Yes, Peter is right, within the last 6 months my PC for XXHE has evolved as follows:

1. Sony laptop - I was happy for many years with this
2. Atom-based PC (following the design on computeraudiophile.com called C.A.P.S.)
3. E8600-based PC in Zalman totally silent (no moving parts) case with Gigabyte mobo, fast memory and one SSD
4. i7-based PC (hyperthreading off) in Zalman totally silent (no moving parts) case with Gigabyte mobo, ultra-fast memory and two SSDs

This evolution has been soley down to using QAP in KS Special/Adaptive mode. Only 3. and 4. can really cope well with this - especially if you have any hi-res FLAC files which take just ages to pre-process on slower machines.

BUT...

I still use the Atom-based PC in my office, where I use it mainly to listen to background music. It works perfectly well in KS Adaptive mode with NO upsampling. With the hiFace, I have the buffer set to 1024. One thing I really love about this machine is that I can leave it on all day knowing full well that its consuming next to no power and will remain totally cool - its incredibly efficient.

EVEN MORE IMPORTANT...

In my office, I use an old MF x24K DAC. Even though this DAC is 24/96 cable, I prefer NOT to use DAP with it, as it sounds 'muffled' compared to straight 16/44.1. (I am convinced this is down to its use of delta-sigma chips.) So, if you have a delta-sigma DAC, I think an Atom-based PC could work well... but just don't use DAP or QAP.

My one single caveat to all this is: if you are interested in buying the NOS1 DAC in the future (and I think anyone who wants absolutely the best SQ possible should be!) then DON'T go for an Atom - you won't be able to use Double-OAP!!!

Mani.

Han,

How does Rainer connect the notebook to the Trinity DAC? I'm assuming he uses something like the Weiss AFI1, no? But the Trinity doesn't have a wordclock ouput, so its clock cannot be used as the master.


I'm sure the Trinity, with its [EDIT: eight 1704s per channel] is an amazing DAC... but I bet it'd be even better if it had a wordclock out.

I know I 'bang on' about wordclocks all the time... but only because I know from experience how important it is for SQ to have the master clock sitting right next to the DAC chips.

Mani.

Oooooh, what a room! On the one hand, I would love a room like this. On the other, I really like my music playing in a 'real' living environment.


On their site they say, "... the design engineers were able to eliminate the oversampling filter to increase sound quality still further...". Surely this is not possible with a delta-sigma chip. The 1704 then?

Mani.

Sorry Peter, I don't think I ever thanked you for the advice... I think I'll stick with my PCI AES-32 in that case - it has low enough latency to allow me to digitise 2-channel analogue at 24bit 176.4K/192K rates.

This will free up some funds to buy the two (I'm serious!) NOS1 DACs that I want from you. Does that put me absolutely top of the list?

Mani.

One of the biggest issues the hiFace has is that it doesn't have a wordclock input - it cannot be slaved to the DAC. You can optimise everything else as much as you want, but it's performance will always be compromised as a result.

M2Tech are addressing this with their 'EVO': http://www.computeraudiophile.com/content/M2Tech-Evo-and-DAC-details-and-photo

BUT... this requires a DAC with a wordclock output!

One thing I will never understand about the Berkeley Alpha DAC is the omission of a wordclock output. In the Pacific Microsonics Model One/Two, they went to such great lengths to get the wordclock right. Why on earth didn't they use this knowledge in the design of the Alpha?

Mani.

I attended the show specifically to listen to speaker systems. And in general, I was pretty disappointed with what I heard.

Most of the ‘conventional’ direct-radiating speaker systems just sounded ‘dead’, at best – what I call ‘cardboard’ sound. Many sounded positively objectionable – and I’m talking about well known high-end brands here.

I did have one interesting experience though. I was walking down a corridor and heard something coming from around the corner that I thought sounded really good. I walked speedily down the corridor and turned the corner to the sight of a live acoustic band playing snare drums, double bass and guitar! I have to say that absolutely none of the systems I heard came even close to this live sound. Could it be argued that this had much to do with bad rooms? Maybe. But it’s interesting that a live band can still sound ‘right’ from down a corridor and around a corner.

I have absolutely no desire to write negatively about anything that I heard. So here are a few systems that I had a positive experience with.

1948 Western Electric Theatre speakers

Driven by Silbatone A-845 mono amplifiers and C-100 preamplifier. 2-way system crossed at 800Hz.
Nice ‘alive’ sound. No real top or bottom to talk about though. Not sure how long I could listen to these – they seemed to have a bit of an edge to them.

Stax SRM-007tII/4070 electrostatic headphones

Really nice. Probably my favourite sound at the show.

MBL 101 X-treme

Very sweet top end – one of the best I’ve ever heard. However, things got progressively worse further down the spectrum. The mid-range seemed thin. There seemed to be a lack of upper bass. The lower bass, although substantial enough, was totally disjointed. But what a top-end!

In summary, although I didn’t hear anything at the show that approached what I consider the ‘real McCoy’, I’m still really glad I went. When I returned, I listened to some music on my speakers and realised that I’m not doing too badly. Yes, I have ‘cardboard’ sound too. But mine is in no way objectionable - just a bit ‘lifeless’... but listenable nevertheless.

Mani.

OK Peter, I've just checked this. This is what happens:

- Start XX (volume at -15 on start up, from previous session)
- Clear the playlist (10 tracks)
- Open Explorer
- Load an album (12 tracks, all 16/44.1)
- Change volume to -12
- Press Play
- Nothing happens... Here's a correction on what I said before - I DON'T think things are happening at snail's pace, things just aren't happening at all
- Press Stop
- Press Play again... all tracks get pre-processed, i.e. tracks get highlighted in blue, one at a time (in 1-2 seconds)
- XX goes into Unattended mode and plays music

This is totally consistent and repeatable with my setup.

HTH.

Mani.

I'm certainly no expert, but...

I have two SSDs, both 30GB Vertex Turbos, with the OS (W7 x64) on one and XX on the other. Furthermore, I use the 'Copy to XX-drive'. All music and artcover sits on a NAS drive (yes, I know Peter doesn't like these) in my basement, connected via a 1Gb LAN cable. I went for the fastest NAS I could find - a Synology DS1010+.

The important point is that I have absolutely no issues whatsoever! No glitches on playback, very little waiting time on loading (even with full 24/176.4 wav albums), and certainly no limit on how many tracks I can load (up to the limit imposed by the size of the SSD).

However, there is one strange occurance. A few weeks ago I connected my DAC directly to the power amp and started using the XX volume control, which BTW is vastly superior to my pre-amp (a top Pass Labs model). Now, whenever I change the volume and then press play, all loading slows down to a snail's pace. But all I have to do is to press stop, and press play again, and everything is back to normal.

On a final note, I have to admit that I have a wierd way of loading tracks. I can't set the 'Music Root' folder to the correct shared root folder in the NAS drive, so rather than using the Library, I simply use the 'Explorer' button. I tend to first clear the playlist and then load a whole album. Probably not very convenient for most people, but I don't mind. Maybe this way of loading changes things also...?

Mani.

EDIT: I know that SSDs are considered to have a shorter lifespan than spinning drives, especially if you're writing to them a lot. In which case, it's probably not a good idea to put XX there and copy files to it. However, I don't mind - if/when the drive fails, I'll just get another!

I switched to an i7 processor a few weeks ago (don't even ask what I had to do to get it working in the totally fanless and silent Zalman case), and I really appreciate the shorter pre-processing time. This is especially true playing hi-rez Flac downloads.

Another advantage (of a faster processor, not extra cores) is being able to reduce the buffer size in KS Adaptive mode. I'm now at 32 samples with Q1=1...

Slightly off topic: I managed to spend a day at the Munich High End show last week. I said a quick hi to Daniel Weiss and also had a quick chat with the guys from RME. Apparently, the PCIe version of the RME AES-32 card has a latency of 24 samples vs. 32 samples for the PCI version (which I have). I'm tempted to buy the PCIe version...

Mani.

Ooh, I'd love a pair of Celestion 6000 subs.

But not for my 2805s - I'm quite happy with the balance I'm getting, and also just love the speed of the bass and the general coherence of the sound... although I must admit it'd be nice to have a little more punch low down.

No, I'd love to mate the 6000s with the SL600s in my office

But you've got me thinking - high pass the 2805s around 100-150Hz, use a pair of 15inch drivers in sealed boxes down to ~50Hz and a pair of subs <50Hz. The only thing I'd be worried about would be speed. Surely no conventional driver could keep up with a 2805/2905s. What's been your experience with the 6000s?

Mani.

I tried this with my Wilson Benesch Chimeras for a few months, and indeed, much preferred them with their ports closed.

HOWEVER...

I was using them with active x-overs (internal passive x-overs removed) and could adjust a whole host of parameters (output, freq., order, Q, etc). If you close a port, you really need to be able to readjust a bunch of stuff. Worth trying anyway though.

I've never really understood ported designs - the ones I've heard and owned have generally sounded terrible. I guess it comes down to the manufacturers being able to quote low frequencies in the specs... This is such a shame.

Mani.

FWIW, it looks like the DAC section uses UltraAnalog chips, bearing a striking resemblance as they do to the older 20bit chips used in the Spectral SDR-2000 (compare the attachment to pics on Romy's site; http://goodsoundclub.com/Forums/ShowPost.aspx?postID=6566#6566). This isn't surprising seeing as Keith Johnson (one of the designers of the Model Two) happens to be the Director of Engineering at Spectral. I'm not sure about the ADC chips though...

Looking at the specs for the SDR-2000, it seems that the people at Spectral share your obsession speed...

Mani.

Wow, it's amazing how much your thinking has evolved in a year!


When you refer to 'measurement', what else are you measuring beyond just THD?

E.g., how important is slew rate in amps? Here's what Nelson Pass thinks (taken from the Aleph 4 manual):

"The slew rate of the amplifier is about 30 Volts/uS under load , which is about 30 times faster than the fastest signal you will ever see, and about 100 times faster than what you will be listening to."

And yet, if you look at the specs for something like the Spectral DMA-360, the slew rate is quoted as "600 volts/microsecond"!

Thoughts?

Mani.

I'm currently driving a pair of Quad 2805s (planars) with a Pass Labs Aleph 4 (single-ended, pure class A, 100wpc... and lots of heat!) and getting a really nice sound. In fact, the most 'coherent' sound I've ever achieved from speakers.

I think this is one (perhaps of many) recipes that actually works... thought not if you're into raves.

If only the Quads were more efficient, I'd be driving them with my Berning Siegfried.

Mani.

Brian,

Thanks - this is something that I'm planning to do this weekend with my two iMon-capable PCs, and it's good to know that it works... albeit only 95% of the time.

Mani.