[MPlayer-users] A question...

Jon Larabee jlarabee at greenapple.com
Mon Feb 10 00:25:54 CET 2003


Apperciate the help... A few things I was wondering:

1) What is the threshold for the 16 bit value? Is it supposed to be zero?

2) Is the frequancy that it is sampling at always going to be the same? If
so could I do the following for better plotting?

	Amplitude (the sample scaled by some factor) * cos(2*pi*(frequancy
of sampling rate * time) = y

In which case I would assume that the frequancy it is sampling at would
never change ie it being the sampling rate, and that the time can be
determined by the position in the file (ie 22,000 hz/second rate is
basically 22,000 samples / second) I don't know if this would work, and
you sound like you know a heck of a lot more about it than I do ;)

On Sun, 9 Feb 2003, D Richard Felker III wrote:

> [Automatic answer: RTFM (read DOCS, FAQ), also read DOCS/bugreports.html]
> On Sun, Feb 09, 2003 at 10:19:19PM +0000, Jon Larabee wrote:
> > Which is a little endian and is defaulted to 16 bit samples. I have been
> > using a signed short to pull them out of a file after recording for
> > analyzation. This has worked quite well indeed. I have svga graphing
> > software I have written to handle and scale the values, it makes a pretty
> > picture, but unfortunatly... I am having trouble conceptionalizing what it
> > is supposed to mean. From the OSS site, the proposed meaning of a sample
> > is the volume of the signal when it is measured. The thing is... How does
> > one have negative volume? The threshold I imagine is zero, or is it that I
> Well volume is a bad word for it; the best word is probably amplitude.
> It can be positive or negative because voltage can be positive or
> negative (i.e. higher or lower potential in the signal line than in
> the ground) and because pressure on your eardrum can be inward (above
> the pressure inside your ear) or outward (below internal pressure) as
> the vibration of the sound goes back and forth. Which direction is
> which isn't terribly important.
> > am simply looking at it wrong and that some other value is the threshold
> > that I should be concerned with? Is there a way I can relate this value to
> > frequancy? Any help anyone could offer in this direction would be most
> This is a very good question. There is no direct way to get the
> frequency since the concept isn't even really well defined. Chopping
> up the sound into blocks and applying a fourier transform of some sort
> is the most common way I know of extracting frequency information.
> There are also various filters and wavelet type techniques I don't
> really know much about.
> Anyway, if you're seriously planning on analyzing sound like you say,
> you really need to get yourself a good text on the subject and read
> up...
> Good luck!
> Rich
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