Maximizing Your Sound System: Understanding FFT and Low Frequencies"

[sophiecentaur]

This is confusing. If you are dealing with the acoustic input from a microphone then what has the source data got to do with it?
I still don't know why you need to use acoustic connection. What's wrong with a double ended jack lead?
If you are using C, then you can surely extract alternate samples from any audio file you want to analyse. There seems to be a'big hole' in all this and I'm not sure what's missing.

It could help you and us if you were to draw a block / function diagram of what you are doing. There just has to be a straightforward answer to your basic problem.

 

[btb4198]

Ok so

I am using a built in computer microphone. so there are no connect to be make.

And I am using a Dell Computer...

now my problem comes in with noise in my input ...
so there no noise in my room but I get this :

and this is which
Math.Abs(F.Magnitude) > 300000

 

[sophiecentaur]

You need to go right back to square one and check out each individual link in the chain.
If you record 'silence' and then a burst of high level 1kHz, what does the straight play back sound like? Noisy? Also, what does high level 1kHz look like when plotted on those two graphs ("my test graph")?
The make of the computer should not be important for this. If it has an internal microphone, does is have an identifiable sound card or does it not have 'slots'?

 

[btb4198]

ok here is how it looks:

 

[sophiecentaur]

You missed out the y-axis on the latest pics. but it looks ok. However, I wonder if you have just set the gain of the mic to auto and you are just seeing a scaled up version of the most recent pictures in the previous pictures.
You should find that somewhere in the options.

 

[btb4198]

ok sorry about the last pic:

now as for the setting of the Microphone there are not auto gain that I can see:

is there any other place I should look?
also do you know what Microphone Array does ?
wiki say's is it a "Systems for extracting voice input from ambient noise (notably telephones, speech recognition systems, hearing aids)"... and that is what I am trying to do..

 

[sophiecentaur]

There must be some level control somewhere - that could still be your problem.
I have no idea about the "microphone array" but you could google PC Sound Control Microphone array.
I know the Apple user interface is very simple but limited (that's what I'm used to) but some of the Windows panels are total gobbledegook to me. I suggest you try the 'Levels', 'Configure', 'Properties' and you must find an audio input level adjust.
What software package is taking the sound into record it? Is there not a control there? The point is that the levels just don't seem to make sense. The noise seems to be higher on 'silent' than with the 1kHz - which implies some expanding is happening. Did you listen to the recording (on headphones or your friend's HI Fi) - you will need loads of Bass because the high level shash on your graph is all at 40Hz and below.

Your problem is that you are trying to do something sensible with some 'toy' kit for listening fun. Home stuff always falls short, one way or another. You just have to battle it out.

BTW, Microphone Arrays are used for directionality (as with radio antenna arrays). You have only one mike so I don't think that's relevant. I have two mikes on my deaf aids and one switch position claims to reduce sounds from behind. It uses a phase shift between them. It does work sometimes but never very well. Is that a tick I can see on the Mic Array setting on the control panel? I don't think that should be selected for serious stuff.

 

[btb4198]

ok i did not see any

 

[sophiecentaur]

I refer to the picture of the control panel in Post 36.