Thanks for the explaination again.  I think my confusion came from expecting an advantage from a more precise interpolation (out of band of course).

For the example I asked about (two adjacent samples differing by one LSB), 4 bits at 16x upsampling does not give the same shape you could get with more bits(before analog filtering).  I was thinking if you used more than 4 bits that the amplitude of any sampling artifacts should go down and the effective digital filter frequency should go up.  4 bits seems a rather arbitrary choice.  Why not 3 or 5 or 8?  You seem to already have proved that x16 sounds better than x8, so why not go for even more smoothing?

It all goes back to my inclination to preserve as many bits as possible for "something".  I'm not sure what the use would or should be and you make a pretty convincing argument that there is no reason to avoid use of bits for digital volume control.

I can see that your design choices for XX and the NOS1 dac are mutually supporting.  I respect that thinking.  It seems to me that your plan is based around attenuating the "loud" rather than boosting the "soft" recordings.

Would it be an equally valid choice to load the PCM1704 for linear full scale, implement lossless digital "gain" and use more bits for upsampling?  That would preserve the minimum parts count you have achieved.  I realize the gain question is tricky.  I guess it all depends on the usefulness of extra bits.

Dunno, just thinkin.
Greg   

Please bear with me, I am going somewhere with these posts and thanks for the reply Peter. 

Still not sure if I should use MSB and LSB for the ends of words that get truncated.  I ended up confusing myself with that.  From now on I'll use the bit numbers as shown on the datasheet.  MSB of the original word is 1 and the LSB is 24 (for 24 bit audio).  Sorry all for the sloppy way I worded that post.  I'll try to clear it up now .

One of the reasons I posted the schematic is I never saw a good explanation of the 1704 internals and what things there affect the sound in which ways.  I did quite a few searches and only found that Japanese application note and I can't read it either BTW   At least the schematic is clear.  It has been posted elsewhere on the net, but with little or no explanation. 

I think I understand the point you are making about dynamic range.  However, I'm still having trouble with valuing bit depth based on the noise floor.

I'm also still wondering about filtering benefits if using more than 4 bits for upsampling.  I'd love to see a spectrum analyser, but maybe it wouldn't resolve.  I suppose you have done listening tests and I cannot do that yet.  I also question if small transients and noise riding up near the high frequency limit, might get imperfectly filtered at 4 bits.  I mentioned in another post that I couldn't understand how adjacent words that only differ by one LSB (at 16bit) can be filtered to an arc with 4bit upsampling at x16.  That is still confusing to me, since the math doesn't work. 


I don't think we are seeing it the same way.

I'll try to explain my mental image of it again.  This is what I see when I look at the schematic:

At zero, the two switches for each bit are in opposite states.  Each controls a separate current for that bit.  One is normally open and the other is normally closed. 

Depending on the state of bit #1 MSB, closed switches can open or open switches can close, for either positive or negative waveforms (from zero).

Starting from silence, as the signal level builds, increasing large bit currents are used starting at bit #24(LSB) and working towards #2 (max current).  The state of the #2 bit switches never changes unless the signal exceeds -3db.  The state of the #3 bit switches never change unless the peak signal exceeds -9db, with bit #4 at -15db and so on.

As TI / Burr Brown says in the datasheet "The sign-magnitude architecture, which steps away from
zero with small steps in both directions, avoids any glitching
or large linearity errors"   (edit:  They call this Colinear in the PCM63 datasheet)

The high level bit switches don't do anything at all unless there is high level signal.  When you digitally attenuate, you are removing the possibility the high level signals can exist, so the high level bit switches sit idle.  The output end of the ladder is just sitting there as passive as can be.  Even the closed switches are a very high impedance current source and have no effect.  There is no switching or glitching, just analog attenuation.

This is very different than my lowly TDA1545 or other dac designs.

Each time the digital attenuation reaches another -6db boundary, another bit switch section can no longer be actuated since those word values cannot possibly exist.  So if I'm right about this .  We have the first boundary at -3db (since there is a possible +3db for peak extend).  Any digital attenuation of -3db makes it impossible for data to exist that could actuate the bit #2 switches, so that section of the ladder then becomes passive.  When digital attenuation of -9 is called for bit switch #3 is likewise idle and another section becomes passive.

Those now passive sections of the ladder each attenuate 6db (when properly terminated) The signal is already analog at this point inside the dac.  It is analog because it is complete and contains all the bit currents.  The bits have been summed (1 bit per rung) through the R2R ladder.  By digitally attenuating, you are progressively moving the active digital portion back up the ladder away from output end in 6db steps (1 ladder rung at a time).  I'm guessing that the rungs are composed of 500 and 1000 ohm resistances, since they claim 1000 ohm output impedance.  As you digitally attenuate, those resistances are still dividing current/voltage, even if there are no bit currents being added at that ladder rung. 

If you took that same resistor network and put it outside the DAC what would you call it?   

I now view this digital attenuation as an analog stepped pad inside the dac with 6db steps, in series with a 0db to -5.999999......db digital pad.  This is inspired by our debate

The "analog" steps should be slightly wrong (not 6db) for the first couple steps since the ladder is intentionally terminated out of spec(for passive I/V), if I understand correctly the NOS1.  This also explains the distortion graphs you gave since the current division in the R-2R should develop small errors as the output end is approached.  The last couple bit currents in the ladder would not get divided as accurately.

The distortion graphs are actually strong evidence that my explanation is correct, I hope, haha.

Maybe this view seems crazy and If I'm wrong I would certainly appreciate a better explanation. 

Best regards,
Greg

I should not be using the term MSB to talk about this.  Yeah, no attenuation on that bit as it always must exist in the word.  Inside the dac is different.  I'm not a digital guy, so I struggle for the words on this.  I should maybe say, the highest current bit instead?  I'll edit my posts so they make more sense.  Thanks for pointing that out.

Greg

Hi Coen,

You are right, in a regular ladder DAC that is how it would be.  However, look at the bit switches and think about the Colinear feature.  In this chip, the MSB in the word determines which bank of switches are used (positive or negative) and depending on which side of zero all bits are processed on either upper or lower section.  If the total value of the word is more than 50% of full scale then the highest current bit of that "bank" is switched on.  This way you never cross zero with MSB currents.  You can see the two banks in that every bit section contains two differential pairs of switches.  Very clever, these guys   At least that is how I think it works.  I have to be careful here in case someone who actually knows this stuff chimes in.

Greg

I envy your capability to post those analyser pics Peter.  I see what you mean about the distortion at FS.  A lot of things make more sense to me now.

I'll try to attach a file.......

Maybe other readers, even if you aren't technical will find this interesting. 

Here is the inside of the PCM1704 (see attached jpeg).  I thought it might be helpful to see where the business of producing the sound is happening.  The diagram shows only two bits due to space considerations.  I circled the MSB (no, should be highest current bit) section in red.  The other bits are represented by the single section circled in magenta and are typical for the remaining 22 bits.

You can see that the highest current bit switch (in red) is connected directly to the output. With a low Z load, nearly all of it's current will sink into the load.  The other bits are summed progressively through the rungs of the R2R resistor ladder and summed to the output.  The rungs of the ladder divide the currents from the lower bits to give the decreasingly less current for each lower bit. 

I think it's interesting that the NOS1 sounds better with some attenuation.  If you digitally attenuate, each -6db, abandons the use of another significant bit.  First the highest current bit is unused and then the adjacent sections as further attenuation in 6db steps.  For each additional -6db, the summing point for the bit currents is moved further down the ladder.

The lower bit switches that connect into the R2R ladder load are already working into a higher impedance than the highest current bit section by design.  They already swing voltage regardless of the load on the DAC, unlike the highest current bit.  (back to the HF filter arguments again, lol) 

Thanks BTW Mani for the report on your past volume control tests. 

I wonder if the slight advantage in dynamics (for digital attenuation) you heard was due to voltage swing issues in the MSB.  Having the load non-ideal (for completely understandable reasons) might make that bit current slightly out of calibration and perhaps affect adjacent bits as well.  It depends on operating conditions of those switch transistors and we can't easily know that.   

On another related topic:

You might see my point now why I tried to say that the resistors in part of the ladder are working in a similar way to an analog attenuator. 

When you are not using the higher bits, those switches are left in a static condition and that unswitched section of the R2R ladder is just a resistor acting as one leg of a voltage/current divider out to the load.  The bit currents have all been summed previously in earlier steps of the ladder and the signal flowing is the complete analog signal.  The resistive divider is attenuating this signal voltage.  That is the definition of an analog volume control.   All IMHO of course and I'll wait the response.

Back into theory heh.  With an ideal load (maybe not possible at this sound quality) then the advantage of digital volume should disappear, yes?

Sorry Peter, but I couldn't resist having one more go at this attenuator sparring (as you say)!  It was just too tempting.  

BTW, I do have 4 of the K spec 1704 chips, but I thought I'd try damaging the regular ones first   I would tell you guys the I/V design, except it might sound like cr*p and I'd prefer at least some of my failures be private!

Anyway, another can of worms is open .

(post edited to correct misuse of term MSB by me)

Cheers,
Greg


 

Lol, I deserved that.  I was trying to figure out how to say what I wanted to without making it very long.


Fair enough, I was referring to sine waves on a 50 ohm termination.  As far as flat, not really of course, but reasonably so.  There are always variations in the signal source (a DAC!).  The attenuated and unattenuated signals into the scope channels will track each other.   You could say (and be correct) that I'm just comparing two attenuators (digital in the scope preamp and analog in the 50 ohm line).  I think this ignores your major point about noise, hence spectrum analyser.

I really need to get my shop computer set up as a test instrument.  Those photo captures are pretty handy.  So, no photos to back up my possibly defeated argument......


You have me there again.  I can't remember, did I mention that I haven't used digital in my home until the past year.  I am trying to catch up now!  I did occasionally listen to pro gear and mixed a few shows on digital consoles/studio work or whatever.  But it's very hard to hear in those situations.  Too many variables.  Anyway, I did not know that lossless volume control is that recent for home playback.   


Amen!  I think this is the most significant statement you have made here.

I do believe that everything we hear can be explained by accurate theory.  I have yet to hear anything that didn't eventually make sense once I had all the information.  Can I measure everything I hear?  No, not very often.  I think also, that without theory, it's easy to get lost down the wrong road, following our ears down the easy path with immediate gratification only to miss out on much better sound by solving parallel, interrelated problems.  I know that seems non-specific theory , so for instance:  I've noticed many times people equalizing with component choices to overcome harmonic content causing what they perceive as a frequency response problem.  Don't even get me started on speaker design.  I would say that maybe "ears don't lie, but they also don't think for you".

 

Yes, the same here.  A TDA1545 and two amplifiers.  But I do not want to defend my current setup.  It's still just for testing ideas.



Yes, I notice the same issues on the 1545 with far less range!

See, I made another incorrect assumption.  I had expected your load (to the 1704) to be active (an emitter or a source) for lowest Z and the translation to voltage happen across an isolated resistor.  I totally agree that with passive I/V, there are no choices when you find the right value.  Noise lies in one direction and distortion in the other.  At least that is what my limited experience is showing.

Thank you for the longish post on bit depth, noise and dynamic range.  I haven't had time to digest it yet.

When you speak of full scale not being ideal on the 1704, are you referring to how it's loaded or to intrinsic properties of the chip itself?  I had meant to ask earlier if there is a sweet spot in the output level and does that give an advantage to digital attenuation in this case?

All for now,
Greg

   
   






 

Thanks for the responses.

I haven't had time to follow up lately as I wished (too much to do)!


No, not at all.  But, it is a complex topic and I got off on a side issue.

My original post was regarding proper use of gain structure on the analog side and the correct location (in my opinion) for attenuation and also using minimum gains and so on to control noise.  We really agree on this stuff so much Peter, that I feel bad arguing.  But, as a little aside, I'm finding that the very best amplifiers at my home are also the best for both horns and electrostats.  My favorites are Class A, unconditional stabile, ultra low distortion, etc.  Sometimes this creates a gain problem with certain sources.  So, while I agree that most people with high sensitivity speakers are using lower gain amplifiers (talking voltage here).  It doesn't always work out. 

I will summarize my complaints about advising use of digital volume as an absolute truth.

1)  Not enough evidence.  Yes, the NOS1 w/digital attenuation is heard to sound better than using an analog pad following.  But, that does not prove that the general idea is correct, only that it is correct with the NOS1.

2)  It's not hard to build an analog pad that far exceeds the bandwidth/distortion performance of any other audio equipment.  For instance, I have a pad on the output of a waveform synthesizer to feed the external sync on my scope.  It's flat to at least 50mhz when properly terminated.  This is not a special or rare device.  Resistors with low noise and excellent voltage coefficients are available.  I'm having a hard time believing that using a different range in the PCM1704 ladder and running the I/V at a different signal voltage has less sonic "character" than using a analog pad.  Also, if you don't like inserting a  voltage pad at the amplifier, why not have different fixed voltage outputs in the I/V.  You are already dropping the DAC current across a load resistance at some point to convert to voltage and using a different load results in a different output voltage (can be attenuation).  No need to discuss the  I/V design, that's not anyone's business by yours IMHO.  I just want to point out that there are spots to pad that introduce no extra components at all.     

3)  Also, I do have a problem with the loss of bit depth.  It seems like you are asking for two incompatible things to be true:  Either the "filtering" action of upsampling and arc prediction is needed or not.  If we use the bit depth for attenuation, then we are simultaneously changing the character of the sound by reducing the number of bits for upsampling.  Are we now saying that changing the number of bits for upsampling does not make an audible difference?  I was believing that one of the main advantages to XX/NOS1 was the method of filtering.


More later....happy listening!

Greg

Yep, actually a few companies tried this.  Beveridge is another and there were articles in Audio Amateur for do it yourself amps.  Quite a long time ago, I contacted EAR to ask about the ones you mention for the ESL57 mod and they no longer did them.  I wasn't clear if just the tweeters were connected to the tube anodes or if all panels were driven directly.  There would have to be a rather difficult network so it was probably just the tweeters. 



You might be surprised how well a dipole sub with a matching radiation pattern mates with a Quad.  I'd say it will play at least 6db louder, but Gradient used to advertise 10db and of course much deeper than a ESL63.  I've read that the larger Quads play deeper, but not much louder because the full range section is still getting the full spectrum voltage signal.   Curious if that is true.

Greg

Hi Paul,  thanks for the greeting.  Congrats on getting your system quiet.  I couldn't agree more that it's essential to getting good sound.
You can kinda see my point about filtering in the dac chip when looking at the rise time of the transients.  Are the peaks full scale?  It looks better than I expected.  Your I/V stage is certainly doing well!!


LOL, me either.  We may have to agree to disagree

Ultimately, all solutions either sound good or they don't, right?  Obviously, exclusive digital volume control is working for many people.  However, I'm having a hard time wrapping my mind around the idea that it is best in all cases.  I hate the idea of using up much bit depth for attenuation, since it's a function that I believe is done very well in the analog domain (I know that is the point, lol).

Various combinations of speakers and amplifiers can have such drastically different gains that I wonder how a fixed output DAC could correctly interface with them all.  I personally own speakers that are nearly 30db apart in apparent sensitivity (horns and electrostats).  At what amount of digital attenuation would you be better off just padding the analog signal?

When I finish my 24bit dac, I'll give it a try and see, maybe I'm wrong!   



Ha ha, not touching that one. 
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Yes, I agree that nothing is free.   Except in the case of my little transformer I/V experiment.  These Altec A256C amplifiers happen to have low Z input transformers as an integral part of the design (phase splitter and balanced feedback mixing).  It was just too easy to hook it all up.  The transformers are already there and I didn't add them in circuit.  Now you see why I said "lazy". 

The amps are old 1950's stuff.  I finally restored them to operation after 20 years on the shelf in my basement.  I felt like I had to use them a bit after such neglect.  I don't have pictures to post, but here's a link for a pair.  http://cafe995.daum.net/_c21_/bbs_search_read?grpid=fDQc&fldid=7foC&contentval=0009lzzzzzzzzzzzzzzzzzzzzzzzzz&nenc=y8McCQFicrtqJ1TisjO-_g00&fenc=&q=&nil_profile=cafetop&nil_menu=sch_updw

Using these amps is kinda like driving an ugly car that just keeps running.  I haven't had the heart to put them back into retirement.  Soon though!

Greg 

It's true that I don't know Peter's system.  I was using examples that I thought most readers could relate to.  My reference to 115db 1watt goes back to Altec(and other) horn drivers I've used.   I'm sure Peter has talked about his speakers, but I haven't read that yet.  As far as chip amps go, I heard a few and repaired a few.  I'm neutral on that topic.  They are what they are and seem to be extremely good for the effort and parts cost.  I have never tried running a chip amp directly into a high efficiency horn driver.  It's interesting that the GainClones are so quiet.

Nice plots by the way, thanks for sharing that stuff 

Maybe I should take this to another thread, because it's going far off topic.  I am interested in the Sauermann, honest!  Sorry for the hijack......

Read datasheets at night to cure insomnia......., I'm actually quite familiar with the internal diagram and function of the PCM1704 via the Japanese application notes.  It's true that the differential bit switches are operating more or less as current switch only, but the impedance of the rest of the R2R ladder is certainly not zero and voltage is swinging(making capacitive reactances matter).  Hence my statement about it being filtered.  Also, if you have succeeded in making a perfect current sink for the I/V conversion, then my hat is off to you Peter! 

I can't think how the I/V can be made a perfect current sink, so I'll assume a small voltage swing on the current output as well.  Therefore, the summed bit current output is limited in the rise time.  This is a filter, it's just so far out of band that it doesn't matter.  Sorry to be argumentative on that point.

It's all just reactances and no different than what happens when you insert an analog attenuator followed by an interconnect, amplifier, etc. 

If a simple voltage divider sounds poor compared to digital attenuation into the input of the GainClones, that says more to me about the input impedance and reactances of the chip amp than about the divider resistors.  Another thing to consider is the sonic effect of the output levels on the I/V.  Different levels might sound different and internally on the DAC chip. From what I can see of PCM1704 there may be sonic advantages to running at a sweet spot in the ladder with regards to major bit crossings etc.  I know the colinear arrangement fixes the zero crossing, but other major crossings exist. 

I don't have a clue what modern commercially available attenuators are out there.  I build my own stuff and also have some lab grade General Radio decades (T and H networks).  I've listened to TVC at a friends house and they just sound like transformers to me.  A bit of dynamics and air lost.  Not my choice.  They did seem to help filter the output a Sabre32 (Buffalo II) as a plus.  BTW, as far as I can tell, this old 16bit Phillips chip beats the Buffalo II when using XX arc prediction.

I'm using transformer I/V right now because I'm lazy and it's very safe for a 4 year old girl to play ABBA and dance   The gain is limited so that 0db cannot arc the Quad ESL63's.  Trying to teach her how to run XX.........

Regardless of my current system, I'm completely in favor of active device I/V.

Back to digital attenuation.......  How can we use the same bits for two different jobs?  If you attenuate 48db, are you not using all 8 extra bits below 16?  At the same time, don't you need those bits to interpolate 16x FS in the NOS1?

48db is a lot more attenuation that anyone needs.  But realistically, 24db is not.  If you use -24db attenuation (gives up 4 lowest bits (6db per bit) and also use 16x oversample, where are the discrete data points to create an arc?  If two words are 1 LSB apart, then having just 4 bits (16 levels) does not allow an arc (only linear), or am I missing something?  Isn't 16 integer points plotted over 16 samples a straight line?

Also, it seems practical experience shows that your digital attenuation sounds musical (many NOS1 users).  Maybe we don't need so many extra bits to make the arc prediction idea work?  I know that on my humble 16bit dac, arc prediction at 4FS seems to work surprisingly well.

More food for thought.  Has anyone analyzed how often two adjacent words in redbook pcm happen to be within a few LSB (least significant bits) of each other.  If that doesn't happen often, it would help explain how arc prediction can work well with less than ideal bit depth (as when using 24db of attenuation).

Yes, I am building a PCM1704 dac (as promised), just very busy right now!

Thanks for the entertaining conversation.  I hope I'm not irritating anyone and again sorry for the topic hijack (I'll move if it continues).  My mind tends to wander around with this stuff.  It's all so interconnected.

Greg


 

Ha,  thanks for the replies.  Maybe I was trolling a bit there :-)



Ah, I see where we are misunderstanding.  No, that is not what I am saying.  Not sound masking, that is not at all a good thing.  I'm just talking about electronic noise generated after the DAC.  If there is excess gain, then there is excess noise.  The only solution for such noise is to reduce the gain by removing (or improving) devices or through attenuation.  A volume control after the noise source reduces both signal and noise.  A volume control before the noise source just reduces signal and reduces the signal to noise ratio.   


So very painfully true.  Just when I think I finally understand the big picture, there is always more.   

One thing I'm not understanding is the dislike for analog attenuation.  I'll bet every user on this site uses analog attenuation.  Even if you have a Gainclone amplifier driven directly off a NOS1 you are using a great deal of analog attenuation in the chip feedback loop.

I could even go so far as to claim that "digital" volume is no different than an analog voltage divider.  It is simply selecting different resistors per word in the PCM1704.  After all, the DAC is mostly a bunch of switched nichrome resistors as a current divider.  Music signal currents still must flow through these resistors and all the effects of analog theory apply.  It's just that by careful design BB keeps any filtering to a higher frequency than we will ever worry about.

For the record, my "analog" volume control is a constant impedance network of switched nichrome resistors.     

Analog attenuation is the lowest distortion building block available to the designer.  I agree that using it stupidly, such as driving an interconnect cable through a pot sounds bad.  But, that is just an accidental analog filter.

Often, there is no good spot in the signal chain to place a volume control.  I will certainly agree that if using a chip amp, there is no benefit to adding anything in the chain.  Putting a volume on that amp front end makes no sense because it would effectively be at the same spot in the signal gain stages as the digital volume. 

However, I must still stick by my statement that an analog attenuator placed as close to the last device as possible has benefits over digital volume control that is upstream of noise generation.

Many amplifiers do generate internal noise that is easily audible if you are direct driving horn compression drivers in a multiamped system.

Cheers,
Greg 

Thanks for the link to the review Mani.  It sounds like the new Quad 2805's are a nice looking repackage of the classic (now) ESL63.  I have't heard the new ones yet, but hopefully one of these days :-)

I love these speakers, so I'm going to talk them up a bit!

It makes me sad to think Peter Walker is gone.  What a genius and a genuine nice guy.  I remember calling Quad on a couple occasions and having him answer the tech support call.  I wish I had understood enough back then to ask some intelligent questions! 

I'm on my third pair of Quad ESL63 based loudspeakers.  The first pair I purchased new were the original ESL63 with the aluminum louvers.  That pair I modified quite a bit and they were the best sounding (after mods).  I foolishly sold them thinking I could get my Altec/Jensen/W.E. horn speakers to that high level of midrange realism.  I was attracted by the amazing dynamics and easy amplifier choices of the big horn system.

Unfortunately, I could never get horns to cross over well enough to beat the Quads in the midrange.  I tried biamping horns with active crossovers.  I designed passive LCR networks for biamping and even a built delay derived linear phase network.  It was much much better, but they never quite had that believable midrange of the crossoverless Quads.   Finally years later, I'm back with the ESL63's again and dealing with the limitations.

I think the Quads are a great speaker if you are willing to spend some time getting the most compatible amplifiers.  I've used a lot of different designs both solid state and (mostly) tube.  Well regulated Class AB or Class A seems to be the key as long as the feedback phase stays proper under the electrostatic load. 

I read Mani's Sauermann amplifier review with interest, but I couldn't tell how the feedback works exactly.  I'm curious if the lowest frequency time constant is at the output stage.  That seems to be the key for hard capacitive loads.

The most interesting amps I've used so far are a pair of 5000 volt Acoustat servo amps that I modded with a different front end.  They are a balance bridge like the Sauermann with a 5000 volt power supply!  I connected the high voltage output directly to the Quad delay lines via high voltage wire which bypassed the Quad input transformer entirely.  One great thing about the ESL63 for direct drive is that you don't need to EQ since all dipole EQ is done by adding extra bass radiator area rather than boosting bass signal like most other electrostats.  I'm afraid to use these (possibly lethal) amps now because I have a young child in the house, but they were great fun.

For Quad users who miss high output, the right subs and crossover can make a big difference.  Once you remove the bass excursions, the Quads are much more competent for rock music or big symphonic stuff.   Right now I have the old Gradient SW63 system designed for the ESL63.  The subs sit under the speakers by design.  It's not the best thing probably and the stock crossover (mod or replace) is fairly awful, but it's a dipole subwoofer and that keeps the radiation pattern nicely matching the ESL63.

With the subs, the ESL63 goes from being a speaker that makes me slightly nervous when playing loud, to one I don't think twice about.

I insist that my "audiophile speakers" sound normal when I'm in the next room and Quads seem to do that very well.  Unlike a lot of other speakers, the response seems balanced wherever you are in the house.  No doubt, this is due to the extremely uniform radiation pattern. 

Perhaps the best thing about the Quads IMHO is what they do when given the best signal quality.  As you improve other components, they seem to grow in stature.  If you get a pair, don't sell them!!  Sell the other stuff :-)

Greg

Update:

I was able to confirm that the difficulties I had were the result of strange User Account Control issues and not the XX installation.

I tested if by running the same installation of XX from other user accounts in Win 8: 

A user account with administrator privileges (default) doesn't seem to work. 

A user account without administrator privileges can run XX if you run as administrator and type in the password.

The built-in administrator account works without hassles.
   
There is a small downside to using the built-in administrator account, since some of the "toys" in the new interface say they are not available with this account.  It's not a big deal if you just want to listen to music. 

I also use this computer for home theater video playback.  Netflix and movies off the hard disk work fine with the Administrator account.

The UAC problem might be a bug in this version of Win 8.  If it gets fixed in an update, I'll post a note.
 

The email didn't come through.  I think we had this problem before.  Can you send via your other account?