Dear users,I am trying to scan the WAV files and compute the sound level parameters based on the recorded calibration signal. My question regards the blosk size of samples used to analyze longer files. While the linearly averaged values do not vary depending on the block size, the exponentialy averaged values do, and so do parameters calculated based on exponentialy averaged sound level. What would be more appropriate, analyzing the files in longer blocks (22050, 44100) or smaller ones (8192).Another thing, while obtaining the Leq values, 0 values sometimes occurs, how can I avoid this? Does anybody know how to calculate the Ln percentile levels used for traffic noise?Thanks

Signal Conditioning

Sound Level from calibrated WAV files

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Hi Travis,
Here are the two VIs: 01 BEFORE is the VI which works for small wav files. 02 AFTER is the VI modified to read the wav file in chunks. Plese take a look and see if you can make something of it. Thank you for the time you invested into my problem!
Milan

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Signal Conditioning

Sound Level from calibrated WAV files

Re: Sound Level from calibrated WAV files