oversample or NOS DAC ??

[rtate]

I am about to venture into the world of computer audio and I'm researching DAC's

What is the consensis on oversampling vs. non-oversampling and its effect on SQ?
Is there a big difference?

[Vadim]

If your sample rate remains 44.1 kHz, there is no way to reconstruct the signal to even 15 bits. I can't speak of sound quality, since it is a personal thing, but I can speak of signal quality, - and here you will come rather short without an over-sampling.

[Bernhard]

Originally posted by Vadim
If your sample rate remains 44.1 kHz, there is no way to reconstruct the signal to even 15 bits.

how do you come to that conclusion ?

[Vadim]

Originally posted by Bernhard



how do you come to that conclusion ?

Bernard,

Simple...really...well, - let's assume that you might be using a ladder type D/A like the Philips used to make or older, but still great Analog Devices AD1862 etc.

With 44.1 kHz sampling rate the image frequency will centre at 44.1 kHz. Considering that the usable data extends to about 18 kHz you will have approximately 8 kHz of 'window' where your anti-aliasing filter will have to work to suppress the image frequency.

Even accounting for the ‘zero-order-hold’ of the D/A, which is a low-pass filter in the shape of a sin(x)/x or sinc function, contributing a few dB of attenuation, still the unti-aliasing low-pass filter will have to be of a huge order, completely impractical, in fact not doable.

You see, 16-bits of signal resolution translate into about 96 dB of S/N. This is what you need in order to recover the 16-bits. Assuming that the sin(x)/x gives you about 10 dB at 26 kHz (44-18) you still need to generate another shall we say 80 dB or so in order to hit the target S/N.

It is not possible to do an 80 dB low-pass filter in a space of 8-10 kHz. So, the over sampling is a must here. It will shift the image frequencies to a higher point. Alternatively you will have you live with a reality of much lower signal resolution. My guess is that even 15 bits is not achievable without an over sampling, - more like 12-13 bits is doable. The rest will be covered by noise.

Regards,
Vadim

[Key]

Which algorithm should you use when upsampling and should you use an anti-alias filter when upsampling? I told someone on here yesterday not to use one when upsampling now after reading about NOS DACs and upsampling DACs my head freakin hurts.

So anyway lets say for this example I am using soundforge's batch converter. It gives you 3 algorithms - linear, lagrange, sin(x)/x. Which would be the best - if there is one - for upsampling? And what would the advantage be of using an antialias on an upsample?

[Bernhard]

Originally posted by Vadim

Even accounting for the �zero-order-hold� of the D/A, which is a low-pass filter in the shape of a sin(x)/x or sinc function, contributing a few dB of attenuation, still the anti-aliasing low-pass filter will have to be of a huge order, completely impractical, in fact not doable.

Vadim, I totally agree with you, non os needs a proper reconstruction filter.

This is what I use: 60dB/oct. & includes anti sin(x)/x, not trivial but doable...

[SoNic_real_one]

Actually, any frequency component above Fs/2 is indistinguishable from a lower-frequency component, called an alias, associated with one of the copies. So anything over 22.05 kHz...
That's why a NOS DAC cannot function proper with most of the DIY filters that are shown on this site (more like no filters at all). Distortion level is extremely high with no abrupt filters.
Why those aren't so good either? Abrupt filters have the habit to make weird phase shifts in the signal. Not good for quality music reproduction.
Going with a simpler filter that has a decent attenuation of image and slow roll-off (low distortion) will be cutting the high-end of the audio band.

So the only way that you can have the cake (maximum bandwidth) and eat it (low distortion) is to use an over-sampling digital signal to relax the requirements for the analog low-pass filter.

LE: Even a filter like the one above that looks good (and hard to make), has only a -30dB attenuation at 30kHz - well into the image field. Thats' not even close of the -96dB that are easily obtainable with a 8x OS and simpler filters.

[Bernhard]

Originally posted by SoNic_real_one

That's why a NOS DAC cannot function proper with most of the DIY filters that are shown on this site (more like no filters at all). Distortion level is extremely high with no abrupt filters.

Agreed, sounds strange.
Kind of artificial sound enhancement.
Some people might like it.

Originally posted by SoNic_real_one
Why those aren't so good either? Abrupt filters have the habit to make weird phase shifts in the signal. Not good for quality music reproduction.

I can't hear it.

Brickwall filters in the early stages of digital audio were mostly Murata hybrid filters with bad caps and bad opamps.
No wonder...

[SoNic_real_one]

It's not about the quality of the components, it's the laws of physics that dictates wild phase shifts in the case of abrupt filters. At the best, soundstage gets completely ruined.
I don't deny that those aliasing distortions some might find enjoyable and give fancy names and qualifications for them. I did listen to a "well made" NOS and it sounded plain bad to me. Am matter of gustinbus non est disputandum, I guess

[Bernhard]

Originally posted by SoNic_real_one
Even a filter like the one above that looks good (and hard to make), has only a -30dB attenuation at 30kHz - well into the image field.

IMHO the images are best removed by removing the sampling frequency and not by removing the images.

Originally posted by SoNic_real_one
Thats' not even close of the -96dB that are easily obtainable with a 8x OS and simpler filters.

But at what cost...