Re: AIC23 _ High Pass Filter_BIG_FUNDAMENTAL_PROBLEM

Akash_DSP wrote:
Akash_DSP wrote:
Akash_DSP wrote:

...

Hey Jerry,

thanks for the reply.

Yes, the periodic vairation is sinusoidal. This means that if I give
an
audio input with a frequenct spectrum (<20 KHz), frequencies that
will
have
come out of the D/A will have different delays.
Different delays for different frequencies, of different delays for
the
same frequency at different times?

...

Jerry
--
Engineering is the art of making what you want from things you can
get.
Hey Jerry,

it is different delays for different frequencies at the same time.
It seems that you have discovered the dispersion of the anti-alias filter. That is normal.

Jerry
--
Engineering is the art of making what you want from things you can get.


Hey Jerry,

you mean to say that if I give an audio input and observe at the output
that each frequency comes out with a different delay. That is normal?

If you program your C6416 or c6713 do you see similar behaviour?

In order to sample successfully, there must be no components of the sampled signal as high as half the sampling frequency. The use of an anti-alias low-pass filter in the signal path before the sampler is so nearly universal that many codecs have a filter built in. Being analog, the a-a filter will affect the phase. The designer is responsible for correcting any phase distortion that interferes with proper the function of his design.

Examining the data *** should show you how much phase shift to expect. Once you plot the filter's phase/frequency characteristic, you will know if what you see is normal.

Oversampling is an effective way to minimize phase shift in the band of actual interest. After sampling, the signal can be further filtered digitally, then decimated.

All real design requires balancing competing objectives; that is an art.

Jerry
--
Engineering is the art of making what you want from things you can get.
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