A how to for a PC XO.

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diyAudio Member
Joined 2004
StigErik said:
I'd just like to add here that I've been using Shin's PC x-over solution for approx 2 years now. Its the greatest thing I've ever tried! Period.

My hardware is a IBM T61 laptop with RME Fireface 800 and RME ADI8-QS AD/DA in addition (it sounds better). Digital input signal comes from a Squeezebox, and from an analog turntable on two mic inputs of the Fireface (RIAA is just an EQ plugin here of course). I use all 8 channels on the DAC, because I have a 4-way speaker system.

On the sofware side there is of course Console, Waves LinEq for x-over, Waves paragraphic for speaker EQ and Voxengo sample delay for time alignment.

A million thanks from me to Shin for his ideas.

Thanks Stig, congrats on the beautiful setup you have. From reading your website it comes across that you don't do things by halves and the PCXO route takes some commitment but glad you persevered and wrought the benefits.

BTW As Arthur points out there's a relatively new kid in town that you might be interested in - Acourate. I find it be very good and sounds even better than the original method.

Let me just go back and dig up some old posts I made illustrating the functionality and philosophy of Acourate... just a minute.

OK found it:

Note: The following two graphs are not driver responses but the visualisations of the mathematical filters and anything can be made to look perfect with maths! So don't pay much heed here. Just know that if the drivers were 'perfect' then they would exactly follow these targets shown. The good news is we know the crossovers themselves are the known target to reach both in the time and amplitude domain. However the bad news is that once a driver is stuck on the end of the filters we must remember that everything is ruined. This is where correction can using Acourate can help out.

The filter transfer functions are as follows and these, when summed together, create a flat response(the black line). The filters in this example are Linkwitz Riley 4th order:

An externally hosted image should be here but it was not working when we last tested it.


This next one is showing time domain behaviour, in the form of an impulse response, of the above. You can clearly see the individual filters are ringing, which is typical of any crossover and increasingly severe with steeper filtering, but when summed together the pulse response in black can be seen to show perfect behaviour due to the individual filters ringing cancelling each other out when summed as one.

An externally hosted image should be here but it was not working when we last tested it.



So what happens to the above when you put your driver in?

Well take a look at this horrible mess showing a typical midrange and its intended passband:

An externally hosted image should be here but it was not working when we last tested it.


There's three elements here:

Red is the filter tranfer function like the ones we demonstrated above, the ideal and what were aiming for in order for the crossover to work accurately and as close to correct as possible.

Brown is the actual driver response with the filter in place and is gain shifted for a better view and not to clash with other the other elements in the graph. Ideally this should perfectly follow the red line but the truth is it couldn't be further from it!

Black(bold) is the minimum phase correction filter needed to bring the brown(driver response) in line with the red(filter transfer function)

Why the need to correct?

Well if you saw that the crossover filters alone are perfect so we must try to make the drivers follow suit and match them to the filter. Its a balancing act here because over-correction ie. correcting every single defect however small or large will lead to a very strange sound that really only works in one very very small sweetspot. An excellent solution is to smooth the correction filter so as to provide a gentler effect that works on the more severe errors but largely ignores the smaller ones that if corrected wholesale could do more harm than good.

Below is a comparison of without correction in brown and with correction in green. Note that the measurements are unsmoothed so what you see is the unblemished truth at the measurement position. You can clearly see the big improvement in overall shape and you can also see that excess correction hasn't taken place because we aren't seeing over correction of the response which brings its own problems. In the end what we have is a driver response that closely follows the filter and should you apply 1/3 octave smoothing to remove the aliasing present in the measurement you'd see that it almost perfectly follows the filter. Do this for every driver, combine the responses and it brings us one step closer to accurate.

An externally hosted image should be here but it was not working when we last tested it.


And here is the before and after correction for step response and min phase behaviour with the example shown directly above:

Step Response

Red = Filter Step Response
Green = Corrected Driver Step Response(unsmoothed)
Brown = Uncorrected Driver Step Response(unsmoothed)

An externally hosted image should be here but it was not working when we last tested it.


Phase

Green = Filter Phase
Blue = Corrected Driver Phase(unsmoothed)
Black = Uncorrected Driver Phase(unsmoothed)

An externally hosted image should be here but it was not working when we last tested it.





And finally an example of complete 3.5way loudspeaker with the above driver correction elements AND room correction combined:

Frequency Response showing +0.5dB / -1dB (1/24th octave smoothing)
An externally hosted image should be here but it was not working when we last tested it.


Overall view individual driver responses and overall response (1/24th octave smoothing). Note that measurement is only valid from 200hz up as bass measurement was omitted.
An externally hosted image should be here but it was not working when we last tested it.


Step response at the listening position:
An externally hosted image should be here but it was not working when we last tested it.


CSD showing decay behaviour at the listening position within the 1st second after signal excitation.

Without bass traps and digital driver/room correction:
An externally hosted image should be here but it was not working when we last tested it.


With bass traps and digital driver/room correction:
An externally hosted image should be here but it was not working when we last tested it.

Hope you'll agree that its an incredibly powerful tool and moves things even further on from the Wave/Console method you currently use.

BTW there's a trail version of Acourate available but it can't generate useful filters because its only a demo to show the functionality of the software. To really hear what Acourate can do you have to buy the full version which is 340Euros.

But if you like, why don't you email me and we can work together to get a test example of some correction filter generated. Then you can hear for yourself. Its quite a simple process and all you'd need to provide is some swept sine measurements. We can talk about the details via email should you wish to do so. I think you'll be surprised at the sound though.
 
diyAudio Member
Joined 2004
m0tion said:
Shin:

How did you eventually decide to tackle the "FIR filter delay and video" problem? Do you use IIR filters when you watch video? I seem to remember something about that. Could you elaborate on your video setup?

For video I have ffdshow and use that to set an output delay matched to the latency of the filters bringing the audio/video in sync. Now the most qualitatively demanding tasks such as music and video are run through FIR filtering.

For real time applications such as gaming the delays are no good and here I use IIR. Compared to FIR these aren't quite as coherent or precise but the difference is small and not easily noticed unless your directly switching back and forth between the two. Audio quality in games isn't really made to tax half decent loudspeakers so it doesn't really matter whether the filters are IIR or FIR. It all sounds pretty much the same.

The two different modes(FIR&IIR) are on scripts set up and controlled by AutoIT. From here I've set the iMON remote software to issue a hotkey command which AutoIT picks up and initiates. The end result is I can switch seamlessly between modes using the convenience of a single key press via the remote.

Screen grab of ffdshow with delay shown:

An externally hosted image should be here but it was not working when we last tested it.
 
Not particularly wanting to 'butt in', but I have recently heard the deqx PDC 3.0 and can let anyone who cares know that it is significantly cleaner, more spacious and open than the 2.6, and I do think it represents an significant upgrade over the 2.6.

Another important point perhaps (not to me particularly) was that we couldn't pick a difference between the digital input or the analog input...sound wise that is, there was maybe a half db level difference which allowed us to pick it, but apart from that it was almost impossible to p[ick them. So any analog lovers out there the pdc 3 becomes a must over the pdc 2.6 for example.

They acknowledge someone for being the inspiration behind the upgrades..


>The unit is based on the PDC-2.6/P Rev-6 PCB that many of you have,
> where there are changes to main PSU, grounding, EMI filtering, op-amps and
> capacitors. The results have been stunning, and frankly I had not believed
> would have been possible were it not for the initial work of Steve Nugent of
> Empirical Audio, who was the first to achieve these type of results with even
> earlier Revisions of the PDC main board (where earthing issues were
> particularly tricky to improve on) so thank you Steve!



I was lucky to have Alan Langford from deqx come out and have a play with it in my system, and interestingly he also showed me a 'new way' of measuring the speakers, completely different from the procedure in the manual (which of course was the method I had always used).

To say the new results were revelatory would be an understatement, my system (which was always pretty darn good) went to a new plane, one that I have never heard before in any other system. Period.

He also showed me a 'pioneering' method of room and bass correction, evidently mine was only about the fifth system they'd tried it on..again completely different from the manual (I think Alan has brought a new way of looking at things in the short time he's been there and so is pioneering new applications etc), again the bass I had previously ( a pair of PHL 18's) which never failed to impress people ha ha took on a new dimension of speed, attack and accuracy.

All in all, after having the deqx for over a year now, and always reckoning it was great stuff for the system (as anyone reading this thread should know eg pc based similar stuff), it's only now that I have realised what the deqx is capable of.

I personally would be very interested in learning this pc based stuff, but unfortunately my computing skills are simply not up to it...so I can fully relate to what Shin was talking about earlier about ease of use etc and pros and cons of pc vs deqx or tact.

I would imagine that many users of deqx would have more smarts than me and so perhaps had better measurements in their correction file than I did so giving them a closer sound to the ideal, but then again I wonder. If you have done the 'old by the book' measurements, well I'd be very interested in finding out if you get any improvement from the new method of measuring and room correction (and hope you get the same dramatic improvements as I did!!)

Not to leave anyone in mystery, but I think the plan is at some stage that deqx will do some sort of video presentation on the new methods they have developed which will be accessible from the website, if I find out I'll let people know if they're interested then they can have a look for themselves.

So, pdc 3.0 is a worthwhile improvement, the new methods (to me anyway, your measurements may be perfect) of measurement also wrought vast improvements, and Alans expertise in driver x-over points and slopes (which he also changed) has led to my system being absolutely top notch, and that is not said in a bragging manner but simply a statement of fact.

BUT, has anyone who is conversant with setting up this stuff on a pc ever used it to give the iterative process on Ambiophonics a go?? and if so what were the results for you? I don't need the pc stuff right now fro stereo as I use the deqx, but I'm very interested in trying ambiophonics with it's crosstalk cancelling algorithms...which throws me straight back into my computer illiteracy.

If however people have tried it and think it's worth someone else investigating Ambiophonics, well it at least gives me an incentive to try and work this stuff out.
 
diyAudio Member
Joined 2004
Thanks for the info Terry. Its good to hear the DEQX aren't resting on their laurels, the PDC2.6 hardware is essentially over 5 year old(my how time flies) and whilst the most important part, the core software has seen updates I felt it was high time for a new part to debut.

I still really like the DEQX, its very easy to use, almost fool proof infact. And the latency, stability and tidy integration into a small 1U enclosure were real draws for those wanting a set and forget unit. However I couldn't escape the fact that it was audibly inferior to filters, created by programs such as Acourate, that were then played back through high quality converters and clocking. If I had to attribute these sonic inferiorities to specifics then it would be down to a couple of things. First and most obvious was it had virtually no real DRC, well it did have a few bands of PEQ intended to be used in the bass but this is just basic amplitude correction rather than the phase/amplitude seen in more sophisticated correction. Another was the basic driver linearisation which wasn't of the same power as the stuff I've shown a few post above this one. Lastly the hardware and converters were quite old designs, although they were admittedly very good when the unit was first released all those years ago.

A point that isn't related to any performance issues but something I none the less was aware of - I found the stereo 3-way only configuration slightly restrictive for my own uses. I'd be out of luck with my current setup since they're a 3.5way design. I could buy two DEQX's and it would work but that's silly money.

The new unit addresses at least one of these issues - the hardware. I'm not so sure about improvements to the software such as amplitude/phase DRC or more rigorous driver correction. I'd imagine given the DSP horsepower of the DEQX these things are probably a stretch for the Dual Sharc's so it could be impossible. Or maybe their aim to keep latency down to a very low sub 10ms is another factor. More info on this would help clarify my guess work, fingers crossed they addressed not only the hardware but improved the software too.

As you can probably tell, I'm quite firmly entrenched back into the PCXO camp despite flitting to and fro over the past few years. Even so, given the slightest chance, I'd jump at hearing the new model in action.

You've also got my interest piqued with the alternative measurement scheme you mentioned Alan was using. Please let us know more if you find out anything new.
 
Shin, I've sent an e-mail if you could check please, I'd be very appreciative.




ShinOBIWAN said:

I still really like the DEQX, its very easy to use, almost fool proof infact. And the latency, stability and tidy integration into a small 1U enclosure were real draws for those wanting a set and forget unit. However I couldn't escape the fact that it was audibly inferior to filters, created by programs such as Acourate, that were then played back through high quality converters and clocking. If I had to attribute these sonic inferiorities to specifics then it would be down to a couple of things. First and most obvious was it had virtually no real DRC, well it did have a few bands of PEQ intended to be used in the bass but this is just basic amplitude correction rather than the phase/amplitude seen in more sophisticated correction. Another was the basic driver linearisation which wasn't of the same power as the stuff I've shown a few post above this one. Lastly the hardware and converters were quite old designs, although they were admittedly very good when the unit was first released all those years ago.

I understand about the PEQ for room correction, I always felt it did a good job tho and would much rather have it than not!!ha ha However your next comments made my ears prick up ( phase and group delay correction in the room correction step). It most definitely WAS something along these lines that Alan did as part of the new procedure..evidently there was the power and facility within the existing software (or whatever, don't want anyone to think I know what I'm talking about) that simply required this new procedure to realise it. The 'discovery' came out of a completely disrelated research line that is still a bit hush hush so won't go into it. However to their surprise it worked in the existing deqx setup.

The measurement manages to take into account the phase and group delay effects of the room on the bass driver (or so I'm told, remember at all times I'm an idiot). Then we still have the parametric bands after that for any remaining anamolies.

All I can say (whatever the truth or otherwise of what Alan said to me) my excellent bass got faster and cleaner with even more slam than before. So something is different. Funnily enough, the old 'walk around the room and hear the wildly fluctuating bass response' has lessened remarkably, as if it corrects throughout the room now, not just the sweetspot.

ShinOBIWAN said:

A point that isn't related to any performance issues but something I none the less was aware of - I found the stereo 3-way only configuration slightly restrictive for my own uses. I'd be out of luck with my current setup since they're a 3.5way design. I could buy two DEQX's and it would work but that's silly money.

yeah that would be silly money, and I too can't see any way to use the deqx in a 3.5 way. Another 'strange' thing about the new bass/room correction methodology..I have never, no matter how much eq and power I threw at it, get my 18's below 29 hz in the room - flat to there then dropped like a stone. So I built subs using peerless xls 10's (someone you may know did a build with them on AVS ha ha) and used a behringer dcx to add extra para etc for the subs. In other words, to get full range I had to resort to extra subs, processors and amps, ie it was a four way.

However, with this new measurement to my surprise the PHL's now go to twenty as we always wanted...????!!I no longer need or use the subs, the PHL's now go down loud and low, don't ask me how or why that worked but what the heck, less boxes in the room now.

So with the correct choice of drivers a three way can be perfectly adequate (that was NOT to imply you had a poor choice of drivers!! IIRC it only became a 3.5 during the build...it was originally 'only' a three way??)

ShinOBIWAN said:

The new unit addresses at least one of these issues - the hardware. I'm not so sure about improvements to the software such as amplitude/phase DRC or more rigorous driver correction. I'd imagine given the DSP horsepower of the DEQX these things are probably a stretch for the Dual Sharc's so it could be impossible. Or maybe their aim to keep latency down to a very low sub 10ms is another factor. More info on this would help clarify my guess work, fingers crossed they addressed not only the hardware but improved the software too.

As you can probably tell, I'm quite firmly entrenched back into the PCXO camp despite flitting to and fro over the past few years. Even so, given the slightest chance, I'd jump at hearing the new model in action.

You've also got my interest piqued with the alternative measurement scheme you mentioned Alan was using. Please let us know more if you find out anything new.

Anyway, maybe that kinda answered your last couple of questions.

I don't want to go into these new measurements stuff, I'd probably do more damage than good simply because I don't know enough to respond to queries, but hopefully all will be revealed sooner rather than later.


I still hope someone will report experiences with Ambiophonics. I didn't mean to hijack the thread.
 
diyAudio Member
Joined 2004
Terry kindly provided me with details from DEQX about the changes and their upgrade policy from PDC2.6 to PDC3.0. I'm posting it here just in case its of use to others:

Dear DEQX PDC-2.6/P owners,

As you may have heard, we are now shipping the new HDP3 (PDC-3) processor/preamp, and will have information on the web site imminently.

The HDP3 is the culmination of a lot of experimenting regarding sound quality through the PDC's ADC, op-amps and DACs, as well as power supply and related issues. The unit is based on the PDC-2.6/P Rev-6 PCB that many of you have, where there are changes to main PSU, grounding, EMI filtering, op-amps and capacitors. The results have been stunning, and frankly I had not believed would have been possible were it not for the initial work of Steve Nugent of Empirical Audio, who was the first to achieve these type of results with even earlier Revisions of the PDC main board (where earthing issues were particularly tricky to improve on) so thank you Steve!

While the changes we have made are not as entirely full on as Steve Nugent's so that costs could be kept within affordable limits, I believe we've achieved outstanding quantum improvements, including where the switching power supply remains! Another factor that has helped the success of this process have been the recent introduction of a new generation Op-amp. However, the improvements we have settled on use a variety of high-end op-amps depending on their function. For example, dual 300V/microsecond slew rate op-amps with high power driving capability are used to drive the AKM Analogue to Digital converter and two each are used for each of the six DACs for current to voltage conversion. All other signal paths use new generation very low-noise, low-distortion op-amps. Another area where changes have been made is with regard quantity and quality of bypass capacitors, while any in-line capacitors have been changed to new gen, very low ESR (and musical) capacitors.

As you may realize there is a significant amount of set-up and labor time required when modifying surface mount boards to the degree outlined below, which is then burned-in and re-tested from scratch. For example, Steve Nugent's Empirical Audio mods cost US$3,000, which some have called bargain! However, we feel these improvements are so significant that we would like too offer them to as many of our users as possible. The only way we have been able to do this at an affordable price was to negotiate a one-off bulk arrangement with one of our assembly contractors that specialize in detailed hand work, as distinct from the automated assembly that we use for all new units. Logistically, we would need your unit for about two to four weeks, in mid April, by which time we will have ordered components and metalwork to fulfil this one-off batch order.

Upgrade options:

Option 1): PDC-2.6/P "HD limited edition" available for all serial # PDC-2.6 and PDC-2.6P:

This is the most affordable upgrade because it can be installed in your present PDC chassis, although we need to supply a new chassis cover. If you have a PDC-2.6 or PDC-2.6P with a serial number above 0411001, then these changes are essentially the same as now incorporated in the new HDP3 processor, less the analogue power supply. If your serial number is prior to that, the changes can still be made, although the benefits of latter PCB layout changes will not be as effective. The changes are as follows:
1. New black textured powder coat chassis cover with improved ventilation. This is required because the new analogue section draws more power than the existing PDC 2.6 and PDC-2.6P.
2. Mechanical modifications to both analogue 12-volt regulators to allow higher power output.
3. Multiple Low ESR FM series bypass caps added to the main +/-12V rails.
4. Low ESR FM series bypass caps added to the main +/-15V rails from switching supply.
5. Multiple Low ESR FM series signal caps replace existing caps in signal path.
6. Extreme high speed, high drive current op-amps added to drive AKM Analogue to Digital Converters (ADC)
7. Extreme high speed, high drive current op-amps used for six channels of DACs current-to-voltage I/V conversions
8. Very low noise, low distortion op-amps used in input stages to ADC driver op-amps
9. Very low-noise, low distortion op-amps used in all DAC filters and output buffer stages
10. Miscellaneous modifications
11. Additional 12-month warranty for serial number above and including 041101.
Six-month warranty for units below S/N: 041101 (which will be older than 3 years).
12. Unit notated as Special HD limited edition.
13. New return packaging.
14. Return freight included

Our price for this upgrade would normally be A$1,550 plus freight - were we to offer it in limited quantities. However, under this bulk upgrade offer, and when provided directly through the Sydney office, the price will be A$875 plus A$120 return P&P worldwide. Our contractors have allocated time from mid April to process the upgrades as one batch. This means that we must receive your unit prior to April 14th. However, we require all orders confirmed by 10th March 2008 to allow time to procure custom metalwork and necessary component purchases. Units will be processed in order of receipt of order with payment.
The upgrade order form is downloadable from our web site.

Order form PDC board upgrade http://www.deqx.com/beta/Q-PDC-UPGRADE.pdf


OPTION 2): Full upgrade to HDP3 (only available for SN:041101 and above):

This upgrade requires the PDC-2.6 Rev6 board and so is only available to PDC-2.6 and PDC-2.6P units with serial numbers above 0411011.

This is a complete upgrade to the new HDP3 using your modified PDC-2.6 Rev-6 main board. The serial number will remain your original serial number but with the HDP3 suffix to differentiate it from new PDC3 units. This upgrade

includes:
1. Items 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 13, 14 as in option 1 upgrade above.
2. New 2U high PDC3 chassis (double height of PDC-2.6) with brushed black or brushed silver anodized 12mm front panel.
3. Allows both digital outputs and balanced analogue (transformer or electronic) outputs simultaneously.
4. Analogue +/-15V 60VA main power supply switchable 115V or 230V.

Our price for this upgrade would normally be A$2,850 plus freight. However under this bulk upgrade offer, and when provided directly through our Sydney office, the price will be A$1,875 plus A$180 return P&P internationally. As above, our contractors have allocated time from mid April to process the upgrades as one batch. This means that we must receive your unit prior to April 14th (note that for this upgrade we can accept PCBs only, because we will not reuse chassis or PSU). As above however, we require all orders confirmed by 10th March 2008 to allow time to procure custom metalwork and necessary component purchases. Units will be processed in order of receipt of order with payment. The upgrade order form is downloadable from our web site.

Order form PDC to HDP upgrade
http://www.deqx.com/beta/Q-PDC-HDP-3-UPGRADE.pdf
 
terry j said:
BUT, has anyone who is conversant with setting up this stuff on a pc ever used it to give the iterative process on Ambiophonics a go?? and if so what were the results for you? I don't need the pc stuff right now fro stereo as I use the deqx, but I'm very interested in trying ambiophonics with it's crosstalk cancelling algorithms...which throws me straight back into my computer illiteracy.

If however people have tried it and think it's worth someone else investigating Ambiophonics, well it at least gives me an incentive to try and work this stuff out.

I have 2 ambio setups, one HW with XTC DSP and JVC XP-A1010 ambience boxes, one with PC and multichannel sound card.
What can I say, the HW solution is more convenient and foolproof, the PC is more flexible to try out things, but has all the disadvantages of a PC :xeye:
 
ShinOBIWAN said:
Acourate. You just take the FIR filters and use the 'convert to min phase' function of the software. Very easy and you keep the benefits like DRC and driver linearisation but you do have to redo the individual driver delays used for time alignment.


I thought min phase FIR was different than IIR? (even if using small number of taps)

The latency of an IIR filter essentially being the minimum your sound card supports.
 
Fascinating thread gents! I commend you on the bleeding edge approach.

I can't get comfy with the idea of trusting my tweets to m$. :( Has anyone tried or thought of trying XP embedded as a host?

I've read the entire thread. I'm left wondering if the DRC is completely dependant on the px-xo or can advantages be gain running just DRC to external, to the PC, passive or active x-overs.

Shin that CSD blows me away, man I'd love to hear these systems.

Regards
 
OTMOPO3OK said:
manisandher,
I'm trying to keep this thread alive so I was wondering if you finally got DEQX and how was your comparison to Acourate and maybe even Frequency Allocator? Whenever you have some free time please let us know your findings.
Thx

Hi Everyone,

Last year, I only managed test the DEQX PDC2.6 against my usual Esoteric D70 and two Pass Labs XVR1 x-overs (one per channel to give me a 3-way x-over).

I used Hypex HG700 amps (four per channel) and my rewired Wilson Benesch Chimera speakers to carry out all listening.

Overall, I was disappointed with the DEQX unit.

I liked the flexibility it offered and the room correction functionality. I could easily adjust for the annoying 50 Hz peak and 100 Hz dip in my room.

Oh and the software was great - I loved it.

However, in comparison to D70/XVR1 combo, the PDC2.6 sounded flat and lifeless.

I returned the unit to DEQX and explained my reasoning to Kim. He suggested that I try the PDC3.0 when it is available and seemed to understand my findings (although he didn't explicitly say this as such) .

I'm still hoping to be able to carry out the full test with the PDC3.0 at some point.

Mani.
 
Great to see that Stig Erik is going the PC path. I am still using my 200kg "Petit Filou" speakers. They combine Lynn Olsens Ariel ME2 and Stig Eriks Almghty Subwoofer in a two meter tall tower.

I use my speaker setup together with my HTPC and 46inch LCD, so mostly TV, Film and a little music, no gaming (I have another PC for that).

Since I have been sensitive to delayed sound in film I have used Thuneau's "Frequency Allocator Light". Given that the FF codec pack allow to set an image delay that point now seems obsolete. I will start playing with Acourate ASAP (I have had a Behringer mike for two years but never got around to use it).

Now over to a few questions, does Acourate support Vista?

I am using an RME HDSP 9632 with a 8 channel expansion board. The sound quality is great, but there is something that really annoys me. That is changing volume! The RME does not allow volume control with a remote control. Shinobiwan has written about it earlier I think, but I have never understood fully how his solution work.
What I have done instead is that I have taken an Audiotrak Produgy 7.1 Hifi and routed the SPDIF into the the RME. So the Wave volume is Controlled on the Audiotrak. The Audiotrak is in effect the default soundcard. I would like to get rid of the Audiotrak card and only use the RME, however I still would like to use a remote, can someone explain how this is done? I need a solution with a high WAF (Wife Acceptance Factor).

The remote/mouse/keyboard I use at the moment is the Logitech S 510, I have noted that the Wireless signal is badly affected when my Amplifier is on. If anyone have had similar problems, what was your solution?

Here is a rundown if some of the equipment I use:
PC: AMD Athlon 64 X2 4200 1 Gb Ram (due for an upgrade)
Case: Origenae X15e
Media Library: I have a dedicated Windows Home Server with 2 TB of fully duplicated storage
Power Amp: Audiodigit T-AMP 8*100W

This is a fantastic thread, I have followed it from the very beginning. Shinobiwan has been a great inspiration.

Many Thanks,
Fredrik Karlsson Peraldi
 
ackcheng said:
I think Shin is using Vista? Anyway, it is a software that do the measurement and generate the neccessary filters. Once everything is done, Acourate is out of your playback chain. so you are free to use OS for your playback

Thanks ackcheng for you quick reply! And I assume Convolver will run on Vista.

One thing that confuses me (given my previous post I must seem like a very confused person) is Vistas Audio Stack. I get the impression that the Vista API's are capable of replacing ASIO and all the Steinberg stuff. But nobody seem to develop applications to run with Vista natively. Am I completely wrong?
 
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