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Upsampling

And oversampling process may simply add its own imperfections. Thus isn't always any kind of cure-all. May simply be a waste of time and money and you'd be better spending more on, say, your speakers.

Correctly sampled, say, 48k material means there is NO INFORMATION left in the series of samples which tells you about any original components above 24kHz. Ideal reprocessing the 48k series up to a higher rate can't recover what isn't in the 48k steries. Not can it 'know' what filtering was employed during the process of creating that 48k series. You can decide to 'guess' and tell it to fiddle about on that basis. But you'd then need to 'know' something that you have to guess/deduce/etc. (Ahem!) This is what 'tone controls' are for. 8->

As far as I understand the goal of resampling to very high rates is to improve reconstruction at Nyquist, NOT to recover data which isn't there.
 
In theory you're wrong, but in practice they do seem to be black swans. However given that no ADC or other component is 'perfect' either, I don't personally loose too much sleep over that. Particularly given the imperfections of microphones in the first place, and speakers+rooms. Those, and how something gets 'mixed' by studio blokes seem to matter vastly more in most cases.

Indeed microphones & speakers+rooms are so imperfect, why do we need electronics with less than 1% THD is something I never understood, or anything better than mp3 or even shellac? Surely anything above that isn't audible...

Also, my theory being wrong it seems daft to oversample 16/44.1...
 
As far as I understand the goal of resampling to very high rates is to improve reconstruction at Nyquist, NOT to recover data which isn't there.

Yes. The point being that this may be a more practical way to do things like the filtering. It also can ease dealing with things like ADC nonlinearities, etc. Hence, for example, the first Philips chipset that oversampled x4. Helped the filtering and eased the effects of the limited DAC hardware in terms of processed sample sizes, monotonicity, etc.

So overampling can be very useful.
 
Indeed microphones & speakers+rooms are so imperfect, why do we need electronics with less than 1% THD is something I never understood, or anything better than mp3 or even shellac? Surely anything above that isn't audible...

Also, my theory being wrong it seems daft to oversample 16/44.1...

The point is that theory and reality aren't usually the same. :) Of course, in theory they are, but reality may not have read that book.... 8-]
 
I'm finding it almost impossible to follow what's been said here, although I'm interested.

I need a simple answer, should I oversample? I'm using Audirvana with an NOS DAC (Holo Spring DAC). If yes to what level?

I don't think I can hear a difference with previous tinkering, but perhaps my guests can so I just want to set it up 'right' and forget about it.
 
I'm finding it almost impossible to follow what's been said here, although I'm interested.

I need a simple answer, should I oversample? I'm using Audirvana with an NOS DAC (Holo Spring DAC). If yes to what level?

I don't think I can hear a difference with previous tinkering, but perhaps my guests can so I just want to set it up 'right' and forget about it.

If you don't hear a difference but want to know, then simply ask your 'guests' if they do. And if so, which they prefer, or if they think it is too small a change to care about. Ideally, not telling them which version is oversampled and which isn't.
 
I'm finding it almost impossible to follow what's been said here, although I'm interested.

I need a simple answer, should I oversample? I'm using Audirvana with an NOS DAC (Holo Spring DAC). If yes to what level?

I don't think I can hear a difference with previous tinkering, but perhaps my guests can so I just want to set it up 'right' and forget about it.

You should oversample/filter, unless you prefer the potentially euphonic effects of not using a reconstruction filter (ultimately it's your call).

The Spring performs best when fed DSD256, or you could try 20-bit 705.6/768kHz PCM.
 
I don't think I can hear a difference with previous tinkering, but perhaps my guests can so I just want to set it up 'right' and forget about it.
I think it’s worth trying two settings: a factor of two and to the max. Not much faffing required, and see if your guests hear any difference.

There are long threads elsewhere proclaiming the benefit for the Holo but don’t tell your guests that.
 
Audial S4 (TDA 1541)

I am guessing that your Audial S4 (TDA 1541) sounds sublime… without any upsampling.

My Musical Fidelity AMS CD player / Dac, automatically upsamples everything to 24/192 and it is great.
But also is the Opera Consonance DAC16 (TDA1543) using the MF as CD transport.
My experiences over the years led me to some disinterest about upsampling… until now, maybe one day :)
 
One possible problem with studio-generated rock/pop is intersample overs. Generating an oversampled/upsampled version can show these up. In some cases it can then reconstruct them if you downshifted the input into a series of samples with more bits per sample. Otherwise you rely on the DACs reconstruction filter.

That may have been a problem in the past but these days most digital audio workstations and certainly all professional mastering engineers have easy access to true peak (dBTP) meters.
 
In theory you're wrong, but in practice they do seem to be black swans. However given that no ADC or other component is 'perfect' either, I don't personally loose too much sleep over that. Particularly given the imperfections of microphones in the first place, and speakers+rooms. Those, and how something gets 'mixed' by studio blokes seem to matter vastly more in most cases.
Indeed microphones & speakers+rooms are so imperfect, why do we need electronics with less than 1% THD is something I never understood, or anything better than mp3 or even shellac? Surely anything above that isn't audible...

Also, my theory being wrong it seems daft to oversample 16/44.1...
No matter what genre of music you listen to, every step of the recording and mixing process was subjected to qualitative decisions based on what the engineers felt best capture the sound in a way that they like: the studio/hall, the microphones, the preamplifiers, the mic compressors, the EQ, the mixing board, and even, yes, the conversion ( for the longest time many people felt that Appogee made the best sounding, "must musical" ADC with its soft clipping facility). Add to that whatever further processing is done at the mixing stage. Only at the mastering stage do they aim for the cleanest processing and conversion so as not to impart flavour (in the subtle imperfect hardware "mojo" sense).

So, if one's objective is to reproduce the performance accurately, it's a fool's errand. If one's objective is to reproduce the recorded signal as accurately as possible, then upsampling to remove even the remotest concerns about the DAC filtering or about digital aliasing in any pre-DAC DSP, is surely a good idea. If one's objective is produce an imagined music performance in the studio/venue of their listening room, then try to find whatever combination of DSP, sampling, and filter settings floats your boat.
 
However given that no ADC or other component is 'perfect' either, I don't personally loose too much sleep over that. Particularly given the imperfections of microphones in the first place, and speakers+rooms. Those, and how something gets 'mixed' by studio blokes seem to matter vastly more in most cases.

This ^^

Rarely heard night and day differences with DACs. Not to many tried, Naim DAC, Hugo TT, Hugo TT2/mscaler and Holo Audio Spring.

They are so easy to slot in and out I think we are all tempted to tweak, whereas speakers are big heavy beasts which are much more difficult to move on.


.sjb
 
Some resampling algorithms are just more precise/accurate (lower distortion products) than others. See https://src.infinitewave.ca/ with 1kHz measurement.

E.g. Signalist 2.9.1, Weiss Saracon and others seem to be impeccable, some others are very good - some others not.

Given that even computer software can vary in quality, it's plausible computer upsampling - if done optimally - could simply be more precise/accurate than some DAC implementations.

Whether this aspect can be audible is another matter but it's measurable at least.
 
Nostromo

My take after many years of futzing about, reading and listening:

Yes, theory definitely says you should upsample. Correctly done (and there's no telling if a given filter is good enough) it will improve the reconstruction of the original signal over not upsampling the same file. The brutal truth is that the samples contain ALL the information that was in the original signal WITHIN THE PASSBAND of the adc filter used to digitize the signal but without passing those samples through a reconstruction filter recovering that information is not possible because of the artifacts of the analogue to digital filter process and the noise that is an outcome of the process.

I have heard some lovely sounding NOS dacs (and some bloody awful oversampling ones) but they are basically doing it wrong (according to theory).

For what its worth, I am up to my ears in the boggy upsampling camp (see the Dac200 thread for the gory details) and I find significantly more music in pushing the process to the nth degree. Just regular hardware upsampling (Limited resources, even with a Chord Dave) or perfunctory software (pretty much everything less than HQPLayer 4 or PGGB) is pretty much indistringuishable from not upsampling.

A Holo Sping or May (not Denafrips, many are convinced there is some processing going on) if a well engineered product - feeding it good upsampled bits should produce better results than basic bits.
 
If you look at the datasheets of some DACs, you can see that at higher sampling rates, the output spurii and non-linearity do get worse. This means that upsampling does have a downside.

That may be due to poor implementation, poor algorithms or bcause the processing in the DAC is generating rubbish.
Measurements from the DAC in question, Holo Spring 3:

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These are from a TEAC UD-501 set to:

44.1/16 NOS,
44.1/16 SHARP,
44.1/16 SLOW,
44.1/16 upsampled to 384/32 into NOS,
44.1/16 upsampled to 384/32 into NOS,
44.1/16 upconverted to DSD128 into NOS

(unfortunately the page formatting has gone mad, but it's easy to get the picture)

 


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