DSP Xover project (part 2)

I had a quick look at AD1865 datasheet, and it doesn't seem to support I2S. I'm not sure whether we talk about the same chip though, the one I looked at is an obsolete part from AD with 18-bit resolution.

If that's the one, then you'll need additional glue logic if you want to hook these DACs to Najda on the expansion port.

Now I'm wondering: the boards that you have, how do you feed audio in them? If you can feed SPDIF, then you just plug your boards to Najda's SPDIF outs and it's done.

Yes you are right, it's an old DAC but with a reputation of being one of the most natural sounding ever. Did not tried all of course but very happy with it. The design of the board is a clone of a famed NOS Audionote designed DAC, although not multi-channel. I feed it in SPDIF.

What worries me is that in XO application, eg 3 channels, I'll have one stereo DAC for Woofer L/R, another for medium and a third one for tweeters, all with their own clock. Isn't it a problem with phase or Jitter ?

BR
 
Understood about the enclosed - I will make it enclosed:)
I'm used to trouble shooting 480v DC KT88 valve amps.
Enclosed ones I've been able to find are few and far between at the rating / spec needed over here. Someone will no doubt prove me wrong...

Found this one that looks promising

http://www.xppower.com/pdfs/SF_ECP40.pdf

Bit beefier than needed but the lower models available here don't have such good regulation figures.
+ as Nick pointed out, I will have plenty of power for the expansion board for 5 / 6 way in the future.

Gonna be fun to start this project...
 
Ladies and Gentlemen,

I received my board yesterday... and fortunately I have a couple of days off. I did some interconnections yesterday evening and here is the first power on !

As you can see on the picture, the idle consumption is about 5VA (3VA on the +/- 12V) and less than 2VA on 5V. I measured 1.5VA in Standby mode.

On the right side of the picture is some of the measurement gear that I will use to check the board now.

Stay tuned

Best from France
JC
 

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I have started to play with it...

On first picture where you can see measurement of a quick three ways Linkwitz Riley 4th at 150 and 2kHz. The additional high pass curve is the one with the the CS3318 set at 6.0 Vrms. Others are outputed with the nominal output set at 1 Vrms.

Second picture is to show the board configuration where you can set the DAC interpolation filter along with the nominal output.

Be aware of a small bug with the current version of software/firmware, the Bass / Trebble optional controls introduce a phase aberration even without any correction selected. Well, it took me a couple of hours, scratching my head with some measured phase deviation before tracking that one !

Jean Claude
 

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A quick illustration of possible use:

on first picture, I have loaded the frequency response of a small mid woofer (Scanspeak 18W4531-G0 obtained by a combination of a near field measurement up to 250 Hz and a free field above on axis. No smoothing at all. You can see how well ScanSpeak was able to control the break up modes ! I love and use this driver by the way!

Chanel 8 is just here to illustrate the target (LR24 150Hz-2kHz).

The next picture is the same response corrected by only a Highself. not bad already !

xx_4a & 4b show you the same driver with the LR24 filters enabled and a couple of EQ added. Please note the vertical resolution (1 dB/step).

The result is quite good but we can see a deviation from theoretical below 100 Hz (only one half of an octave below the cut off freq).

xx_5a & 5b are just to show you that, sometimes, it's better to use different type and order for the filter to meet the target. We are now close to the target down to 60 Hz with the use of a Bessel third instead of the LR24 and all that with only 5 EQ beside the filter by themselves.

So what, at 300€ it looks to me a real bargain, thanks Nicolas. Of course there is a lot of room for improvement but with a firmware labeled at rev 0.3 and a software at rev 0.01 ? congratulation, you have already a winner
 

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@ jcga: I had planned to post some of my own efforts here. I use the exact same driver as a mid.

I would really like to see an FIR implementation. Any recommendations for suitable software to do this?

I'm used to play with IIR filter for a long time now but I solely bought this board to start to "play" with FIR implementation. Unfortunately I'm a complete noob on the subject but I will dig in it in the upcoming weeks and we could share our findings if you wish.

Someone has already developed "Rephase" which looks very nice and, if I understand it correctly, will be able to give the FIR values needed. I will investigate this way.

JC
 
What worries me is that in XO application, eg 3 channels, I'll have one stereo DAC for Woofer L/R, another for medium and a third one for tweeters, all with their own clock. Isn't it a problem with phase or Jitter ?
I wouldn't worry too much. All 3 DACs will lock on the same clock, and actually the reason for providing SPDIF outputs on Najda is precisely to allow you connecting your own DACs.

Found this one that looks promising

http://www.xppower.com/pdfs/SF_ECP40.pdf

In the datasheet, they say "10% minimum load required on V1 & V2 of multi output versions". So you must draw at least 500 mA on 5V and 200 mA on +12V, which is a lot. I'd recommend that you look for a smaller supply.


Be aware of a small bug with the current version of software/firmware, the Bass / Trebble optional controls introduce a phase aberration even without any correction selected.

I'll look at this and I'll fix it for the next release. Phase should be flat with Bass and Treble both set to 0. Thanks for reporting that.

Sorry if this has been covered before. Can we input biquad filter coefficients with this software?

That will be made available with the next release.

@ jcga: I had planned to post some of my own efforts here. I use the exact same driver as a mid.

I think we could open a separate thread with speaker tuning examples ;)
 
PSU's.
It's quite hard to find one with a good regulation, low ripple, high freq and not too thirsty..
This one is 3.2A, 2.0A and 0.3A but still at 0.4A and 0.2A
http://docs-europe.electrocomponents.com/webdocs/066e/0900766b8066ef1f.pdf

So, does it mean it won't work at all if Najda is not drawing enough or other nasties?
Can always get it to power something else to draw more and waste it:(

Alt is 2 or 3 separate supplies I suppose.
 
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Yes it's taking a bit of time to find the proper PSU. That's a major part of the system.

If you're looking at the datasheet of the one I'm using, you'll see there's no minimal load (table on 2nd 'Output Specifications', min Load). So it's not always that PSUs have such min load requirements.

I would really get a single unit that delivers all 3 required voltages.

PSU's.
It's quite hard to find one with a good regulation, low ripple, high freq and not too thirsty..
This one is 3.2A, 2.0A and 0.3A but still at 0.4A and 0.2A
http://docs-europe.electrocomponents.com/webdocs/066e/0900766b8066ef1f.pdf

So, does it mean it won't work at all is Najda is not drawing enough or other nasties?
Can always get it power something else to draw more and waste it:(

Alt is 2 separate supplies I suppose.
 
Hi all,
Got my board yesterday and connected it up with LCD, LED's and buttons and it works great.
It would be nice with a possibility to load the biquad coefficients from e.g. "ACD".
I tried loading a FIR filter that I made in "Acourate". I had to generate a 512 tap filter (1024 is one tap too much) and save it as TDS *.dat file and it loads fine, but the low freq resolution is not good enough at 512 taps:
see attachment...65535vs512 taps
How many taps is possible for each channel?

I see that HOLM HOLM Acoustics is using something they call "16x decimation" that equals 16368 taps @ 44khz.

Anyway, this seems very promising for the first release. Thanks a lot Nick, you have done a great job!

best,
Paal
 

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Is the Nadja sensitive in which order power comes up, and what restriction on rise time do we have to consider?

Otherwise I would prefer to apply power to digital section and ADACS first, and then when this is out of "boot" then apply to the analogue output section to minimize any experiences with digital noise output on speakers...

Still risks of thumping?