to a sequence of periods. I usually start with this approach because you can save the data in a much more compact form if your sampling rate is high. Obviously over a specified period of time, if there are no zero crossings the frequency is zero, so I was a little incoherent before. The real issue is the latency caused by averaging. Your frequency measurement is most accurate half your window size before the current time, so smaller windows give you more up to date measurements, but also more noisy. If you need the most up to date measurement, then you have to use the time since the last crossing which is the case where the frequency never goes to zero. That's probably not a realistic situation. If you needed less latency, you could use a prediction algorithm.
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Attachments:
DigRandFreq.png ‏62 KB
DigitalRandomFreq.vi ‏34 KB

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Hi all,
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Dear Norbert_B.
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shof55,
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Solved!
Go to Solution.

idma-
Check out the LabVIEW FPGA example http://zone.ni.com/devzone/cda/epd/p/id/6126
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This question was solved.
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Hi,
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This question was solved.
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Hi:
Please see if installing the Intel Chipset driver installs that device.
https://downloadcenter.intel.com/Detail_Desc.aspx?DwnldID=20775&lang=eng&ProdId=816

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Solved!
Go to Solution.

See the attached file
Attachments:
Digital Signal.vi ‏335 KB

How to input/output a digital signal and acquire an analog signal at the same time?

Dasylab, version: 8.0.04
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by the way, I come from China, my English isn't good, I'm sorry.
Allen, China.

Hi,
Have things changed over the years?
I need to syncronise a digital output (Modul NI9474) and an analoge input (AI-Modul NI9203) module. I need to measure time intervals from a flank in signal A to a flank in signal B. I would like accuracies of the order of 1 ms. Currently, the signals are not synchronised, with errors of the order of 2 times the block length (block size x sample rate), sometimes much higher. The best I got so far was a block size of around 20 with a sample rate of 1 kHz.
If I use the master and slave settings on the RTSL settings, my program doesn't run properly.
If I use digital signals for input and output, I can syncronise them with RTSL settings and everything is good, but I can't always do that.
Also, if I do anything in the GUI (such as scrollowing something or going to another window), my output gets screwed up properly.
1. What can be done to synchronise AI with DO?
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Thanks in advance!

Digital Signal Processing

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