DSP Xover project (part 2)

Hi Vincenzo,
Glad to read you're going well with the Najda.
Indeed, in order to clarify, all FIR filters are open circuits by default, so you need to make sure the appropriate coefficients are loaded for the whole signal path - including the input section.
Regarding the preset slots. You can only store 3 FIR presets, in slots 1 to 3. IIR presets can be stored in any slot 1 to 9. The difference between FIR and IIR is the number of coefficients involved in a preset. Not too many for a IIR preset, quite many for a FIR. Basically, there wouldn't be enough space in the memory for 9 big FIR presets.
This is all documented in the NUC help file. Not that it bothers me to reply, but just to say I've put quite a lot of information in this document and I hope you could find useful stuff in it.

Nick
 
When can we expect the mixed IIR-FIR option that was on the "top 5" list for the next version ?
Jean Claude

Hi Jean-Claude,

Sorry for being late with this upgrade. I'm currently busy with preparing a new production batch because I'm going to be out of stock soon. The first batch was produced in China, the next batch is going to be produced in Europe - so I'm currently stocking boxes of parts at home. I'm also working on the expansion board, it will have an additional symmetric analog input and 2 additional analog outs.
The IIR/FIR mix will definitely be included in the system, please have a bit of patience - or join me in the programming team if you want to speed up the stuff.

Nick
 
Hi Jean-Claude,

The IIR/FIR mix will definitely be included in the system, please have a bit of patience - or join me in the programming team if you want to speed up the stuff.

Nick

I was just wondering as it seemed to me it was expected last November... but anyway, not being a programmer myself I can't help here, and as you said I just have to wait...

Jean Claude
 
Hi Jean-Claude,

Sorry for being late with this upgrade. I'm currently busy with preparing a new production batch because I'm going to be out of stock soon. The first batch was produced in China, the next batch is going to be produced in Europe - so I'm currently stocking boxes of parts at home. I'm also working on the expansion board, it will have an additional symmetric analog input and 2 additional analog outs.
The IIR/FIR mix will definitely be included in the system, please have a bit of patience - or join me in the programming team if you want to speed up the stuff.

Nick

Two more channels? That´s amazing. That really makes the whole thing future proof. Will the two extra channels be seamlessly integrated (will volume control work in the same way for example?) And will the expansion board be 100% compatible with the old batch of boards, or does it make sense to wait for the new batch?

One more question: If I wanted to get a Najda box and not have the "nevrosa" about not having the most silent implementation possible, is there any "best practice" in terms of power supplies and grounding? Ideally, I would like to build it once, never open it again, knowing that it´s a top implementation. Any tips?(perhaps it has been covered already in this long thread?)
 
Ok, here´s a first draft of the Najda front panel I might use:

Two displays, two IRs, two USB connectors, one button. :)

The first display will show the Najda information.
The second will show the source selected from Twisted Pear Mux, and also the status from the LEDs outputs from the Najda. An Arduino will set the sources and read the LED status and display both on the second display. It will also read the button and send the signal to the Najda and possibly other things, such as triggering amps on/off etc.

The button is a heavy duty chrome button from parts express, flush mounted.

The two USB connectors are for programming the Najda and the Arduino. The Neutrik USB connectors are reversable - I will have the square end facing out, not the rectangular one.

LCD and IR receivers protected behind acrylic.

I will follow Nick´s suggestion and build a small keypad and hide it inside the cabinet, so that I can (re)program the remote when necessary.

Comments? Suggestions for improvements?
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Will the two extra channels be seamlessly integrated (will volume control work in the same way for example?)

Yes: the main board Najda will detect the expansion when installed and unlock settings options in the firmware and the software.

And will the expansion board be 100% compatible with the old batch of boards, or does it make sense to wait for the new batch?

There's no functional difference between the present batch and the forthcoming one.

One more question: If I wanted to get a Najda box and not have the "nevrosa" about not having the most silent implementation possible, is there any "best practice" in terms of power supplies and grounding? Ideally, I would like to build it once, never open it again, knowing that it´s a top implementation. Any tips?(perhaps it has been covered already in this long thread?)

Hm I suspect there might be as many ideal setups as there are users. As far as I'm concerned, I've just connected a SMPS and haven't looked back since then. Some users have built their own linear supplies. Most power lines have their own regulators on board. Actually, the only lines that are not regulated are the +/-12V rails powering the opamps.

Ok, here´s a first draft of the Najda front panel I might use:

Two displays, two IRs, two USB connectors, one button. :)

The first display will show the Najda information.
The second will show the source selected from Twisted Pear Mux, and also the status from the LEDs outputs from the Najda. An Arduino will set the sources and read the LED status and display both on the second display. It will also read the button and send the signal to the Najda and possibly other things, such as triggering amps on/off etc.

The button is a heavy duty chrome button from parts express, flush mounted.

The two USB connectors are for programming the Najda and the Arduino. The Neutrik USB connectors are reversable - I will have the square end facing out, not the rectangular one.

LCD and IR receivers protected behind acrylic.

I will follow Nick´s suggestion and build a small keypad and hide it inside the cabinet, so that I can (re)program the remote when necessary.

Comments? Suggestions for improvements?

Looks nice! But why would you use the Arduino to set the sources? That's not clear to me.

Do you have a source for LCD acrylic protective covers? I'm looking for a supplier ;)
 
Looks nice! But why would you use the Arduino to set the sources? That's not clear to me.

I interpreted this as meaning - set the sources on the Twisted Pair mux

I rather like this concept - an independent second controller and interface for any additional functions not built into najda. I also like the concept of the second to display/instead of the status LEDs.
 
Hm I suspect there might be as many ideal setups as there are users. As far as I'm concerned, I've just connected a SMPS and haven't looked back since then. Some users have built their own linear supplies. Most power lines have their own regulators on board. Actually, the only lines that are not regulated are the +/-12V rails powering the opamps.

Would it make any sense to add a regulator here?

Is it a problem to use separate linear power supplies for the different voltages and connect their grounds together?

Looks nice! But why would you use the Arduino to set the sources? That's not clear to me.

I have a few sources, TV decoder, Sonos Connect, XBOX 360, Mac Mini in the rack. Perhaps I'll add an Apple TV. Will probably keep the Najda on one source, and have the inputs selected from one or several Mux-es. Another option is to combine it somehow - but I haven't worked that out yet. But that would add some complexity in use (or I have to be clever with the Arduino). I have the transformer-isolated input boards, so with those I could avoid any ground connection between the sources and the Najda. The squeezebox for example, does not have a transformer on the output (which it should I think). I don't use it anymore, as I switched to Sonos, but I think it's safer just to have transformers on all the inputs.. So: More inputs, and no ground connections are the reasons.

Do you have a source for LCD acrylic protective covers? I'm looking for a supplier ;)

For a one-off project, I was planning to have modushop or Schaeffer mill the front panel. I don't know if modushop can also provide the acrylic cover, but I know that Schaeffer / front panel express can make them in different thicknesses. Probably too expensive for production runs though. I know a company in Norway that could probably do it - called Kreativ Plast (www.kreativplast.no). I'm not sure if they are competitive on large quantities, but they can do everything with acrylic.
 
So you haven't tried the digital volume control in the Sabre/Buffalo? You should. You may change your mind :)

I'm using the digital volume controls on the Opus DAC from TP audio now (6 channels to feed my Linkwitz Orions). Volume is set via SPI from an Arduino.

What I notice is a slight hiss from the midranges/tweeters when the room is totally quiet. As I'll probably experiment with something high sensitivity, I had concluded that a analogue volume control (digitally controlled) would be the best option, as it will keep S/N ratio more constant (and therefore keep the speakers more quiet when they are not in use). Do you think that digital volume control can be used with high sensitivity speakers and the 26db gain in the Hypex amps, or do you think that one would need to lower the gain on the amp side of things to keep that noise away? In other words - can the system be quiet (given a good grounding scheme etc) with digital volume control and 26db gain amplifiers?

For the Orions I hardly need 26db gain with the Opus DACs and the Orions except for the woofer channels, so I'm imagining lowering the gain for the midrange/tweeter channels anyway for several reasons:
* Lower hiss
* Safety (less "Bang" if something happens)
* Use more resolution in the midrange/tweeter channels digitally
 
OPamps replacement

Hi to all,

last week F2a11 and I tried to see if there is a difference in sound quality (SQ) by replacing the LM833 Opamps of the Outputs from Najda board. We tried the LM 4562, LM 2604 and the Opa 627 ( 2 of them on a PCB adapter). Our audio system is a stereo horn arrangement with 3 drivers per channel.
We know that just replacing the OPamps is not the absolute correct way, because we did not adapt the peripheral parts connected to the Opamps to optimize their working point.
Replacement had an influence on sound quality ( not a great one) and here are our (perhaps subjective) results that we perceived:

1. In our opinion the original LM833 in the Najda board works very fine. Tight bass, well resolved mids and highs, perhaps no so much microdetails in the highs, but a warm and a very analoglike reproduction.

2. Replacement by LM4562: The bass reproduction was not so tight, a litte bit better resoultion of the highs, sometimes a little bit harsh, overall impression was that SQ looses the somewhat warm, analog character, a little bit cool reproduction.

3. Replacement by LM 2604: Almost the same as with the LM 4562, more analytical.

4. Replacement by OPA 627: Again a wonderful bass reproduction, very tight ( similar to the LM 833), but the best resolution of the mids and highs, a little more microdetails in the highs, very dynamic, the analog character is still there.

In summary: The differences heared between the LM833 and the other OPAs, although perceivable, were not large, and also depend on great extent of the music which is reproduced. The one we prefered was the OPA 627 because of the little more resolution of the highs, but could also perfectly live with the LM833. Congratulations Nick!
 
What I notice is a slight hiss from the midranges/tweeters when the room is totally quiet. As I'll probably experiment with something high sensitivity, I had concluded that a analogue volume control (digitally controlled) would be the best option, as it will keep S/N ratio more constant (and therefore keep the speakers more quiet when they are not in use). Do you think that digital volume control can be used with high sensitivity speakers and the 26db gain in the Hypex amps, or do you think that one would need to lower the gain on the amp side of things to keep that noise away? In other words - can the system be quiet (given a good grounding scheme etc) with digital volume control and 26db gain amplifiers?

For the Orions I hardly need 26db gain with the Opus DACs and the Orions except for the woofer channels, so I'm imagining lowering the gain for the midrange/tweeter channels anyway for several reasons:
* Lower hiss
* Safety (less "Bang" if something happens)
* Use more resolution in the midrange/tweeter channels digitally

What amps are you using?
what hiss do you hear if your short the amps' input? - ie. is the hiss coming from the amp or the DAC?
What (digital) output level are you using on the DAC?

You ought to try to gain balance the whole of the system. I don't believe the inherent noise in the Hypex is a problem - as I understand it, the noise level is pretty well constant whatever the output power (hence more powerful class D amps have a better overall s/n ratio, simply because the signal part is bigger)

I'd suggest you'd want your typical/high volume listening levels to be at, say, -10 to -30dB on a 32bit DAC, simply so you're limiting how much of the resolution you're throwing away. With (say) a 115dB s/n ratio from an ESS DAC, the noise level should therefore be between -100db to -85db (ie you shouldn't hear any hiss)
 
What amps are you using?
what hiss do you hear if your short the amps' input? - ie. is the hiss coming from the amp or the DAC?
What (digital) output level are you using on the DAC?

You ought to try to gain balance the whole of the system. I don't believe the inherent noise in the Hypex is a problem - as I understand it, the noise level is pretty well constant whatever the output power (hence more powerful class D amps have a better overall s/n ratio, simply because the signal part is bigger)

I'd suggest you'd want your typical/high volume listening levels to be at, say, -10 to -30dB on a 32bit DAC, simply so you're limiting how much of the resolution you're throwing away. With (say) a 115dB s/n ratio from an ESS DAC, the noise level should therefore be between -100db to -85db (ie you shouldn't hear any hiss)

Opus DACs have max 2vrms balanced outputs. I am setting the volume on the mid and tweeter DACs 15db lower than the woofer DAC. This is just to preserve some bits due to the high dipole compensation for the woofers.

I´m using Hypex UCD180HG for the mids/tweeters. I can´t test what you suggest now because they are in a tight chassis with a nanoDigi and DACs, and I wouldn´t want to open it to experiment.

I doubt that the amps are generating noise by themselves. However, any noise input to them is amplified with 26db. When using a digital volume control, any noise from the DAC is always there, while with an analogue solution the noise would be attenuated. However, you are probably right that a good DAC implementation would have low noise.. But in case it´s not optimal, an analogue volume control prior to the Hypex amps would attenuate the noise along with the music signal, reducing the noise to be amplified At least that´s what I was thinking.. I also think it´s useful to have an analogue volume control in the system in case of a glitch in the DACs. I have had a couple of times where a light switch close to the system has managed to throw off either the nanoDigi of the DACs, and all I got was white/pinkish noise at rather high levels. Not too pleasant if the amps are directly connected to a compression driver I suppose..
 
For a one-off project, I was planning to have modushop or Schaeffer mill the front panel.
I've ordered a modu case with custom CNC-ing which is currently in transit on its way to Australia. The CNC work I wanted done was slightly ambitious, but well within the capabilities of CNC machining. I am pretty hopeless with CAD, so sent them a PDF generated by something like powerpoint with the measurements, cutouts and other design work and paid their fee to have them generate the CAD files.

After working on the drawings for maybe 2-3 weeks over the Xmas break, they got back to me with a very basic design that had cutouts for my buttons, LCD display and meters. My meters were not flush-mounted and the primary 'really cool!' part of my design (actually shaved ~15-20% of the surface, so kinda ambitious) was completely missing, replaced with the horizontal 'feature' line seen in one of the photos of a case on their site. To be told they couldn't do my design after already taking my money and working on an alternative was disappointing to say the least. We went back-and-forth a few times but in the end my meters are still not flush-mounted and my case will look far more mainstream than I wanted.

So.. I guess the point of that story is to make sure you understand what the design house you are working with is capable/willing to do for you. Your design is reasonably simple but you'll need to make sure modu/schaeffer will do the flush-mounting as modu didn't for me.

Had they been up-front, I wouldn't have used them as I have access to friends who are good at CAD and there are plenty of local joints willing to CNC aluminium :/
 
Opus DACs have max 2vrms balanced outputs. I am setting the volume on the mid and tweeter DACs 15db lower than the woofer DAC. This is just to preserve some bits due to the high dipole compensation for the woofers.
What I was trying to find out was what is your typical digital attenuation (dB) at normal listening levels. The 2V RMS will only occur with a full scale digital signal and the DAC at 0db attenuation

I doubt that the amps are generating noise by themselves. However, any noise input to them is amplified with 26db. When using a digital volume control, any noise from the DAC is always there, while with an analogue solution the noise would be attenuated. However, you are probably right that a good DAC implementation would have low noise.. But in case it´s not optimal, an analogue volume control prior to the Hypex amps would attenuate the noise along with the music signal, reducing the noise to be amplified At least that´s what I was thinking.. I also think it´s useful to have an analogue volume control in the system in case of a glitch in the DACs. I have had a couple of times where a light switch close to the system has managed to throw off either the nanoDigi of the DACs, and all I got was white/pinkish noise at rather high levels. Not too pleasant if the amps are directly connected to a compression driver I suppose..

Indeed, it's most likely that attenuating the relative noise from the DAC will cure the audible noise you've described (I don't think that's really open to doubt).

The question is how best to achieve this.
- you can change the gain on the amp or you can attenuate the output from the DAC (attenuating the noise with it)

If you attenuate the DAC you can do it by:
- changing the output circuit amplitude/gain
- fixed attenuator
- volume control

Using a whole new circuit for a volume control open up all sorts of new discussions about the audibility of volume controls :D
 
What I was trying to find out was what is your typical digital attenuation (dB) at normal listening levels. The 2V RMS will only occur with a full scale digital signal and the DAC at 0db attenuation

Right.. lets see. Because the woofer channels need 20db gain at 20hz, I need to lower woofer output levels by 15db flat to ensure no digital clipping. This leaves me with an "anemic" bass output from the DSP, but gives the best use of the digital resolution in all three channels (bass, mid, tweet).

At the DAC level, I then always set the bass level 15db higher than the other channels. So that when I am at 0db (maximum), the only channel actually at 0db is the bass channel - the others are at -15db.

This provides ample maximum volume for the Orions in my home, ensures no clipping, and places volume control in the DAC only instead of halfway DAC/halfway DSP. I didn´t measure the SPL this gives me, but unless the recording is very quietly recorded, the SPL is sufficient.

So - if I wanted more SPL out of the Orions, I would either need to stop playing music with very deep bass content, or add a highpass filtre to the woofer channel (which I tried, and the tightness of the bass disappeared, so I´d like to keep it as flat/low group delay as it was intended).

What I did have in mind, to optimize the system as it stands, was to just remove the "gain-setting" resistor from the 180HG amps, as this would give 13db gain instead of 26db for the mid/tweet amps. I think that would be just about perfect gainstaging in both the digital and the analogue domain.

At normal listening levels, say, watching TV in the evening, I guess I have the volume control at -25db, which means that the bass DAC is actually outputting -25db while the mid and tweeter channels are at -40. But late night after a few beers friends will happily play at 0db (-15), and the speakers won´t complain.

Indeed, it's most likely that attenuating the relative noise from the DAC will cure the audible noise you've described (I don't think that's really open to doubt).

The question is how best to achieve this.
- you can change the gain on the amp or you can attenuate the output from the DAC (attenuating the noise with it)

If you attenuate the DAC you can do it by:
- changing the output circuit amplitude/gain
- fixed attenuator
- volume control

Using a whole new circuit for a volume control open up all sorts of new discussions about the audibility of volume controls :D

True. I was thinking that an analogue volume control in a chip (like the one in the Najda) would be as transparent as a fixed attenuator. Certainly more flexible. Changing the amp gain seems to me the best solution altogether, as it makes little sense in attenuating something just to amplify it again. :)

Is there any data on the effects of these volume control chips? And how does the DACs in the Najda compare to say a Buffalo DAC? (note: A Buffalo DAC has different performance if used in stereo and 8-channel mode as you probably know)
 
Would it make any sense to add a regulator here?

Is it a problem to use separate linear power supplies for the different voltages and connect their grounds together?

If your supply is reasonably clean, then I don't think a regulator is required for the opamps.

I don't see any problem with having a common ground for all the supplies.

For a one-off project, I was planning to have modushop or Schaeffer mill the front panel. I don't know if modushop can also provide the acrylic cover, but I know that Schaeffer / front panel express can make them in different thicknesses. Probably too expensive for production runs though. I know a company in Norway that could probably do it - called Kreativ Plast (www.kreativplast.no). I'm not sure if they are competitive on large quantities, but they can do everything with acrylic.

Cheers! ;)


Thanks Pascal, that's interesting too indeed!

last week F2a11 and I tried to see if there is a difference in sound quality (SQ) by replacing the LM833 Opamps of the Outputs from Najda board. We tried the LM 4562, LM 2604 and the Opa 627 ( 2 of them on a PCB adapter). Our audio system is a stereo horn arrangement with 3 drivers per channel.
We know that just replacing the OPamps is not the absolute correct way, because we did not adapt the peripheral parts connected to the Opamps to optimize their working point.
Replacement had an influence on sound quality ( not a great one) and here are our (perhaps subjective) results that we perceived:

1. In our opinion the original LM833 in the Najda board works very fine. Tight bass, well resolved mids and highs, perhaps no so much microdetails in the highs, but a warm and a very analoglike reproduction.

2. Replacement by LM4562: The bass reproduction was not so tight, a litte bit better resoultion of the highs, sometimes a little bit harsh, overall impression was that SQ looses the somewhat warm, analog character, a little bit cool reproduction.

3. Replacement by LM 2604: Almost the same as with the LM 4562, more analytical.

4. Replacement by OPA 627: Again a wonderful bass reproduction, very tight ( similar to the LM 833), but the best resolution of the mids and highs, a little more microdetails in the highs, very dynamic, the analog character is still there.

In summary: The differences heared between the LM833 and the other OPAs, although perceivable, were not large, and also depend on great extent of the music which is reproduced. The one we prefered was the OPA 627 because of the little more resolution of the highs, but could also perfectly live with the LM833.

That's a great feedback Sigi, thanks a lot for that. I can send you a few capacitors if you want to experiment with that also.