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Why NOS actually may make sense.
Why NOS actually may make sense.
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Old 10th November 2017, 01:35 AM   #51
abraxalito is offline abraxalito  United Kingdom
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Quote:
Originally Posted by Extreme_Boky View Post
Anti-aliasing filter removes high-frequency content from an analogue signal before it hits the ADC, the band-pass/cut-off high frequency will depend on a sample rate we want to record at. Later on, we can use a gentle analogue filter after a DAC, to remove what’s left of aliases as the result of sampling of an analogue signal at the recording studio. Please correct this if I’m wrong.
The DAC's AIF (anti-imaging filter) can do nothing at all about aliases introduced at the recording stage. That's because they're in the audio band. The AIF is entirely concerned with removing images introduced by the DAC because those are all above the audio band.

Quote:
From what I measured, it seems the drop is around -3dB at 20kHz. The input signal was 20kHz wav file ripped off from a Burson CD (sampled at 44.1kHz in studio)
Yes the 'NOS droop' is about -3.2dB @ 20kHz.

Quote:
My finding is that if I send the 8X oversampled signal to NOS DAC, the frequency response gets ruler-flat far beyond 20kHz. This mitigates the -3dB drop at 20kHz side-effect of ZOH in NOS DAC. Subjectively, the sound had the same high-frequency extension as the other DAC that was processing PCM signal at the same frequency. NOS DAC sounded very nice and extremely spacious – closest to an analogue recording (played on a record player)
With 8X OS you'll get negligible 'NOS droop'. I have also found increased spaciousness moving from NOS to 2X OS.

Quote:
The 8X oversampled signal fed to NOS DAC sounded expansive, but unnatural.
Try a passive filter straight after the DAC chip itself. It won't need to be particularly complex given that the frequency domain space available for the transition band is huge, compared to NOS. A two inductor, three capacitor filter probably will do an excellent job.
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Old 10th November 2017, 10:58 AM   #52
DF96 is offline DF96  England
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Quote:
Originally Posted by Extreme_Boky
I was thinking of a digital (pre-equalisation) filter before DAC chip (it’s also called FIR filter I think…),
FIR (Finite Impulse Response) is a type of digital filter. Equalisation is a filter application.

Quote:
Anti-aliasing filter removes high-frequency content from an analogue signal before it hits the ADC, the band-pass/cut-off high frequency will depend on a sample rate we want to record at. Later on, we can use a gentle analogue filter after a DAC, to remove what’s left of aliases as the result of sampling of an analogue signal at the recording studio. Please correct this if I’m wrong.
First part right; second part wrong. Anti-alias filter before ADC to prevent aliasing. Reconstruction filter after DAC to remove images. Images are not aliases or the leftovers of aliases. The reconstruction filter cannot be too gentle.

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But, this 24/192 material was produced in a studio where they used reel-to-reel analogue tape as a source…
That would be much worse as a source of hi-fi than almost any competent digital audio chain.
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Old 11th November 2017, 07:59 PM   #53
Svitjod is offline Svitjod  Sweden
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Yes, a reasonably sharp analog filter could be some sort of good compromise between NOS and NONOS.

Look at my attached images. The Butterworth LP filter cuts at around 30khz.
At 2khz square wave output you can notice that there is no pre ringing as in a typical OS DAC.

At 20khz output, the signal resembles a sine wave pretty well.

Now to a second interesting thing, related to this:

If one implements the reconstruction filter as IIR, would that be less aggressive when it comes to "smearing" and pre ringing etc...

The filter below is very easy to implement as IIR. Could it be so that IIR filters are more natural to our ears than FIR filters? The transfer function is similar to a typical IIR. Actually, I think I must experiment a bit with this.
Back to Matlab...
Attached Images
File Type: png BW2000.png (27.3 KB, 92 views)
File Type: png BW20000.png (26.7 KB, 92 views)
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Old 11th November 2017, 08:37 PM   #54
Svitjod is offline Svitjod  Sweden
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One more interesting issue.

We all know the dreaded preringing of a pulse in a OS DAC. But the thing is that a single pulse that goes from zero to one and back to zero ( for one sample duration) is impossible to record using normal aliasing filters.
If you feed an ADC with a sharp spike with the duration of around one sample, it will be "smeared out" by the anti aliasing filter quite a bit and there will be no ( or very small)pre ringing in the following DAC.

But I don't really get how such a spike would look like ( in the digital domain) after it has been aliased. Would it exhibit some ringing similar to what the reconstruction filter of the DAC does?
Furthermore, if so are these ringings adding up or are they cancelling out each other. If the latter is true, then the whole NOS-thing will fall into the placebo bin.
But it's more likely that they adds up?
This should be fairly easy to experiment with.
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Old 11th November 2017, 09:07 PM   #55
MarcelvdG is offline MarcelvdG  Netherlands
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I think you should read Peter Craven's article on apodizing filters:

Peter G. Craven, "Antialias filters and system transient response at high sample rates", Journal of the Audio Engineering Society, vol. 52, no. 3, March 2004, pages 216...242

In a nutshell, when you put a filter with controlled pre-ringing at one and only one place in the signal chain, ensure that its stopband starts before the transition bands of all other filters in the chain, and use phase-linear filters with very little passband ripple everywhere else, then the pre-ringing of the entire chain is controlled by that single filter with controlled pre-ringing.

Last edited by MarcelvdG; 11th November 2017 at 09:26 PM.
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Old 11th November 2017, 09:40 PM   #56
MarcelvdG is offline MarcelvdG  Netherlands
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Quote:
Originally Posted by Svitjod View Post
Yes, a reasonably sharp analog filter could be some sort of good compromise between NOS and NONOS.

Look at my attached images. The Butterworth LP filter cuts at around 30khz.
At 2khz square wave output you can notice that there is no pre ringing as in a typical OS DAC.

At 20khz output, the signal resembles a sine wave pretty well.

Now to a second interesting thing, related to this:

If one implements the reconstruction filter as IIR, would that be less aggressive when it comes to "smearing" and pre ringing etc...

The filter below is very easy to implement as IIR. Could it be so that IIR filters are more natural to our ears than FIR filters? The transfer function is similar to a typical IIR. Actually, I think I must experiment a bit with this.
Back to Matlab...
When you make a minimum-phase filter, you get the smallest amount of pre-ringing for a given magnitude response, but especially for filters with a steep roll-off, you also get a far worse phase response than with a linear-phase FIR filter.

IIR filters and lumped continuous-time filters are minimum-phase filters as long as you don't include any all-pass sections. FIR filters can be minimum phase, linear phase, some compromise between the two (such as asymmetrical Wilkinson filters), whatever you want.
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Old 11th November 2017, 09:52 PM   #57
MarcelvdG is offline MarcelvdG  Netherlands
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Quote:
Originally Posted by Svitjod View Post
One more interesting issue.

We all know the dreaded preringing of a pulse in a OS DAC. But the thing is that a single pulse that goes from zero to one and back to zero ( for one sample duration) is impossible to record using normal aliasing filters.
If you feed an ADC with a sharp spike with the duration of around one sample, it will be "smeared out" by the anti aliasing filter quite a bit and there will be no ( or very small)pre ringing in the following DAC.

But I don't really get how such a spike would look like ( in the digital domain) after it has been aliased. Would it exhibit some ringing similar to what the reconstruction filter of the DAC does?
Furthermore, if so are these ringings adding up or are they cancelling out each other. If the latter is true, then the whole NOS-thing will fall into the placebo bin.
But it's more likely that they adds up?
This should be fairly easy to experiment with.
SAMPLED IMPULSE:
Theoretical:
When you have a Dirac impulse exactly at a sampling instant and a perfect anti-aliasing filter with a cut-off frequency of fs/2, then all other samples are taken at moments when the filter's impulse response passes through zero, so you get precisely one pulse on the digital side.

Practical:
When the impulse does not exactly occur at a sampling instant and/or the filter is not perfect and/or the filter's cut-off frequency is not exactly fs/2, then you indeed get sampled pre- and post-ringing on the digital side.

CASCADE OF LINEAR-PHASE FILTERS:
Theoretical:
A cascade of two ideal low-pass filters with equal cut-off frequencies is also an ideal filter, and therefore has exactly the same impulse response as a single filter.

Practical:
A cascade of two linear-phase FIR filters with similar cut-off frequencies has longer pre- and post-ringing than a single filter, but the part with the strongest ringing is very similar to what you get from a single filter.
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Old 11th November 2017, 09:53 PM   #58
Svitjod is offline Svitjod  Sweden
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Marcel, it's clear that you know a lot about the math behind digital processing.

And it's also clear that I didn't understand you. What I'm trying to find out is how that "smearing" effect relates to human hearing and what we perhaps can do to improve the subjective impressions.

If a traditional brick wall FIR filter smears things, what can we do to make some kind of compromise between OS and NOS?

So what I want is an open mind.
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Old 11th November 2017, 09:57 PM   #59
Svitjod is offline Svitjod  Sweden
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OK, you posted a second before my previous.

So, the aliasing filter and the reconstruction FIR filter gives ringings that adds up.

Good to know for a NOS-fan.
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Old 11th November 2017, 10:04 PM   #60
Svitjod is offline Svitjod  Sweden
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Abraxalito, that PDF you mentioned:

http://www.diyaudio.com/forums/attac...stribution-pdf

That was the most interesting document about psycho acoustics I have ever witnessed!

Take this quote that relates to the NOS issue:

"The impulse response of an ‘ideal’ Shannon-sampled
system is a ‘sinc’ function which has a fairly sharp central
pulse but also a pre-ring and a post-ring, which build up
and die away slowly.
Some may wonder how a time-domain analysis can tell
us anything different from a more conventional
frequency-domain analysis, since it is known that the
frequency-domain and time-domain descriptions of a
linear system are completely equivalent. If a human
cannot hear above say 18 kHz, how can a pre-ring at a
frequency of 20 kHz or 22 kHz be of any consequence?
One answer is to consider that a Fourier analyzer uses a
window that extends both forwards and backwards in
time. Thus although the two descriptions are equivalent
if one considers the global signal, the frequency-domain
description is very unhelpful in thinking about the
situation at a particular point in time when the future of
the signal is not known. A neuron has to make a decision
on whether or not to fire on the basis of what it sees now."
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