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Audio Sampling Question

E

Eeyore

Jan 1, 1970
0
Jon said:
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 bullshit comming from your mouth.

Look here sunshine.

The circuit for the oversampling ADCs I've been using does not have any front
end filering that applies to THE AUDIO BAND.

Yes, it has a simple RC 1st order filter @ ~ *** 75kHz ***.

There WILL NOT be any issues of component tolearance causing phase or amplitude
mismatch to an AUDIO SIGNAL. The required filtering for the audio signal is
applied post sampling by a high accuracy digital filter.

And that will match from channel to channel. End of story.

Graham
 
E

Eeyore

Jan 1, 1970
0
Tom said:
Sorry, Graham, that's not quite true. Because the sampling is at a
high frequency in a delta-sigma converter, typically a couple MHz, the
alias filtering can be very "gentle", but it IS necessary if there's
any chance of high frequencies getting through otherwise.

By which you mean ultrasonic frequencies (close to the Nyquist limit imposed by the
over-sampling) though. NOT audio frequencies.

If used strictly for audio frequencies, a front end filter truly isn't required. The
rather gentle 6dB/octave LPF most application circuits suggest is simply to filter
any out of band signals. The recommended values for the ADC I've been using give you
a 1st order LPF @ 75kHz. That assuredly isn't going to give filter match issues in
the audio band. And, as I said, it could be left out if you're confident of the
bandwidth of the incoming signal ( i.e it doesn't contain anything in the megahertz
region.

Graham
 
V

Vladimir Vassilevsky

Jan 1, 1970
0
Eeyore said:
Jon Slaughter wrote:




Look here sunshine.

The circuit for the oversampling ADCs I've been using does not have any front
end filering that applies to THE AUDIO BAND.

Yes, it has a simple RC 1st order filter @ ~ *** 75kHz ***.

There WILL NOT be any issues of component tolearance causing phase or amplitude
mismatch to an AUDIO SIGNAL. The required filtering for the audio signal is
applied post sampling by a high accuracy digital filter.

And that will match from channel to channel. End of story.

Not quite.

1. The first order RC at 75kHz has the attenuation of 0.3dB at 20kHz.
This doesn't look good on the spec sheet.

2. The attenuation of aliases is about 38dB assuming the typical
performance audio ADC with the sample rate of 6.144MHz. Will fail the
EMC susceptibility test unless there is some other filtering on the way.


Vladimir Vassilevsky

DSP and Mixed Signal Design Consultant

http://www.abvolt.com
 
E

Eeyore

Jan 1, 1970
0
Vladimir said:
Not quite.

1. The first order RC at 75kHz has the attenuation of 0.3dB at 20kHz.
This doesn't look good on the spec sheet.

Ok, make it a 150kHz filter in that case. It's largely irrelevant.

2. The attenuation of aliases is about 38dB assuming the typical
performance audio ADC with the sample rate of 6.144MHz. Will fail the
EMC susceptibility test unless there is some other filtering on the way.

It won't fail any EMC tests you utter clot. What utter and complete tripe !

Graham
 
M

MooseFET

Jan 1, 1970
0
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.

You flew right over an important point about a delta-sigma converter.
The workings of the converter supresses signals above several times
the intended clock rate of the converter. The band that matters the
most is the one near the clock rate.
 
N

Nico Coesel

Jan 1, 1970
0
Phil Allison said:
"Nico Coesel"
"Eeysore PITA Fool"


** The A to D sampling rates used in audio are typically 96 kHz, 192kHz or
even higher when Sigma Delta modulation is used.

Plus all audio signals are band limited to not much over 20 kHz by the use
of microphones and their associated pre-amps.

Makes the use of dedicated, audio band limiting filters redundant in most
cases.

Like I said earlier in this thread: the anti-aliasing filter comes for
free because of the limited bandwidth of the preamps etc.
 
G

Guy Macon

Jan 1, 1970
0
Nico said:
Like I said earlier in this thread: the anti-aliasing filter comes for
free because of the limited bandwidth of the preamps etc.

One thing to be careful of is to not depend on your analog
amplifier stages for anti-aliasing unless they are a true
low-pass filter. If they are slew-rate limited, small
amplitude spurious signals that are above the Nyquist limit
can end up aliased. It doesn't take much to turn a 16-bit
ADC into a 14-bit ADC.
 
P

Phil Allison

Jan 1, 1970
0
"Nico Coesel"
"Eeysore PITA Fool"
Like I said earlier in this thread: the anti-aliasing filter comes for
free because of the limited bandwidth of the preamps etc.



** No - you fucking wog dim wit.

It comes mostly for free because of the sharply limited bandwidth of
microphones, music and speech.

The main aliasing worry with 192+ kHz sampling is residual, supersonic white
noise contributed from some mic or mic pre-amp.



....... Phil
 
E

Eeyore

Jan 1, 1970
0
Nico said:
Like I said earlier in this thread: the anti-aliasing filter comes for
free because of the limited bandwidth of the preamps etc.

Really.

My audio preamps typically have -3dB points in the 100kHz + region. That's quite
normal these days actually.

Graham
 
E

Eeyore

Jan 1, 1970
0
Phil said:
"Nico Coesel"
"Eeysore PITA Fool"

** No - you fucking wog dim wit.

It comes mostly for free because of the sharply limited bandwidth of
microphones, music and speech.

The main aliasing worry with 192+ kHz sampling is residual, supersonic white
noise contributed from some mic or mic pre-amp.

Phil speaks the truth.

Quite simply, there are few musical signals with a content above 20kHz and fewer
still transducers that might pick them up.

Graham
 
G

Guy Macon

Jan 1, 1970
0
Eeyore said:
Quite simply, there are few musical signals with a content
above 20kHz and fewer still transducers that might pick them up.

There are, however, plenty of EMI signals with a content
above 20kHz and the wires to those transducers might very
well pick them up.

With 192 kHz sampling and a 20 kHz signal, you don't need a
very sophisticated or expensive antialiasing filter, but not
having one at all is like playing with fire; eventually you
will get burned.
 
P

Phil Allison

Jan 1, 1970
0
"Guy Macon"
Eeysore said:
There are, however, plenty of EMI signals with a content
above 20kHz and the wires to those transducers might very
well pick them up.


** The universal use of twisted pair, balanced, shielded audio lines
basically precludes such injection - the sources of which are normally at
a very low level in any case.


With 192 kHz sampling and a 20 kHz signal, you don't need a
very sophisticated or expensive antialiasing filter, but not
having one at all is like playing with fire; eventually you
will get burned.


** If only to preserve the best possible s/n ratio.




......... Phil
 
N

Nico Coesel

Jan 1, 1970
0
Eeyore said:
Really.

My audio preamps typically have -3dB points in the 100kHz + region. That's quite
normal these days actually.

So they still act as an anti-aliasing filter when sampling at high
frequencies (>1MHz).
 
E

Eeyore

Jan 1, 1970
0
Guy said:
There are, however, plenty of EMI signals

EMI signals ?

with a content above 20kHz and the wires to those transducers might very
well pick them up.

Have you never heard of screened cable and balanced (differential) circuits ?

With 192 kHz sampling and a 20 kHz signal, you don't need a
very sophisticated or expensive antialiasing filter, but not
having one at all is like playing with fire; eventually you
will get burned.

Most modern sampling is way above 192kHz.

I suggest you keep to stuff you know about (although it seems that may not be
very much).


Graham
 
P

Phil Allison

Jan 1, 1970
0
"Eeyore"
Guy said:
Most modern sampling is way above 192kHz.


** Obviously " modern sampling " = that done above 192 kHz.

So there are not many " modern " A to Ds in service - only the old hat
kind, according to the Graham Stevenson charlatan.





........ Phil
 
E

Eeyore

Jan 1, 1970
0
Phil said:
"Eeyore"


** Obviously " modern sampling " = that done above 192 kHz.

So there are not many " modern " A to Ds in service - only the old hat
kind, according to the Graham Stevenson charlatan.

What ARE you rambling on about now ????

Modern A-Ds that provide so-called '192 kHz' actually sample the signal at
several MHz.

Graham
 
P

Phil Allison

Jan 1, 1970
0
"Eeysore"
What ARE you rambling on about now ????


** Your infuriating ambiguity in the use of everyday words.

What the **** does " modern " mean - when the Graham Stevenson charlatan
uses it ??

The word NORMALLY refers to what is current practice.

So translating your:

" Most modern sampling is way above 192kHz "

MUST cover nearly all the A to Ds in current service !!!!!!!!

Which is, of course, utter bollocks.


Modern A-Ds that provide so-called '192 kHz' actually sample the signal at
several MHz.


** Which, if true, is an entirely different claim from the first one.

Here "modern" = what is currently on offer from chip makers ???

One ** hell ** of a difference.




........ Phil
 
T

Tom Bruhns

Jan 1, 1970
0
By which you mean ultrasonic frequencies (close to the Nyquist limit imposed by the
over-sampling) though. NOT audio frequencies.

--Of course, it's not "Nyquist limited," but rather signals within
20kHz (or whatever your baseband upper frequency limit is) of the
analog sampling frequency and its harmonics...the signals that will
alias into your passband.
If used strictly for audio frequencies, a front end filter truly isn't required. The
rather gentle 6dB/octave LPF most application circuits suggest is simply to filter
any out of band signals. The recommended values for the ADC I've been using give you
a 1st order LPF @ 75kHz. That assuredly isn't going to give filter match issues in
the audio band. And, as I said, it could be left out if you're confident of the
bandwidth of the incoming signal ( i.e it doesn't contain anything in the megahertz
region.

Graham

Well, I'm glad to see that you apparently agree that protection
against aliases, however you do it, is a good thing. Clearly, if my
input has only "audio" frequencies in it, I don't need an alias
protection filter if I sample directly at 44.1kSa/s. You may have the
luxury of dealing with systems in which there is no explicit need for
an alias protection filter; when I was designing with delta-sigma
"audio" converters, I didn't have that luxury. You may even have the
luxury of not having to consider the effects of a first-order filter
with 75kHz cutoff at frequencies below 20kHz, but I also did not have
that luxury. Fifteen degrees phase shift (at 20k) is huge when you're
reporting phase to millidegrees. Even the 40dB attenuation of that
filter by 7.5MHz is inadequate when you're dealing with systems whose
input you guarantee to your customers to be protected against aliases
to 100dB. It's nice to live in a world where you don't have to worry
about little details like that, but in my world, I have to pay
attention to such details.

Cheers,
Tom
 
P

Phil Allison

Jan 1, 1970
0
"Tom Bruhns"
Well, I'm glad to see that you apparently agree that protection
against aliases, however you do it, is a good thing. Clearly, if my
input has only "audio" frequencies in it, I don't need an alias
protection filter if I sample directly at 44.1kSa/s. You may have the
luxury of dealing with systems in which there is no explicit need for
an alias protection filter; when I was designing with delta-sigma
"audio" converters, I didn't have that luxury. You may even have the
luxury of not having to consider the effects of a first-order filter
with 75kHz cutoff at frequencies below 20kHz, but I also did not have
that luxury. Fifteen degrees phase shift (at 20k) is huge when you're
reporting phase to millidegrees. Even the 40dB attenuation of that
filter by 7.5MHz is inadequate when you're dealing with systems whose
input you guarantee to your customers to be protected against aliases
to 100dB. It's nice to live in a world where you don't have to worry
about little details like that, but in my world, I have to pay
attention to such details.



** Bruns lives in the " world " of charlatans, con-artists, criminal
fraudsters and the biggest the scumbags on the planet.

He feels very comfortable in that world.

FYI

- it is the world of lunatic fringe audiophools and the pimps that live off
it.





......... Phil
 
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