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Split waveform, psychoacoustic analysis

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Hello I am writing here as my last resort as I am trying to get all of the sound quality results for each data point in my waveform, that was measured with a 192kHz sampling frequency, the length of the measurement sample itself is 30s. As the result I am looking to split 30s into 300 1/10th's as I have successfully split other measurements such as skewness.

 

The main problem I've been running into is that every iteration results in only the first data point being calculated by the given sound quality standard, when I change the number of iterations of the for loop I just get the same result pasted over and over for the whole length of the array. The main difference is that with sound quality I am working with waveforms and my knowledge ov LabView is already shoddy at best when working with regular.

 

TLDR: I would like to get an output of 300 data results from an input of 5760000 data points in 30s but unsure where to continue since my knowledge is very limited.

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I can see the problem quite clearly: You've coded it incorrectly.  Just fix the code and it will be fine.

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Solution
Accepted by topic author Kathoshi

See if this gives you an idea on how to solve your problem,

santo_13_0-1734450325828.png

 

Santhosh
Soliton Technologies

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Message 3 of 6
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So, we can guess that you are working with high resolution audio since you mentioned 192KHz, but other than that, we have no idea what you are trying to do.  🙁

 

Splitting your waveform into 30 slices is easy, but why are you doing it?  What is the "output" you are trying to "report"?

 

Please help us help you:

1)  Post your current code.  You clearly have a problem indexing points out of your waveform array.

2)  Post the actual waveform so we can debug and run your code.

3)  Tell us what you are trying to achieve with your analysis.

4)  Tell us if you are using toolkits (Sound and Vibration?)

LabVIEW Pro Dev & Measurement Studio Pro (VS Pro) 2019
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@Kathoshi wrote:

I have successfully split other measurements such as skewness..


So how did you do that and what is preventing you from doing the same here?

 


@Kathoshi wrote:

The main problem I've been running into is that every iteration results in only the first data point being calculated by the given sound quality standard, when I change the number of iterations of the for loop I just get the same result pasted over and over for the whole length of the array. The main difference is that with sound quality I am working with waveforms and my knowledge ov LabView is already shoddy at best when working with regular..


If this is the main problem, it would help to also tell us the secondary problems. As has been mentioned, you are obviously doing it wrong, but since you are not showing what you are doing, we cannot help. There are gazillion ways to do it wrong and thousands of ways to do it right. You need to narrow it down.

 

The correct solution will be general and  independent of sound quality or sampling frequency.

 

Can you also define the ambiguous terms you use ("sound quality standard", "pasted", "regular", etc.)

 


@Kathoshi wrote:

Hello I am writing here as my last resort as I am trying to get all of the sound quality results for each data point in my waveform, that was measured with a 192kHz sampling frequency, the length of the measurement sample itself is 30s. 


A single datapoint has no sound quality. (no frequency, no noise, no nothing!).If you want a result for each point, you need a suitable window of data around that point. If you just split into sections, you will have significantly fewer results than points.

 

 

 

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Thank you so much for your help, this was exactly what I was looking for, now I can append everything to the final array without any caveats inbetween.

 

I also want to thank everyone else in this beautiful community because of the fast responses and lending your free time to solve a novice's problems.

 

KR K.

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