Re: Audio Sampling Question

On Sun, 22 Jul 2007 01:17:59 +0000, Guy Macon
<http://www.guymacon.com/> wrote:




Eeyore wrote:

Nico Coesel wrote:

Eeyore wrote:

Nico Coesel wrote:

Eeyore wrote:

Most modern audio ADCs don't require a front end filter.

Sorry, but that can't be true.

It is true. They have no analog front end filter.

If you are sampling, then you'll need to get rid of the
frequencies which are above fs/2 otherwise you will get
aliasing problems. Thats a law of physics like gravity.

Onboard digital filter.

The ADC oversamples. That's how they avoid aliasing.
It digitally filters at the oversampled rate and then
downsamples to 44.1kHz.

No analogue front end filter's required so no issues
with filter component tolerances. The digital filters
match perfectly of course.

I hate to be the one to break this to you, but you are
suffering from a fundamental misconception as to how
digital sampling of analog signals actually works.

*No* digital technique can act as an antialiasing filter.
It's impossible. By the time the digital filter gets the
information, a vital piece of information -- whether the
signal is an alias or not -- has been lost.

Oversampling is simply sampling at a higher rate, thus
moving the need for antialiasing higher in frequency.
It also allows an analog filter that interferes with
the highest portion of the signal more, because the
frequencies being affected are still higher than the
signal of interest. The digital filter takes out these
higher-than-the-signal frequencies but cannot perform
the anti-aliasing function. That is and always will
be, an analog function.

I'm remembering that a 64X oversampling Delta-Smegma A-D converter
needs a single order R-C LPF with a cut of around 3dB at the sample
frequency. Not ~much~ filtering, but a teeny bit.

boB




.

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