Hi,
In order to do frequency analysis in LabVIEW for your sound sources, you need to convert the time domain input signal (for your case the array you got from your sound file or your sound card)to frequency domain using FFT(fast Fourier transform) and you also need to know the sampling period (or sampling frequency) of your sound card or your sound file. Fortunately LabVIEW already has this function to convert signals from time domain to frequency domain. I think the vi "Amplitude and Phase Spectrum.vi" is closest to what you need. This vi is located in Functions->Analyze->Signal Processing->Frequency Domain group. What you will see is the spectrum of the input signal in the form of amplitude of each frequencies in your signal and the phase shift for each frequenc
y.
The amplitude will represent the loud level of each frequency (wie laut ist die Sound). If you don't care the phase shift, you can ignore it for your statistic analysis. From the output of "Amplitude and Phase Spectrum.vi", you can get the maximum value of "Amp Spectrum Mag (Vrms)" array using "Array Max&Min" function in Functions->Array group, and you can also get the index of this maximum value, using this index you can calculate which frequency it is. To display, you can use XY Graph, X axis will be frequencies, Y axis is the amplitude.
To get a resolution of 1HZ, you need to sample at least once in a second, so the sampling period should not be greater than 1 second. To normalized to the input tone power (gerundet auf die Tonleiter), just divide the amplitude to the input tone power.
If you need more information, post again and I will do my best to help if I can.
Irene
irene_he@hytekautomation.com
www.hytekautomation.com
