Re: Audio Sampling Question

On Jul 21, 3:51 pm, n...@xxxxxxxxxxx (Nico Coesel) wrote:
Eeyore <rabbitsfriendsandrelati...@xxxxxxxxxxx> wrote:

Nico Coesel wrote:

Eeyore wrote:
Guy Macon wrote:
Henry VIII wrote:

I'm sampling high-fidelity analog audio at 44.1 kHz with a 16-bit ADC. The
analog audio is noise-free for the purposes of this question.

The ADC data stream goes to a microprocessor that compares the data in
blocks of multiple samples to a previously stored set of data. When the ADC
output data matches the stored data, the microprocessor generates an output
pulse. Some amount of processing time "X" is needed to recognize a match
and generate the pulse.

My question, again assuming zero analog noise, is: what is the time
uncertainty of the output pulse? In other words, if I split the same analog
audio into two of these circuits in parallel, how much could their output
pulses differ in time? Is the answer simply the clock frequency accuracy?

There are several possible sources of variation. How bad each one
is depends on your hardware and possibly on the nature of the signal.

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.

--
Reply to nico@nctdevpuntnl (punt=.)
Bedrijven en winkels vindt U opwww.adresboekje.nl

No, not true. It's only a problem if the aliased frequencies fall
within the band of interest. In a delta-sigma converter (used almost
universally for audio these days), the sampling is typically at some
MHz, and a digital filter wipes out everything above 20kHz
(generally). So it's only frequencies within 20kHz of the sampling
rate or its harmonics that alias to frequencies which fall within the
passband of the digital filter.

A specific example: a delta-sigma converter that oversamples at 64
times the output rate, which is 44.1ksamples/sec. 64*44.1kHz =
2.8224MHz. 2.8224MHz - 20kHz = 2.8024MHz. Obviously, normal audio
transducers shouldn't be putting out any significant signals up there,
but if you want to claim alias protection to some level in the event
that users DO put in some high frequencies, you only need to provide a
rolloff that starts somewhere above 20kHz and drops to the desired
point by 2.8MHz--a much easier task than in the "old days" of
converters that sample directly at 44.1kHz. Instead of needing a high-
order (e.g. 11th) elliptical filter that cuts off by 24.1kHz, you can
get about 90dB protection from a third order Butterworth that's only
down 0.1dB at 20kHz.

Cheers,
Tom

.

Quantcast