With the latest version everything seems to be OK now. Cover art works even better than the coverflow feature of iTunes. I can play all the formats without dropout or speedproblem. I can play 24/96 hi-rez files through my USB interface. SQ is fantastic by all means, eclipsing the original mastertape sound. Or so it seems.

Peter, you can consider XX ready made now and realese it to the world marked. Prepare for great success and no need to spend any more nights on the forum with us geeks.

Best regards
Olaf Pirol

Yes, I think the 400 is a better candidate. Less (unnecessary instrument) features and better price. It also includes BNC input of external word clock. (see next point)

I have spend some hours scanning the world market for digital audio interface candidates. There is a very big choice from inexpensive soundcards costing peanuts, to pro equipment with lots of unnecessary (for my, that is) features costing thousands of euros. None of the very cheap ones mention any active jitter reduction circuits. Fireface 400 is actually one of the few who really discuss it's jitter reduction system in detail and who reprints jitter measurements. you say 800ps is not "that" good. But what is then?

To answer my own question, I found the Apogee Big Ben costing 1,500USD. It's actually a quite universal product, like a Swiss army knife, regarding my needs. Apogee does not mention any specific measurements, but they "claim" it's very good (low). But is it below 800ps?

-------------

Some people think the digital interface is solved with inexpensive soundcards. I think they are wrong. Let's say you install a soundcard in your PC, which outputs a AES/EBU 110ohm digital signal (squarewave signals up to 5volts). That is a VERY tough task for the powersupply. -Which happens to be the shared "Mickey Mouse" powersupply of your PC. -Which is not only very weak, but also very poluted with RF noise. I believed that such a digital signal will be very "jittery" when it arrives to the DAC. (Although I have not seen measurements of such a case, I have discussed the topic with a High End DAC designer and manufacturer).

So, probably, it's better to take the signal from the PC by Firewire into a separate box like the Fireface 400 with its own powersupply/onboard local power regulators. Then feed the Fireface 400 with good clean power (including ferrits on mains and other usual High-End tricks) and let it generate the spdif signal, going to the DAC. Eventually the user can choose between spdif RCA, spdif Toslink or AES/XLR depending on his DAC input.

I believe (hope!) this will be a "very good" solution at a not "so high" cost.

Eventually, an external clock (like Apogee, Esoteric or others) can be added to the Fireface 400's BNC input for external word clock. (I think the future will bring more choices of separate word clock to be purchased separately). Also future stand alone DAC's will have separate in/out BNC clock options.

To sum up: "Good" SQ can be made with inexpensive soundcards/interface, but I think that further margins can be gained when doing as discussed above.

I do not have any particular technical insight into this technology. But I like to get to "the bottom" of all aspects of my hobby. -So, any views on the topic are appreciated. Hopefully we can reach some absolute conclutions. At least I will report back later on after purchase and testing.

 


PS: Some say a 75ohm RCA coax will perform better if it's LONGER than a meter, because the additional length will attenuate the reflections inside the cable.

So we have learnt that USB sucks. Because it has a "speed limit" of 16/48. That is USB in its current version 2. Maybe the next version 3 will improve on this? Time will show. But then it will be another plug format, and probably not compatible with present USB equipment. A friend told me that Apogee DAC's DO accept 24/96 through USB, due to custom made in-house drivers. If there are any software pro's reading this topic, maybe they can shed some more light on the situation.

In the year 2008, if I wanna play hi-rez music, I have to go back the SPDIF interface using RCA plug format (made by the Radio Corporation of America in the early 1940s). Qiuite ironic, isn't it?

My PC (located in the basement) has Firewire 400 output. I will run a 5meter Firewire cable up to my stereo in the living room. Then I will connect it to a Fireface 800 box (purchase price about €1,200.-). The Fireface 800 box will be positioned in my equipment rack next to the Eximus DAC. Then I can run a 50cm short 75ohm coax RCA cable into the DAC. With this setup I can get my 24/96 through all the way to my DAC.

SOME NOTES:
I will experiment with high quality glassfiber optical Toslink between Fireface and Eximus. A galvanic separation can imporve the sound itself, due to reduced HF noice, mains ground loop, etc. If the Toslink interface works well, then the RCA connector will be donnated to the local museum.

€1,200.- for the Fireface 800 is a lot of money. The box has many more features which I never will use. (Build in A/D, D/A converters of medium quality, headphone outlet, digital volume, instrument inputs, etc). BUT: It also has a superb re-clocking circuit, which provides a State Of The Art low-jitter output of the SPDIF signal in both Toslink and 75ohm RCA format. Also it's not bad to have same configuration as Peter himself, when playing through XX.

There are cheaper "boxes" out there who can convert your PC signal into SPDIF for less money. The question is, if they have as low jitter performace as Fireface 800?

LynxAES16 (estimated price €900.-) is another option. It is also said to have high quality low jitter circuit. It delivers a SPDIF signal, but in the AES 110ohm XLR plug format. Unfortunately my Eximus doesn't accept this plugformat. Of course you can use a simple mechanical plugg adapter, but I'm afraid it would not be optimal.

Any comments, Gentlemen?

YYYYYEEEESSSS - WWWEEE AAARRREE REEEAAADDYYYY FFFOOORRR WWWWHHHAAATTT EVVVEEERRR YYYOOOUUU HHHAAAAAVVVVVEEE IIIINNNN MMMMIIINNNDDDD! Go for it!!!!!!!!!!!!!!!!!!!!!
 

No, just an error.

Question: IF I had a DAC who accepts more than 16bit, then, would the use of digital volume in XX introduce a SQ improvement? Or to ask in another way, is there any “secret” trick hidden in the XXs volume feature which betters the SQ?

-Or is the SQ improvement due to the omission of the preamp (shorter signal path etc)?

Please specify.

-------------------

Apart from this, I must state that the Eximus is build to a very high standard, and it is the best sounding DAC I have heard. The SQ of a DAC is based on several factors, and although Eximus doesn’t accept hi-rez through it’s USB input, it excels in other areas. For my ears, it sets a new standard for neutrality and resolution. It will remain in my system for a long time to come. And who knows, maybe there will be a future upgrade choice for it’s digital input?

BTW: Eximus offers 4x upsampling internally, before the D/A converting takes place. It’s a worthwhile SQ improvement on all quality recordings.

----------------

After a short initial listening test today, I feel that vU2 is another improvement compared to T-5. U2 is simply more transparent. I hear more recorded depth and more acoustic space surrounding each instrument. The bass is tighter. I can play several dB louder (always a good sign) without listening stress.

This is when playing through my external preamp as usual (XX volume = 0). My system needs between 20 and 30dB attenuation, depending on song material. (Some CD’s are recorded louder than others). So, no way to relay solely on the XX volume due to Eximus’ 16bit restriction. But I could eventually do -12dB digitally with XX and the rest -8 to -18dB on my active preamp. Are there any theoretical reasons (SQ) for me to try this out?

Dear Peter,

See DAC test below. Not much to brag about here...
My Eximus DAC is manufactured by APRIL, same company as maker of Stello DAC. Probably they all perform equally regarding digital input capabilities.

What does this means SQ-wise when using XXHighEnd? -Are we missing any sound quality [WHEN PLAYING REGULAR WAV FILES RIPPED FROM CDs - which is probably 99% what all of us are doing till now] because of this? Please advice. Several XX users have Eximus/Stello DAC's, so I believe this question is important.

What about SUPERIOR SOUND QUALITY, for a start.

I agree with Peter: In practical listening, we have no need for more than 16bit PLAYBACK performance, because teoretically 96dB S/N is more than enough to handle practically all excisting recordings.

On the other side, I think we can't have a high enough sampling frequency. Not because we can hear much above 20kHz (which is the limit for most red book DAC's) but because we can hear the sideeffect of the very steep lowpas filters at 22kHz. Pushing the sampling frequency up to 96k is certainly much better, soundwise.

I spoke with a renowned music producer who does all his recording and editing with the latest SOTA digital equipment (DSD/DXD systems at 32bit). He said that his classical recordings on SACDs has maximum obtainable dynamics [considering noisefloor of microphones etc], copied directly from his mastertapes without any compression. "Wow", you think, and start envying those with SACD players. BUT, then he continued and said: His CDs are only 3dB worse than his SACDs! And that these 3dB occurs on the SACD only for a fem milliseconds a handful of times during (let's say) a 5 minute song. So for 99,999% of the duration of the song, the dynamics of the CD equals the dynamics of the SACD. In other words 16bit is enough for PLAYBACK. (-Of course, he needs more than 16bits when mixing and editing. Also he recomends SACD for its multichannel capabilities etc, but that's another story).

He also confirmed that there is majore sound improvement when increasing the samplerate from 44k til 96k, but above 96k the SQ improvement is less noticable.

It should also be noted than very few, recordings are made so purist and dynamic as described above.

Terje is a seasoned high-end'er with very good system and listening experience. I hope he can contribute more here on the forum. Peter needs all the feedback he can get. XXHighEnd can be considered as a joint venture between him and all the native users. Remember we are still in "Beta-mode".

Also, what he told me, but forgot to tell you, was the following: His 24/96 file (downloaded free Mozart 2L.no sample, same as mine) plays perfectly in Foobar, but exactly the same track from the original CD (in 16/44 ofcourse) sounds better through XXHighEnd! 

In other words, XXHighEnd has ALLREADY reached 24/96 performance (relatively speaking) from playing 16/44. So thumbs up to you Peter! 

Keep up the good work Peter. I am sure you will solve the 24/96 related bugs in the near future. In the meantime we all are enjoying XX 16/44 quality second to none!

pedal

Hello again,

IN SEARCH OF HI-REZ - PART 2:
Continuing my efforts to play Hi-Rez, I have tested a 24/96 WAV file. No music. I tried all the various DAC settings in XX "settings". So whatever settings, I am not able to play WAV 24/96. The error code is "Can't play! - Wrong Stream Status (1)".

Then I asked VISTA to update my drivers, but they reply that my drivers was the last versions. See below.

So what can I do? -Is it possible to test out alternative USB drivers? Mayby it will work with USB drivers sourced from other companies?

---------------

I purchased the €5.000,- Eximus 24/96 USB CD/DAC in order to play hi-rez files.
Now I realize it doesn't work.
Will there be more developements in this field, Peter St.? 

After all there must be many users with hardware of same technical category as me.

Yes I did.

Hi Peter,

I am now using an Eximus CD player/DAC. It's USB input automatically detects and accepts signals between 16/44 and up to 24/96.

In your installation note, part 2, you state Rather important: Probably you are used to set your sound card to a certain amount of "buffer". E.g. for the Fireface that would be 48 (samples). Keep in mind that the below requires more of that buffer. E.g. with 96Khz/24 files (for me) that would be 96 (samples). Better higher this buffer at this stage, or *that* may give you crackles. Note that with 16 bits in either rate of upconverting (to e.g. 24 bits) my Fireface again allows for 48 samples. But start with higher anyway!

My PC-knowledge is limited, but as far as I understand, XXHighEnd is feeding the signal DIRECTLY to the USB com-port. So, there is no sound card buffer setting to change. Or what? Can you pls clarify this matter?


I have installed 09.U-0a and it plays music. The digital volume works. (I still use my active linestage for practial reasons BTW). I think it is a slight SQ improvement compared with XXHighEnd 0.9T-5. More fluid and "lighter" character, without loosing dynamic contrast or slam. (It reminds me about the positive SQ results of Q1 = -2, but without it's softening of the transients). I am using Q1 = 14, by the way.

However, I am not able to alter the XXHighEnd settings of "DAC is 16 bits   44.1 KHz". If altering to fex. 24/96 I get error code UNSUPPORTED FORMAT + FAIL (and no music). -I find this strange, considering that I have a 24/96 DAC. Whats happening? Can you pls clarify this matter?

Thanks in advance,

Dear PeterSt:

I appointed core sucessfully when installing M-1.
Then I downloaded S-1 a week ago (installed all the files from the new folder) and applied same settings and startet playing, with better SQ as a result. I also checked with Taskmanager and it was working similar as before (as with M-1), with a lot of activity in core one, and minimum activity in core two. See photo. It is taken when Engine 3 is playing a playlist in unattended mode, while I am keeping the PC busy with other tasks.

What do you think is the situation now? Should I download new version again, or is eveything OK?

Please advice.

EDIT: Just downloaded S-2 and all is OK!

Sorry for been unclear. Writing about technical matters during the night in foreign language, while drinking Cognac,  may introduce errors. It’s known as “midnight jitter”.

I just wanted to confirm your findings that jitter could be more audible at low frequencies than (much) higher up. I have done some listening tests with a software based tone generator. A 100Hz square wave sounds very different from a 100Hz sinus tone. But with frequencies above 10KHz the audible differences between square and sinus waves becomes less. (Of course you hear the difference, but less than compared with lower frequencies). Also, it must be added, that in music the highest levels are found at lower frequencies, among the instruments fundamental tones. In the treble it’s seldom max amplitude.

 -Yes, it was a reminder about the fact that if jitter is causing square waves, it will not pass through the DAC as a TRUE squarewave. (Most DAC’s have a capacitor on it’s output stage, to filter DC from entering the amplifiers. Including my DDDAC from Doede Douma). But there are a few DAC’s who are DC-coupled. (f.ex. AYRE). This can easily be read from its specifications, if the frequency response is stated from 0 Hz and upwards. So I think such DACs could be VERY sensitive to jitter at low frequencies, because they can transmit TRUE squarewave distortion into the amplifiers.

---------------

Something else: I have an analogy about audible jitter at low frequencies. With a jittery signal, the sound is similar to view something through an out-of-focus lens. When something is out of focus it tends to look bigger and bloated (“MORE”). But when you adjust the lens into sharp focus, then the objects you watch becomes slimmer and more precise (“LESS”). So when removing low frequency jitter with XXHighEnd S-1, the improved bass will be perceived as “LESS” (compared with the bloated jittery bass). This can be compensated with adjusting bass level or loudspeaker position. The result being a more correct bass performance, both in terms of accuracy and level.

pedal

I did, and I am really impressed with your discoveries which are covering new territory not mentioned in any other publication or magazine, as far as I know. In fact most “experts” still in 2007 says that the nature of audible jitter is “a mystery”:
Jitter causes more squaryness. This by itself would be audible, but never mind that.
A bass driver would have more difficulties with that than a midrange driver. Remember, a voltage jump with near infinit rise time is already hard to be followed by electronics, let alone a diaphragm. And of course, the more weight the diaphragm has, the more difficult it will be to follow the steep rise.

Digital, followed by a nos-DAC, will indeed feed that to the drivers. So, with a nos-DAC we're asking for trouble.
I believe you are right when you state that:
-Jitter causes squaryness.
-Squaryness interacts with the rest of the equipment, with different result between different systems and components.
-Squaryness is worse for bass than treble. (If you try to listen and compare square and sinus tones above 10kHz, they are in fact not much different).

BTW, “99%” of all DACs have a capacitor on its output, so they don’t pass on any DC components to the amplifiers. So true square waves cannot enter the amplifiers.

Regarding bass SQ of version S-1 I can testify that the performance is awesome. I have 2 pcs subwoofers with 7 pcs 12” drivers in parallel, in sealed 220cm linesource enclosures, performing flat to about 8Hz in-room. They are driven by very stable and powerful bipolar amplifier with 1000w each channel @ 2ohm. When playing Sheffield lab Drum Record (XRCD24) track 2 with Ron Tutt (Elvis’ drummer!) on very loud levels, his kick drum is simply scary. An experienced friend of my said he hadn’t heard such realistic kick drum even on live stage! I believe this could not have happened if not XXHighEnd had such superb performance and low jitter levels. With version S-1 I think jitter levels must be veeeery low for all frequencies.

Keep up the good work, PeterSt!

pedal