Pro-Ject Pre Box S2 Digital: MQA HW decoding at reasonable cost

[Miska]

34 minutes ago, left channel said:

 

As others said, it depends on how loud the volume is. Loud volume = loud pop. At my usual volume setting I can live with this temporarily, and anyway it doesn't happen every time, only when it changes between series (44.1 kHz vs. 48 kHz and their multiples). It's in the left channel when switching higher, and in the right channel when switching lower.
 

 

At least there seems to be a mute relay on the output, so it is likely fixable with a firmware update...

 

[Miska]

9 hours ago, MagnusH said:

I am surprised about the lack of reviews, but maybe the Pro-Ject hasn't sent out any to reviewers, or maybe it has to do with the later launch in NA. I am also looking forward to some comparisons with other DACs in the same price range.

 

IIRC, it has been reviewed in UK's HifiNews and also here in Finland local Hifimaailma. Probably some others too, otherwise it wouldn't have gotten EISA award.

 

[Miska]

2 hours ago, MagnusH said:

Sure, when it plays MQA it shows the symbol and the sample rate of the fully unfolded music. Remember to remove all DSP though before the DAC, or the MQA information is lost (this includes volume leveling). This has nothing to do with the S2 DAC though, just the way MQA works, it needs data to be bit-perfect.

 

Or just let for example Tidal application do the MQA decoding and let the DAC do proper filtering instead. The end result is much better that way...

 

[Miska]

On 10/21/2017 at 12:28 AM, MagnusH said:

Tidal (or any software) can't do full MQA unfolds, you need the DAC to do the last unfold (or so I have been told).

 

"Full unfold" means "upsampling with a crappy filter".

 

So Tidal and other software unfolding does as much as there's to do. Rest of the steps just increase distortion, but don't add any value and are better done with good digital filters instead.

 

[Miska]

8 hours ago, MagnusH said:

Its a lot more than just an up-sampling, and as far as I know it does not add any distortion. Still, I would imagine the first unfold is the one that gives the biggest increase in SQ.

 

At least all the information and measurements gathered so far has been that it is nothing more. MQA doesn't encode more information than 2x the base rate, in addition due to the filter they use there anyway wouldn't be any relevant information produced beyond that.

 

And yes it adds distortion due to poor filters.

 

[Miska]

3 hours ago, Jazz55 said:

I am facing the same problem as arcman is doing when trying to connect Bluesound node 2 with Pro-Ject Pre Box S2 and play MQA. The answer I get from the suppliers of Bluesound and Pro-Ject in Sweden is that Pro-Ject can only take in MQA signal through USB and not coaxial or optical. Blusound node 2 can only send MQA signal through coaxial or optical.

 

Just forget MQA...  

 

[Miska]

7 hours ago, left channel said:

 

Full 4x "unfolding" is only possible on Meridian-certified hardware. Not promoting, just saying.

 

 

You mean 2x unfolding followed by 2x upsampling with a poor filter...

[Miska]

One thing I want to note here, based on measurements, remember to check that "Distortion Compensate" menu option is "Enabled". Otherwise the performance will significantly suffer!

 

[Miska]

6 hours ago, mansr said:

Is this the THD compensation feature in the ESS DAC or something else?

 

I believe it is some feature of the ESS DAC chip. The difference in distortion is more than one order of magnitude!

[Miska]

7 hours ago, MagnusH said:

I had mine "Disabled", since if you select "Best" for Audio it turns to disabled. I thought it should be enabled for measurements and disabled for listening? To late to test sound quality now though, will experiment some tomorrow.

 

Don't select that unless you want to listen to a distortion generator!

 

"Fast rolloff" filter + "Distortion Compensate Enabled" gives pretty decent performance. Although the ESS digital filter has only 100 dB stop-band attenuation (typical equiripple).

[Miska]

IMO, only good solution is one that both instruments and ears like.

[Miska]

2 hours ago, GUTB said:

To clean up your digital playback chain you need more than a USB isolator. You need a dedicated audio PC tuned for playback, SSD filtered or removed from the audio circuit, use of a PC linear power supply. All audio components on a dedicated circuit or on a filter. Until this is done a PC based system won’t be able to compete with analog — or even a high end CD player.

 

It would be useful to have some objective data on this. There are also lot of DACs running from switched power supply, including the Pro-Ject Pre Box S2 Digital. Which, when running from my normal workstation, puts out better analog output performance than lot of high end analog gear.

 

The figures I've measured for it, running at DSD512:

SNR: 120 dB

THD: 0.00082%

IMD: 0.00009%

 

Source is my Xeon E5 workstation running Ubuntu Studio Linux.

[Miska]

17 minutes ago, GUTB said:

Switching mode power supplies are the enemy of audio. There can’t be a SMPS anywhere on your audio circuit. The absolute worst place you could plug in your audio equipment is in a power bar where all your non-audio stuff resides (PC, monitor, printer, lamps, etc). Same with linear PSU — you can’t plug them into a non-audio circuit and expect NOT to get hit with a SQ degradation. I recently tested that myself with my new TeraDak ATX LPSU. Basically, switching PSUs are your enemy, and they can’t be in or anywhere around your audio. If you’re considering a component that uses a switching supply, you have to discard it from consideration. If it uses a DC power input, you need to use a battery pack or a LPSU supply.

 

 

See those little coils? That’s the hallmark of a switching power supply. If you see those, look elsewhere. No serious audio component is going to use a SMPS.

 

Let's not try to turn world black-and-white. You think devices like Mytek Brooklyn DAC and bunch of others are not serious audio components? In many cases if you have a DAC that takes single voltage DC input, it likely has a DC-DC converter inside which is SMPS to generate bunch of different voltages like +-5 V, +-12 V and such...

 

So it is a bit too much to say A systematically leads to B, while truth is much more complex. I have quite many audio components using SMPS that have very good performance (much better than many "high-end" components using linear PSU).

 

Sure, I have two mains filters, one for devices with SMPS and another for amps and such that use only LPSU. But that doesn't mean there would be something wrong with either one. For a lot of computational power SMPS is needed and it certainly isn't a problem per se based on my measurements. I can use a NAA to isolate the computer and DAC when necessary. But now there are computer motherboards with USB ports specifically designed for audio, with low noise etc.

 

[Miska]

3 hours ago, Display said:

Yes - but only if "audio" means: analog component with bad power supply system...

 

I didn't say what you quoted!

[Miska]

2 hours ago, Kakel said:

Thanks for the download, left channel. The upgrade went well. No problems. I also solved the DSD problem. I forgot to patch the Mac OSX 10.13 system for the direct mode which apple stopped after El Capitan (10.10). When I did that, DSD was back in business again.:-)

 

Hmmh, I've had DSD working just fine all the time, now on High Sierra, without any patching... Don't give my Mac's bad ideas...

 

[Miska]

7 hours ago, Display said:

And are you sure Miska, that when playing DSD files, channels are not reverted?   (firmware 2.10) 

 

Yes, channel swap is certainly an issue, but other than that it works fine. I can swap the channels in the player, so it is possible to work around.

 

But patching macOS doesn't affect the channel swap in any case.

[Miska]

20 hours ago, left channel said:

The 2.11 firmware update resolves the DSD channel reversal, and a "pop" sound when the sample rate clock changes. PM me if you need the firmware file. You'll need Windows to install it though.

 

I already did...

 

I have to admit I have not noticed any pops though. Maybe it is because I mostly keep my DACs running at constant rate though...

 

But my point was that this firmware update has nothing to do with macOS compatibility or the other related issues.

 

[Miska]

3 hours ago, bhobba said:

Yes - I see your point - but its a bit more nuanced than that.  They don't even reveal what order of spline they use for the downsampling and matching up-sampling - I think it varies and is encoded in metadata.

 

I have heard both the correct unfolding from a  Direct Stream and Explorer 2, and whatever up-sampling Auralic came up with - I cant really say which I prefer.

 

However having your HQPlayer I suspect you might do a better job than the Auralic mob.

 

Yes, but that doesn't matter at all. Do not try to copy MQA, at least not the filters, you'll just degrade quality. Certainly you don't want to use spline for downsampling. And I don't think they actually use either. Their "splines" at upsampling (what they call rendering) phase remove most of their decimation filter. But there's no magic there.

 

If you want to save bandwidth/file size to equal amount or more than MQA while preserving better quality, use good filters and cut out the stuff above music content. This won't affect the music content's transient response as long as your filter transition band doesn't hit the music content. And you don't want that if you have DXD to begin with. You'll keep the transient response intact!

 

You can get more than "benefits" of MQA by not trying to do the same, but doing it different and better.

 

[Miska]

49 minutes ago, mansr said:

MQA "rendering" is a perfectly normal FIR filter (with bizarre coefficients). No splines involved.

 

Yes, I've seen it. Of course you can also turn IIR filter into FIR under certain conditions, but not vice versa. Not claiming anything about splines here, just continued to use the same terminology not to confuse more (that's the reason for double quotes).

 

I guess the spline part could theoretically be involved in creating those bizarre coeffs... Or someone just pulling those out of their sleeve, that is also possible (except for 14), given the very low attenuation and number of taps. For some reason 14 looks like normal equiripple FIR, 1 and 3 being sort of low order IIR looking. Others look like EQ'd version of 1 and 3.

 

[Miska]

28 minutes ago, mansr said:

Yes, 14 stands out as fairly normal-looking (has anyone seen it used?). All of them seem to have fairly deliberate frequency responses, so I doubt someone was just drawing shapes.

 

Btw, what factors did you use to draw the plots? Because the plots I get for the x8 rates look different...

 

Looking at the plot for 3 I feel like I have idea what they did.

 

[Miska]

5 hours ago, left channel said:

We've tested other anti-jitter products on it, and they did make a difference. Using the external power adapter (instead of just USB power) improves the sound even more, and some report that upgrading the power adapter may help as well.

 

Do you happen to have measurement figures for that?

 

[Miska]

10 hours ago, mansr said:

Factors? I used the freqz function in Matlab/Octave. This is filter 3 at x8:

Looks a lot like your plot but with higher resolution.

 

I mean the plots you shared earlier where there were all the filters collected in single picture. Those didn't look like these, those were for the 2x filters?

 

I used my own little Octave script that calculates various things about filters. But it doesn't interpolate the responses like that, so it is more like bare data. Same result anyway.

 

[Miska]

4 minutes ago, mansr said:

Interpolate? The frequency response exists for all frequencies. I don't see why you think calculating it at only a small number of points is more accurate.

 

Where did I say it is more accurate? 

 

My plot is just from plain 2x padded complex representation of the coeffs. For me it is easier if I can see actual data points in the plots.

 

But I don't know how relevant the plotting details are for this thread anyway. I already regret posting anything about the topic.

 

[Miska]

1 hour ago, bhobba said:

You of course can use an apodising filter that puts all the ringing after the pulse - that is much better since we are used to hearing ringing after pulses - but before - our ears aren't quite used to that - it never occurs in nature.

 

Apodizing filters don't need to be minimum phase, they can be also linear phase or something between the two... Or a filter like MQA rendering filters. The properties just make it replace side effects of the original "filter kernel" with the other one.

 

1 hour ago, bhobba said:

At the moment all I can say is I, and pretty much everyone I have listened with, prefer Tidal MQA to Tidal 44.1.  It seems one of those rare things most people I know agree on - like I say its usually all over the place.   But I have seen posts here where others have a different experience - so really its just the people I know - in reality like nearly every other thing in audio its all over the place.

 

What is MQA's advantage then?  Deciding will need much more material with actual content over 50k - and listening tests using that.  Only with such material can the claim of MQA that the recording with the better lime smear is always preferred be put to the test.   My opinion is it will be the same - all over the place - some will say - wow - others blah.

 

The problem I have is the technical approach that is trying to enforce certain types of filters on everyone. While also hampering possibilities to use things like digital room correction DSP that certainly have much bigger effect on sound quality. And as we know, it doesn't really save bandwidth either compared to properly encoded FLAC.

 

I don't mind so much Tidal using it, since that is only a single service and their application can decode it. And Tidal content is not preserved for long time. But I would mind if digital downloads I purchase would suddenly be available only as MQA. I don't want to risk that I would need to purchase same content again at later time. Plus I'm not too fancy paying MQA-tax on everything from content to DACs.

 

[Miska]

2 hours ago, bhobba said:

Ok lets take a look at polynomial interpolation - they are FIR:

https://www.dsprelated.com/freebooks/pasp/Lagrange_Interpolation.html

 

Now I haven't seen anything similar for spine interpolation, so I cant be sure in a mathematical sense if its FIR or not, however since the claim of MQA is to reduce time smear as much as possible I cant see how it can be anything but FIR.

 

I have two different spline interpolators in HQPlayer. IMO, the "polynomial2" is the better one. And IIRC, Wadia also used spline in their DACs. I have Lagrange too, but I don't think it fits these use cases at all, I've only used it to interpolate spectrogram plots.

 

I personally don't like such because of the poor stop-band attenuation. Time domain of course looks nice.

 

One can of course create discrete time sampled version (FIR) of those functions just like for IIR too. As long as function can be applied as convolution, it can converted to a FIR.