Upsampling to anything other than your DAC's internal conversion rate

[semente]

I wonder why, with all the positive feedback of audiophiles using software upsampling and filtering, there aren't more manufacturers producing NOS filterless DACs...

 

Does it somehow disturb the status quo?

 

R

[semente]

More expensive parts than using a commodity DAC chip, and puts you on your own as to how to implement the circuit (thus requiring engineering time and again more expensive). And to reach what fraction of an already small market?

I guess you're right.

Still, someone could take advantage of the tiny niche...

 

R

[semente]

I know it's not often people take the time to make the distinction. I use it this way:

 

native streaming vs DoP streaming (that's until the USB interface).

 

native DSD DAC or non-native DSD DAC (DAC-level).

 

 

Can we feed a non-native DSD DAC with a DSD stream?

DAC chip is PCM1794.

 

R

[semente]

No, won't work with that chip. That's a PCM DAC chip, which is confusing the matter even further (please don't call a PCM DAC Chip a non-native DSD DAC chip).

 

With this one you can: http://www.ti.com/lit/ds/symlink/dsd1793.pdf

 

So, if you want to listen to DSD with that chip, you will have to pre-convert the DSD to PCM before sending it to the DAC, for instance real-time with Audirvana+ or HQ Player.

Thanks.

So if I understand correctly there are DSD-able chips and those which do PCM only.

 

R

[semente]

Yggy upsamples , not to Dsd you are correct .

 

What I meant to say is the Yggy sounds better without software doing any of the upsampling and letting their filter do the work .

 

But if you feed the Yggy with upsampled or higher sampling material it'll still use its filter won't it?

 

R

[semente]

Great discussion.

 

But I do want to bring this back to the central point about DAC knowledge. How do we know what the magic sample rate is for each DAC that minimizes the DAC's internal proceeding?

 

I doubt DAC vendors are willing to disclose this, are they? For example, my Ayre Codex DAC supports up to DSD128 for DSD, but 32/384 for PCM.

 

Even if I accepted your point that something like HQPlayer is going to sound better, what would I set the upsampling to?

 

 

Sent from my iPhone using Computer Audiophile

 

The Teac UD-H01 that I used to own upsampled everything to 192k with its ASRC, it says so in the website literature.

But when I moved from BitPerfect to HQPlayer and fed the DAC with upsampled redbook the improvement in sound quality was quite noticeable, which means that the ASRC was rubbish.

 

I have recently ordered a custom designed and built NOS filterless DAC which can be fed PCM at 384k, making the most of HQPlayer's PCM processing capabilities.

The alternative would have been a DSD-able NOS DAC.

 

Which commercial DACs besides the Teac 501 can have filtering and possibly the ASRC turned off?

 

R

[semente]

The easy alternative is to find a DAC whose highest input rates aren't upsampled internally. In other words, say you have a PCM DAC that would upsample lower rate inputs to 352.8/384KHz, but allows you to feed it 352.8/384KHz and doesn't do any further resampling. Or, more typical these days, you have a DAC that accepts DSD128 and won't do any further resampling to input at that rate. So you just feed these DACs the max rate and it's the same as using an NOS DAC.

 

So it's easier to find a DSD-able commercial DAC that doesn't upsample?

 

R

[semente]

If I understand correctly, upsampling affects the audibility of filtering more than anything.

Current knowledge states that 2fs puts the filtering too close to the top end of the audible band and because filtering requires some slope there will be audible consequences.

 

My question is when we upsample, shouldn't the filtering become inaudible?

I thought that one would be filtering (the upsampled redbook) at 4fs or 8fs, etc.

 

R

[semente]

Thanks, that makes it clearer.

 

R

[semente]

There are two things involved, digital filtering and analog filtering. Digital filtering exists to help analog filtering, which is always mandatory, to do it's job better. Reason is that it is feasible to create accurate steep filters in digital domain and avoid extra problems of filters operating in analog domain (noise, distortion, etc). This way the analog filter corner frequency can be moved away from the audio band and the slope (filter order) can be relaxed.

 

If you start with RedBook, there's no way you can avoid having the filtering (or lack of) implications right at the edge of the audio band. You just need to pick something that is as optimal as it can be. I've spent enormous amount of time to come up with filter design methods that are as optimal as possible in both time and frequency domains simultaneously. That is tough challenge to push closer to the limits, because ultimately the two are mathematically bound by the 1/x relationship and the Nyquist fs/2 limit puts hard boundary for the bandwidth. I take objective-subjective approach to my work, so I want things to measure well and once they measure well they also need to sound good. Otherwise I'm not happy and wouldn't have peace of mind. But ultimately you need to make your own choice based on your particular hearing sensitivities and the material you listen. That's why there are all those options with linear- and minimum-phase responses and such. Different types of content emphasize different kind of filter properties, so it is good idea to select a filter that fits both your hearing (different people are particularly sensitive to different sonic properties) and your material.

 

The (anti-alias) filter properties become embedded to the source material when the sampling rate is reduced to 44.1k. With apodizing digital filters it is possible to replace/modify these filter properties that have been embedded to the source material. With suitable apodizing filter you can get "de-blur" effect.

 

For example rock recorded in studio puts demands on the transient response, so a minimum phase filter that doesn't have pre-ringing is usually good fit, since there's usually not much real acoustics involved anyway. While classical music recorded in real acoustics puts demands to the sound field/space, so a linear phase filter is usually good fit, since there are no strong transients or at least very few.

 

As the source sampling rate goes up to higher resolutions, the effect of filters also gradually diminish. So for hires content you have less to think about in that respect. It is much easier to do good D/A conversion for hires than it is for RedBook! For delta-sigma DACs the effect of modulator however is persistent at the same level regardless of the source sampling rate. And that is the another 50% of the DSP-side performance involved.

When you have time, could you elaborate a bit more on the effects of the modulator?

 

Cheers,

Ricardo

[semente]

On the run typing there, I did put the word "filter" in, should maybe used a different word, plugin perhaps, Mode 2 does add warmth to the sound, from the A+ manual.

 

Mode 1 that brings the highest transparency, and soundstage depth

Mode 2 that is more on the warm side

 

Adding even harmonics does create the warm sound, in an attempt to re-create this situation, experimented with Tube plugins, so far not settled on anything for a workable solution. Mode 2 is a good compromise. It's a preference.

 

I did read somewhere here at CA, that the Nadac has added distortion to make it sound audiophile...whereas the pro components HAPI and Horus don't.

 

In my experience warmth comes from exaggerating the upper bass range or a response tilting down from lows to highs, whilst even lower order harmonics add a sense of reverberation or space.

 

R

[semente]

What Miska is trying to do is to let you start with the cleanest possible digital signal.

How you soil it from there onwards is your responsibility.

 

 

I would start with a NOS DAC.

 

R

[semente]

Does your measurements really prove how you hear ?

 

Here's your chance today to show us the analogue eventuality—your preferred equipment setup and listening-room environment—from your digital mastery.

 

So far' date=' it's been a leap of faith that it translates well in widely applicable, real world-listening-situation terms.[/font']

 

Wilhelm,

 

I think you are missing the point of measurements.

They're not used to prove anything relating to hearing, only to determine the accuracy of a certain action, how much it interferes with the signal.

 

The goal of a high fidelity system is to tamper as little as possible with the recorded signal from the moment it's read until it's out in the room.

 

R

[semente]

∴ Miska is a-solution but not the-solution ? DSD may or may not work great if...

My DAC doesn't do DSD but I still find his PCM solution (upsampling and filtering) a better solution.

Moving the processing away from the DAC, which should only do what it's supposed to: Digital to Analogue Conversion.

 

R

[semente]

For a while this was a very interesting and informative thread...

 

R