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Why NOS actually may make sense.
Why NOS actually may make sense.
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Old 6th November 2017, 07:45 AM   #11
oon_the_kid is offline oon_the_kid  Malaysia
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Originally Posted by Svitjod View Post
In order to do so, the interpolator – a FIR filter – looks backwards and “recreates” the sine wave. You can say this algorithm has an averaging effect. And this is probably the culprit.
In order to make sine waves look good at the oscilloscope ( very good amplitude related distortion) it assumes that this very wave doesn't change it's amplitude abruptly.
Hi,

In general I also prefer the NOS DAC, there is one factor that is quite often forgotten. Music is a bunch of random signal that is quite often non-repetitive. The problem with the reconstruction filter is that it can insert points that are not in between the two voltage levels, that is what it is suppose to do. That is why it is called a "reconstruction filter". The problem with that is what if the algorithm guesses wrongly, so it could inadvertently insert another signal that wasn't there in the original music. I don't think this area has actually been studied and there is no way of studying it. So while looking at a music signal it might decide that the waveform must have come from a 10KHz signal, when it is not, and creates one. that would have been one nasty "sss" that you would hear..

Oon
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Old 6th November 2017, 08:03 AM   #12
abraxalito is offline abraxalito  United Kingdom
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Quote:
Originally Posted by oon_the_kid View Post
In general I also prefer the NOS DAC, there is one factor that is quite often forgotten. Music is a bunch of random signal that is quite often non-repetitive.
Yep, I agree.

Quote:
The problem with the reconstruction filter is that it can insert points that are not in between the two voltage levels, that is what it is suppose to do. That is why it is called a "reconstruction filter". The problem with that is what if the algorithm guesses wrongly, so it could inadvertently insert another signal that wasn't there in the original music.
If the interpolated point calculated by the OS filter wasn't in the original music then that can only mean the aliasing requirement at the input has been violated. Otherwise the number of possible points it could be is heavily constrained by the input bandwidth restriction. Its not really true to say (as I've heard Mike Moffat say) that the interpolation filter 'guesses' at the missing value. If its a guess then its a highly 'educated' guess.

Of course differently designed filters will give slightly different values for the educated guesses. But then so will analog filters after the DAC, which as E_B has already pointed out, already exist in the downstream system and we can't avoid them.

@Marcel - yes indeed, an HF roll-off definitely 'smooths' the perceived sound and makes things sound more 'distant'.
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Old 6th November 2017, 08:24 AM   #13
oon_the_kid is offline oon_the_kid  Malaysia
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Quote:
Originally Posted by abraxalito View Post

Of course differently designed filters will give slightly different values for the educated guesses. But then so will analog filters after the DAC, which as E_B has already pointed out, already exist in the downstream system and we can't avoid them.

@Marcel - yes indeed, an HF roll-off definitely 'smooths' the perceived sound and makes things sound more 'distant'.
Well one major difference between analog filters and reconstruction filters (RC) is analog filters do add in a voltage that is higher than the two points. It is not physically possible. For example the 1st point is 5V, the second point is 2V, the in between point must be somewhere between 5V and 2V. Everything is real time.

A reconstruction filter on the other hand, if it guesses that this might have been from two points of a 10 KHz signal but the peak is in between, could insert in a 12V point to create a sine wave.

That is why NOS even with the best analog filters still produces lousy looking 16KHz signal. Because there is not enough points to create the signal properly. The peak is missed. So if you actually played a 16KHz through a NOS DAC, it will appear as a 16KHz signal amplitude modulated at 8KHz (22.1-16KHz). No amount of Analog can save that.

On my personal level, I prefer the mids from a NOS dac, but the highs suck. The OS DAC, the highs are nice but the mids suck...

I have read some articles that says a reconstruction filter with a more gradual slope can be the best compromise between the two...

Oon
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Old 6th November 2017, 09:32 AM   #14
abraxalito is offline abraxalito  United Kingdom
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Quote:
Originally Posted by oon_the_kid View Post
That is why NOS even with the best analog filters still produces lousy looking 16KHz signal. Because there is not enough points to create the signal properly. The peak is missed. So if you actually played a 16KHz through a NOS DAC, it will appear as a 16KHz signal amplitude modulated at 8KHz (22.1-16KHz). No amount of Analog can save that.
Turns out what you're saying isn't borne out in practice. I have a filter which is a 5th order LC (quasi elliptic, much steeper than I've seen elsewhere on another DAC) which does a pretty good job on a 16kHz sinewave - if you (or anyone else for that matter) want to see how it looks I'll take a pic or figure out how the screensave function works on my 'scope and post it up. Analog filters can indeed do the job as well as digital (oversampling) ones - in this case the image freq of 16kHz (which is 44k1-16k0 = 28.1kHz) gets about -30dB suppression.

Quote:
On my personal level, I prefer the mids from a NOS dac, but the highs suck.
I agree - but only on NOS without the steep anti-imaging filter. With the filter, the highs are nicely sweet.

Quote:
I have read some articles that says a reconstruction filter with a more gradual slope can be the best compromise between the two...
I've read that too - most recently in the paper by Bob Stuart. I'm staying in the camp of 'steep is best' for now though.
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Old 6th November 2017, 09:54 AM   #15
oon_the_kid is offline oon_the_kid  Malaysia
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Quote:
Originally Posted by abraxalito View Post

Turns out what you're saying isn't borne out in practice. I have a filter which is a 5th order LC (quasi elliptic, much steeper than I've seen elsewhere on another DAC) which does a pretty good job on a 16kHz sinewave - if you (or anyone else for that matter) want to see how it looks I'll take a pic or figure out how the screensave function works on my 'scope and post it up.
I don't think that is actually physically possible. Look at it from a physical sense. I actually have no clue about the filter you spoke of. But it would be really interesting to see the waveform before the filter and after the filter.

Let's say for the first point of sampling a 16KHz sinewave is at 0 at the beginning of the waveform. There are a total of 3 samples from the a single 16KHz cycle. The second point would be just before the signal transitions from positive to negative (would be a very low amplitude signal and the third point would be just before negative swinging to positive). The sampling has missed out all the high peaks of the signal. Therefore the NOS DAC would only produce the a low level signal. It would look just like a 22.05KHz being modulated by a 8KHz signal. At one of the later waveforms the signal would only actually capture the peak. Pretty much like an radio AM signal. I am not sure how an analog filter could add in all the missing peaks the signal that the NOS DAC sends out...
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Old 6th November 2017, 09:56 AM   #16
oon_the_kid is offline oon_the_kid  Malaysia
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Quote:
Originally Posted by oon_the_kid View Post
Well one major difference between analog filters and reconstruction filters (RC) is analog filters do add in a voltage that is higher than the two points.



Oon
Sorry Typo there. Should have read

Well one major difference between analog filters and reconstruction filters (RC) is analog filters can't add in a voltage that is higher than the two points.
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Old 6th November 2017, 11:13 AM   #17
abraxalito is offline abraxalito  United Kingdom
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Here's a quick snap from my scope of the filtered DAC playing out a full-scale 16kHz sinewave - there's still some modulation effect as you can see, but the signal level isn't reduced according to your hypothesis. In fact with NOS the filter is required to provide some gain as the ZOH incurs about 3.2dB attenuation at 20kHz - this filter incorporates that.

I can't do a before-after filter comparison as with this DAC the filter's prior to I/V meaning there's only current possible to 'scope, no voltage and I don't have a sensitive wideband current probe to employ. I can jury rig another NOS DAC to give a comparison shot though. Will look into that.
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Last edited by abraxalito; 6th November 2017 at 11:26 AM.
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Old 6th November 2017, 11:41 AM   #18
DF96 is offline DF96  England
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Quote:
Originally Posted by oon_the_kid
In general I also prefer the NOS DAC, there is one factor that is quite often forgotten. Music is a bunch of random signal that is quite often non-repetitive. The problem with the reconstruction filter is that it can insert points that are not in between the two voltage levels, that is what it is suppose to do. That is why it is called a "reconstruction filter". The problem with that is what if the algorithm guesses wrongly, so it could inadvertently insert another signal that wasn't there in the original music. I don't think this area has actually been studied and there is no way of studying it. So while looking at a music signal it might decide that the waveform must have come from a 10KHz signal, when it is not, and creates one. that would have been one nasty "sss" that you would hear.
No. No guessing takes place. The reconstruction filter does not know (cannot know) and does not care what the original signal (taken as a whole) was. It simply calculates the next output value from what it has already seen. This area is well-studied and can be found in any good textbook on digital signal processing.

Quote:
Originally Posted by abraxalito
If the interpolated point calculated by the OS filter wasn't in the original music then that can only mean the aliasing requirement at the input has been violated.
Yes.

Quote:
Originally Posted by oon_the_kid
Well one major difference between analog filters and reconstruction filters (RC) is analog filters do add in a voltage that is higher than the two points. It is not physically possible. For example the 1st point is 5V, the second point is 2V, the in between point must be somewhere between 5V and 2V. Everything is real time.
No. You may be thinking of a low order low pass filter, which is essentially just a smoother. Higher order filters (whether digital or analogue) can do what you believe to be impossible.

Quote:
That is why NOS even with the best analog filters still produces lousy looking 16KHz signal. Because there is not enough points to create the signal properly. The peak is missed. So if you actually played a 16KHz through a NOS DAC, it will appear as a 16KHz signal amplitude modulated at 8KHz (22.1-16KHz). No amount of Analog can save that.
Unfiltered or poorly filtered 16kHz from a NOS DAC will look like 16kHz plus a smaller amount of 28.1kHz, which may look rather like 22.05kHz double-sideband modulated at 6.05kHz. The reconstruction filter removes the 28.1kHz image to leave 16kHz. A good analogue filter can do this.

Quote:
I am not sure how an analog filter could add in all the missing peaks the signal that the NOS DAC sends out...
That much is clear. You need to do some more reading and some more thinking, then you can be sure. It took me quite a long time to get my head round digital audio; I kept thinking of snags (just like you are doing) but as I gained understanding I realised that there were answers to all of the apparent snags so digital audio actually works as it is supposed to work.

As a first step to enlightenment, stop thinking of a reconstruction filter as a smoother or interpolator. Consider instead an ideal brick-wall filter which can remove the 28.1kHz and leave the 16kHz. Real filters are not that good, but they are not a simple smoother.
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Old 6th November 2017, 01:31 PM   #19
Ken Newton is offline Ken Newton  United States
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Quote:
Originally Posted by oon_the_kid View Post
...A reconstruction filter on the other hand, if it guesses that this might have been from two points of a 10 KHz signal but the peak is in between, could insert in a 12V point to create a sine wave.

That is why NOS even with the best analog filters still produces lousy looking 16KHz signal. Because there is not enough points to create the signal properly.

Oon
Thinking about what a reconstruction filter does in the time-domain (as an oscilloscope would show) isn't particularly helpful to understanding the nature of an NOS or unfiltered signal. Rather, I suggest thinking about it in the frequency-domain (as an spectrum analyzer would show). It's important to recognize that an unfiltered NOS DAC signal FULLY contains the desired signal band information. This becomes clearly apparent when the unfiltered signal is viewed in the frequency-domain. The desired signal doesn't need to be guessed, or created, or even reconstructed, really. Instead, the issue with unfiltered DAC signals is that they also contain repeating copies of the desired signal band information, which are spectrally shifted up in frequency and centered at interger multiples of the sample rate.

The job of the reconstruction filter, whether it's digital or analog, is to remove the repeating image bands leaving only the desired signal band. Viewed this way, you can see that the reconstruction filter doesn't need to guess values in order to reconstruct the original analog signal, it just needs to strip away all of the undesired image bands. The image bands are what give the analog output signal an discrete looking appearance when viewed on an o-scope, even though the DAC's signal is entirely analog even without any reconstruction filtering being applied at all.

Removing the image bands via sharp filtering and the underlying smoothly appearing signal is revealed. Keep in mind that it was completely in there all along, sort of hiding if only viewed in the time-domain. I suggest thinking of the desired signal band more as being revealed by the reconstruction filter, rather than as being created by it.
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Old 6th November 2017, 07:10 PM   #20
MarcelvdG is offline MarcelvdG  Netherlands
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Quote:
Originally Posted by abraxalito View Post
Thanks, I was aware that JRS was a fan of 'leaky' filters which allow substantial aliasing. Which then violates the Nyquist criterion and that, according to Bruno Putzeys means the timing accuracy is compromised. So which to choose - Nyquist violation on a grand scale or 'time smearing' ?
I think their point is that with fairly high sample rates (96 kHz) and music of which the spectrum already rolls off naturally, you can get away with a relatively smooth filter and still keep the aliasing products well below the noise floor. Of course that means that the choice of anti-aliasing filter becomes dependent on the characteristics of the music; if you would want to record a church organ, you would need to switch to a steeper filter when the bats wake up.

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Just scanning the paper - why would he give an example of 8 30kHz 2nd order filters in sequence? The frequency response being shot at 20kHz from such would mean no self-respecting engineer would introduce that degree of HF attenuation in a line-level audio product. Completely unrealistic in my estimation.
I think the point here is that errors may accumulate over the entire signal chain. Many people would not object against a piece of equipment that rolls off by 1 dB at 20 kHz, but cascade many of them (microphone, microphone preamplifier, ADC, DAC, preamplifier, power amplifier, loudspeaker) and it becomes a different story.
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