Re: Audio Sampling Question


"Eeyore" <rabbitsfriendsandrelations@xxxxxxxxxxx> wrote in message
news:46A2AC19.40ABF49C@xxxxxxxxxxxxxx


Nico Coesel wrote:

Eeyore wrote:
Nico Coesel wrote:
Eeyore wrote:
Guy Macon wrote:

Are the analog filters identical? A slight difference in phase
shift will introduce an error. Even if you use the same analog
input and compare measurements at different times there could be
variations due to temperature.

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.

Graham


You have no clue what your talking about... what a suprise!!!


The oversampling is to reduce the complexity of the analog filtering. You
get aliasing no matter what unless you can be 100% sure that your signal's
bandwidth is 1/2 the sampling rate. If there is any spurious noise then that
will be reflected back onto the first nyquist zone and degrade the signal.

By oversampling, say, 64 times, then we have 64x as much room to roll off
the highs. Then we can digitally use a FIR low pass that removes all the
noise above the signal itself.

Since the gain of a simple low pass filter is rolls off about 6dB every
octave, every time you double the bandwidth(sampling rate) of the sampler vs
the bandwidth of the signal you get a 6dB roll off of the max frequency
which means you reduce all higher frequencies that normally would be aliased
by k*6dB.

If the bandwidth of the signals you are working with is B and you oversample
it k times, then any aliasing will be below -k*6dB. Of course now your
working with a signal that has a much larger bandwidth than what you
actually need so you downsample to reduce it back down to the original
bandwidth. Thats not before you digitally filter the signal to remove all
the frequencies between B and k*B so they do not get aliased by into the
signal. Its called decimation.

Obviously this is just another example of you talking about your ass. Maybe
one day you'll get a clue and actually learn something or at the very least
clean the bull*** comming from your mouth.

Jon







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