darrenyeats;675483 Wrote: 
> Yes, I see your point. What I meant is we don't hear sound as a series
> of pressure values. Instead, the sound is perceived as a set of
> frequencies. Each of these perceived frequencies will have a loudness
> yes, but that's a different thing.
> 
> One is time domain, the other is frequency domain. This is what Fourier
> transforms are about.
> Darren

I'm not sure, but I think we strayed from the point.
Let's look at from the playback side since I think this is is easier to
visualize.  Let's say the correct output value of the DAC at a
particular instant in time is 9.5 but due to the resolution limits of
the DAC, it can only be either 9 or 10.  This means you will have an
error of 0.5 out of 9.5 which is almost 5% distortion.  
There is nothing that can be done about it.  The DAC can only output 9
or 10, never the correct value of 9.5.  Dithering cannot fix this.
If the signal was steady for long enough for several samples to be
taken of the value, then dithering does help.  The random noise added
during recording that causes the dithering will result in the signal
being recorded as a random set of 9s and 10s.  Then during playback,
this random set of 9s and 10s would average out (after the low pass
filter) to 9.5.  The distortion is fixed.
But how does this work when the value isn't steady enough to get
multiple readings of the value before it changes?  The DAC simply
cannot output the correct value.  The input value to the DAC can only
be 9 or 10.  It cannot be the correct value of 9.5.  That number
doesn't exist in this DAC's numbering system.
Imagine the stream of values that should be representing the music
waveform.  How can this ever be anything more than a constant stream of
wrong (approximately right) values.  The will on the average be wrong by
one half the minimum step size of the DAC resolution.  I don't see how
dithering can fix this unless we are talking about a repetitious signal
so that we can average the errors over multiple samples.

Terry

Terry


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