The making of: The Two Towers (a 25 driver Full Range line array)

The Xonar ST has a Molex input plug for power:
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Koldby did explain it. But what I simply do not understand is why did the schematics show a 6dB gain pré-amp circuit (which is present in the real deal, not in my amp as koldby left that part out) while the amp already has 29 dB gain without it. Is this to be able to adjust the gain with plus or minus 3 dB steps, ranging from +9 dB to minus 9 dB? Something the real deal has as well according to its specs.
The reason the preamp section is there is to cope with a wide variety of sources that put out very little voltage in comparison to a 2V DAC. Kolby built it for a 2V DAC no need for a preamp. Most consumer sources run at -10dBv, then there is the fairly standard 0dBu and the professional +4dBu here is a shot from the sengelpiel calculator showing the RMS and peak values of these standards. We are talking 0.3V to 1.2V RMS. This is why a preamp with gain is needed if you want to get maximum power out of a source with that sort of voltage. These are not unusual so commercial manufacturers allow for them.

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Regarding amplifier input you only need to know 1 parameter, amp gain. From there you can calculate the required input voltage for required output wattage.

Use maximum wattage required, calculate voltage then use the gain factor of amp to divide that voltage. This will give you the input voltage required to drive the amp to that wattage. Assuming the amp can supply this wattage.

P = (V*V)/R to get amplifier voltage, V. Gain = 20*(log)(V/Vo) = amplifer gain. Calculate Vo. I'd get a dac that can go slightly higher than Vo just to be safe, sometimes when DAC reaches its maximum Vo things get dodgy, depends on the DAC.

This is the same calculation included in the Hypex datasheet, and the answer to it does not make how much single ended voltage is needed any clearer, as the Hypex datasheet say this is differential voltage, so perhaps they are using a combiner with gain to keep the voltage the same after conversion to single ended or maybe they have moved to differential amplification the datasheet is not clear, they use the terminology Hot and Cold to describe the output which is often used with differential signals. So perhaps it does need 2.35V single ended to reach full power but for a different reason.

Without knowing the input stage of the amp or measuring it is hard to guess. Asking Hypex directly would probably clear it up.

Use this calculation for your Goldmund.

350W output at 8 Ohms load needs 52.9V (sqrt of 350x8)
Amp has 29dB gain this is a factor of 28.18

52.9 / 28.18 = 1.877 V Input needed to produce 350W with an 8 ohm load

With a 4 Ohm load it is only 1.32V to produce the same power. The difference is the current needs to be higher to produce the same power with a lower voltage.

Hypex 500W 4 Ohms (sqrt 500x4) = 44.72V / 19.05 = 2.347V same as 2.35V on the datasheet.
 

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Another thing to put it in perspective for the Goldmund is that reducing 2V by -0.75dB makes the voltage 1.82V so it is very close to making 350W at 0dBFs with an 8 ohm load.

This is another reason why the 6dB of preamp/buffer gain is not a bad idea as it allows the reduction in digital level to give some room for intersample overs and still let the amp produce all it can when needed.
 
While leaving room for overshoots as stated in their specs?
Goldmund specs said:
Power
Nominal power : 350 W RMS (2 - 8 Ohms), 175 W RMS (1 - 16 Ohms).
Instantaneous power (8 Ohms) : > 400 W
Maximum instantaneous power : 1000 W
Maximum voltage swing : 70 V peak.
Maximum current swing : 10 A peak.

That's basically the whole idea, as gain is added in a buffer stage, the same gain number (plus the -3 dB I was using) is reduced inside JRiver.
That's exactly what I'm doing with the Atom pré-amp, -6 dB inside JRiver, +6 dB added by the Atom.
 
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I thought it interesting that the D50 performed best on battery power since a switching power supply would otherwise be a primary source of noise. I hope its ability to run off a single 5V supply (USB) doesn't mean its op amps don't have the customary +/-15V supplies.
The D50 comes with a power adapter but it runs at +/-5 V, so no it does not have the voltage swing of 15V opamps but it is using the mobile version of 9038 chip, as it is single ended there is more than enough voltage swing.

Hi fluid:
What is your opinion as to dynamic headroom and how does that figure into the discussion?
I'm unsure if you are referring to amplifiers or DAC's now.


I guess I've ended up agreeing with your comment. I'm obviously biased in favor of balanced. Its kept me out of trouble more than once and I have an ART Clean Box in daily use between the RCA sub output from an AV RCVR and a pro amp.
Try using a pseudo balanced cable and getting rid of the Clean Box. Unless it is very well designed or you have significant noise problems the result will probably be better or equivalent without it. The balanced receiver by itself can still deliver 40dB of common mode rejection when pseudo balanced instead of 90dB for a full balanced connection. In a home over a short distance is 90dB needed?

If you have a truly differential DAC which most of the best chips are, then to get the best performance out of it a high quality I/V stage is needed the output of this is inherently differential and maintaining that all the way to the amplifier input makes a whole lot of sense as the only thing you need is a balanced connector and an amp with a balanced receiver built in or added.

To convert that differential output back to single ended another opamp combiner is needed inside the DAC which will reduce performance slightly.

Some DAC's built to a budget don't have the full blown I/V stage and do it all with a single opamp which gives a single ended output.

So in my mind at home using balanced or single ended is more about the source you have as it makes sense to use it as is. Adding extra balancing and unbalancing stages makes no sense to me unless noise is an issue.

If the devices are inherently balanced then it makes no sense to use them single ended either:)
 
While leaving room for overshoots as stated in their specs?


That's basically the whole idea, as gain is added in a buffer stage, the same gain number (plus the -3 dB I was using) is reduced inside JRiver.
That's exactly what I'm doing with the Atom pré-amp, -6 dB inside JRiver, +6 dB added by the Atom.
The power available over the rated power would depend on the the main voltage rails and how much they can supply, if the rails are high enough then those amounts are realistic. Those specs suggest it is current limited rather than voltage limited so lighter loads would let it put out more power.
 
The D50 comes with a power adapter but it runs at +/-5 V, so no it does not have the voltage swing of 15V opamps but it is using the mobile version of 9038 chip, as it is single ended there is more than enough voltage swing.

I'm unsure if you are referring to amplifiers or DAC's now.

If you have a truly differential DAC which most of the best chips are, then to get the best performance out of it a high quality I/V stage is needed the output of this is inherently differential and maintaining that all the way to the amplifier input makes a whole lot of sense as the only thing you need is a balanced connector and an amp with a balanced receiver built in or added.

To convert that differential output back to single ended another opamp combiner is needed inside the DAC which will reduce performance slightly.

Some DAC's built to a budget don't have the full blown I/V stage and do it all with a single opamp which gives a single ended output.

So in my mind at home using balanced or single ended is more about the source you have as it makes sense to use it as is. Adding extra balancing and unbalancing stages makes no sense to me unless noise is an issue.

If the devices are inherently balanced then it makes no sense to use them single ended either:)

Does it matter if I was referring to amplifiers or DACs? The discussion about precisely how much voltage was needed to drive the Goldmund to its full output was beating with the discussion about headroom. I wanted to know where you stood on the latter. As far as the gain issue goes, I would be interested to know if the absolute peak value out of the DAC is sufficient to drive the Goldmund to its 1000W instantaneous spec rating. Ideally gains would be set so that DAC and Amp would hit their absolute peaks at same input level. Else one distorts before the other and there would be incremental improvement if gain were changed so as to lessen the distortion at the earliest distorting stage.

This brings up the fundamental question I still have after all this time:
Is the e.g. 2V RMS, 5.76 Vp-p rating a maximum undistorted output such that gains must be scaled so peaks never attempt to exceed it?

Since music doesn't have a precisely defined crest factor, adequacy of headroom is always at issue unless we resort to overkill (and maybe even then depending on level of paranoia:)). That is why I like fully balanced DACs and interconnects and high voltage complementary op amp power supplies, and oversized power amps, and so on.

Is it necessary to go to this extreme? Probably not but I do recall (but haven't been able to find) a post by Tom Danley back in the day where he discussed dynamics and stated the true peak to average ratio for music was a lot more than the 8-12 db crest factor typically quoted for music and systems that provided extra headroom would be perceived as more dynamic. One of the things that distinguish a pro-audio Synergy used in the home is huge amount of headroom. Arrays are similar above their excursion limits and below where combing sets in.

You've convinced me on balanced vs single ended. I could/should have simply cut the shield return inside the XLR end of my RCA to XLR cable (hindsight). I'll do that on the odd interconnect if the need arises in the future but I will keep the bulk of my system as balanced and ground isolated as I can.
 
I remember reading that from Tom, wasn't there an old thread on the specifics of what makes a speaker sound dynamic?

Much of my exercise here is just that, trying to create headroom wherever possible. It is why I got a Hypex NC500MP instead of the NC250MP.
It is why I was pretty anxious to see what that Goldmund could do having way more power than the Pioneer it replaced. It blew my mind in our first test, but I did struggle to find that exact ingredient again until recently.

The magic with the arrays is in the midrange, which is where most music lives. If the gain structure is close to right, it can sound clean and feel like there's no limit, effortless. Heck, turning it up does not even get noticed at first if it is a clean sound. Unless you get a reference, for instance trying to talk to the one sitting next to you. :p

For quite a while I had quite good sound, but less of that effortless feel. This made me search for the reason why, first in processing and more recently in the other parts of the chain.

I guess I would favour overkill as well. Even if we use a fraction of a watt most of the time, it's good to have lots on tap. :)
 
Does it matter if I was referring to amplifiers or DACs? The discussion about precisely how much voltage was needed to drive the Goldmund to its full output was beating with the discussion about headroom. I wanted to know where you stood on the latter. As far as the gain issue goes, I would be interested to know if the absolute peak value out of the DAC is sufficient to drive the Goldmund to its 1000W instantaneous spec rating. Ideally gains would be set so that DAC and Amp would hit their absolute peaks at same input level. Else one distorts before the other and there would be incremental improvement if gain were changed so as to lessen the distortion at the earliest distorting stage.

This brings up the fundamental question I still have after all this time:
Is the e.g. 2V RMS, 5.76 Vp-p rating a maximum undistorted output such that gains must be scaled so peaks never attempt to exceed it?

Since music doesn't have a precisely defined crest factor, adequacy of headroom is always at issue unless we resort to overkill (and maybe even then depending on level of paranoia:)). That is why I like fully balanced DACs and interconnects and high voltage complementary op amp power supplies, and oversized power amps, and so on.

Is it necessary to go to this extreme? Probably not but I do recall (but haven't been able to find) a post by Tom Danley back in the day where he discussed dynamics and stated the true peak to average ratio for music was a lot more than the 8-12 db crest factor typically quoted for music and systems that provided extra headroom would be perceived as more dynamic. One of the things that distinguish a pro-audio Synergy used in the home is huge amount of headroom. Arrays are similar above their excursion limits and below where combing sets in.

You've convinced me on balanced vs single ended. I could/should have simply cut the shield return inside the XLR end of my RCA to XLR cable (hindsight). I'll do that on the odd interconnect if the need arises in the future but I will keep the bulk of my system as balanced and ground isolated as I can.

Yes I think the difference between DAC's and amps is significant in regards to headroom. I typed most of what I am about to write before then deleted it as I thought the discussion was confusing enough at that point but as you asked...

In a digital system there is the theoretical peak of 0dBFS. Nothing should exceed that level to avoid digital clipping which is not a nice sound, sounds like the drivers suspension is broken if there is enough of it.

Due to bad mastering intentionally or otherwise signals can be generated and encoded that exceed 0dBFS. Usually this is the issue of intersample peaks where the interpolation of two samples allows there to be a peak in between them when converted back to analogue. Some DAC's handle this quite well most do not. The simmple solution is to add between 2 and 4dB of digital attenuation in the DAC or before and this problem goes away.

The reason I say there is a difference in headroom is because a digital system should be run as close to it's maximum level as can be to maximise resolution. This is less of an issue in 24 and 32 bit converters because they have inbuilt headroom digitally over what they can realistically reproduce in analogue terms. In the Audioscience review and measurements you will see a linearity figure in bits quoted. The best DAC's are around 20 bit some nudging towards 21 bits. That is the limit of the analogue resolution they can provide and this figure has not improved significantly over time as it is set ultimately by the thermal noise of the resistors in the circuit.

So between 4 and 12 bits can be lost in digital terms without affecting the output.

The second factor that determines the headroom requirement is whether the system is a pure replay system where the peak level can be set or whether it is live or somewhat uncontrolled where performers or inputs can vary beyond the control of the engineer.

Wesasyo and I are using volume levelling in Jriver that he explained well before. The analysis of the files will determine peak and average levels and will work as a sort of intelligent software limiter so the average levels are well controlled but the ultimate peak could still be somewhat variable so the 4 to 10dB digital reduction is sensible. making this back up in analogue gain or adding even more allows the amplifier to produce more power when needed but there will be limits to how much gain makes sense.

So then we get to the question of whether an amplifier can produce enough power to deal with the peak or have enough input voltage to get there anyway. And whether that is going to be a peak or RMS value. I would suggest that when describing instantaneous power that would prefer to peak voltage.

Using the Goldmund specs as an example for 1000W in 8 ohms 89 volts needs to be produced which would seem to exceed the 70V rails by more than is realistic, so 1000W in 8 ohms seems unlikely.

To get 89 volts with 29dB of gain would need 3.17 V which is more than the peak of 2.828 volts that could come from 2 VRMS.

600W in 8 ohms is 69 volts so that would seem realistic which could be achieved with a peak input voltage of 2.45V, which is available from the 2VRMS DAC.

So headroom in an amplifier makes sense too, how much depends on the rest of the system.
 
Just for your information, rails voltage is (plus/minus) 80V in the Goldmund amp according to the schematics.
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This example from the schematics shows 4 pairs which would use 60 volt rails, mine has 6 pairs that use the 80 volt rails mentioned. :)

On another note, the M1 DAC is said to have a resolution of 20 bit (Stereophile's second test corresponding to my model) and I use a fixed average listening level
not to eat too much of those bits as the DSP settings will use up (digital) headroom.
 

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Benchmark has nice article on intersample overs if anyone is interested.

Intersample Overs in CD Recordings - Benchmark Media Systems

Also it is my understanding that most power amplifiers are voltage gain devices. This means there are hard limits on the voltage rails (unless you are using class G/H device, but in either case the voltage rails are defined. This means for a given DAC input voltage you should get a defined output voltage (for a purely resistive load) base on amplifier gain.

Since we are talking power which has a current and voltage requirement, the headroom discussion seems to point to the capabilities of the amplifier itself (voltage and current capabilities) and not the DAC driving the amplifier.

The DACs job as I see it is to drive the amplifier to a given voltage. The amplifier then takes care of delivering voltage and current ie. power within its capabilities. I don't understand the talk of a dac having headroom? A dac can either drive the amp to maximum voltage or it cannot (assuming we are not clipping). I get confused when people are speaking about a voltage device but then talk about a voltage and current device in the same sentence. Not to muddy the board but I find it confusing :).
 
I guess I was looking at the right thread:

Hi Art
One of the things I have been talking about here is that the dynamic range of music can vary from type to type by such a large amount that this test does not take this into account.

I was wondering if you have ever looked at the amplifier output voltage while playing music and watching the level indicators?

Crown used to have the Idep or whatever it was (like the psa-2 etc) that made the light change if there was even an instantaneous difference between the input and output wave shape.
It was subjectively much too sensitive and was ignored in favor of the nice red “getting your monies worth” light that came on later.

About 10 or 12 years ago, QSC had an A/B/X test rig going around doing amplifier demo’s and they came by the shop when worked in Glenview. We collected a number of pro amplifiers to use as comparisons and I brought my Threshold stasis II in from home that I used as my reference. The only thing special it had was it could drive my electrostatic speakers or normal speakers and didn’t seem to care either way.

Using my music choices and after sitting for a while, the amplifiers seemed to fall into two groups and one exception which was my threshold.
It sounded exactly like the others in one group except that with some things, it sounded less dynamic.

On the fast looking VU indicators, the peaks were -10 to -15, never any clip lights. Being puzzled, I got a scope and looked at the outputs. At around the time it sounded different (and not that loud), the output Voltage was reaching the rails and flat topping on very short single transients. This was NOT audible AT ALL unless one was comparing to an unclipped signal and then one heard it as being less dynamic.

At that point, I switched to a larger pro-sound amp (then a pl-236) and have not gone back. I was taken back by that experience and since then often enough one can find not only amp limit issues but gain structure problems.
I think there is some confusion too on the title of this since what is needed depends on many things and in spite of many attempts to differentiate loudness from dynamic range I have been unable at least for many. Feels a lot like the old days trying to explain extension is not the same as loud bass haha.

Lets say one does find it takes 2Volts RMS to produce a sine wave test signal like here as loud as you would care to have it. With a sine wave, all you need is an amplifier than supply that 2VRMS and no more.
So let’s say your amplifier can swing the suggested 4X or 8Volts RMS and all is well.

Now, one puts in AES pink noise at 2VRMS, to supply this signal, one needs an amplifier than can swing 4VRMS because this signal’s envelope has a 6dB peak to average ratio over the sine. Use pink noise from Smaart or any RF test gear used in audio and one finds about 10dB peak to average ratio in the envelope. This test signal requires an amplifier than can deliver 6.6VRMS in order to produce the 2VRMS signal.

As you play older recordings before the loudness wars or modern ones that aren’t compressed, the peak to average ratio gets larger than this and one either clips instantaneous peaks (which if short enough you can’t hear like normal clipping) or turn the average level down.
For an extreme example if one plays either the Harley recording or fireworks recording at the same volume setting as Velvet revolver, one will hardly be able to hear it, yet they both would have the same digital peak levels and if you turn the gain up, you will be raising the peak level accordingly and then clipping it off at 8V.

By the time speakers sound strained or clipped, one is massively nonlinear measurement wise and I am talking about faithful reproduction of the input signal.

My futile point has been that one must consider both the dynamic range of what one is producing AND the desired average loudness in order to estimate if one has enough power (because if you don’t, the peaks will be clipped and you may never know it).
The ONLY way to know in a given circumstance is to look with a scope.
Best,
Tom

Source of the quote...

This story here above comes quite close to what I am saying/experiencing with the Atom Pré compared to without it.
Remember, I'm not trying to get to a higher level with this tweak, merely trying to preserve what should be there in the first place.

So far it has been working and even my family has noticed it, by complimenting the system while not being aware of what has changed.
I'm always doing something as far as they are concerned (which is true :D). My son is enjoying the sound immensely, so far (without him
knowing) I've established that the M1 DAC is the favorite. By him not knowing and just going on what he has to say I can compare to my
sighted experience.
I have not redone any processing. Even my 'Home Theatre zone' sounds way better, where the processing dates back to August 2019.
 
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Benchmark has nice article on intersample overs if anyone is interested....I don't understand the talk of a dac having headroom?
Nice article that explains well what I was trying to describe in words. Made me chuckle that they describe it as "high headroom interpolation" though ;)

It's really allowing a margin of error to allow for inter sample overs, EQ considerations etc.

The DACs job as I see it is to drive the amplifier to a given voltage. The amplifier then takes care of delivering voltage and current ie. power within its capabilities.

A dac can either drive the amp to maximum voltage or it cannot (assuming we are not clipping). I get confused when people are speaking about a voltage device but then talk about a voltage and current device in the same sentence. Not to muddy the board but I find it confusing :).
Very true, part of the discussion was whether a given DAC could provide the voltage needed to allow a power amplifier to reach it's limits. When you reduce the digital output to make allowances for digital hiccups some extra analogue gain may be required.

The power supply in the amplifier is what will determine the headroom as Tom explained in the quote. If the voltage rails are not stiff enough then the rail voltage will drop and if a high input voltage has been applied the peak will not be able to be produced by the amplifier.

Some of those problems of peak to average signal are made less of an issue by using volume levelling as the gain has been reduced to compensate for the differences. This is quite a different situation to a CD player being plugged directly into an amplifier and wanting to leave the volume at one setting.

A big amplifier with a strong power supply is a good way to avoid any of these issues. My MOD686 has an 800W SMPS per channel for a 400W amplifier for this reason.
 
I've got an answer from Hypex:
Hypex Support said:
Hello Ronald,

For the power amp it does not matter whether the input is single ended or balanced, the input sensitivity is 2.35V.

On some pre amps that have both unbalanced and balanced outputs, the output level for the unbalanced output is only half the level of the balanced output.

Kind regards,

So this confirms what happens within the Hypex amp.
For now my route will be getting 3 Universal Buffers and making sure the difference of going differential on the sub is taken care of in the gain needed.
Now I only need confirmation what happens if going from SE to Differential in one of Tom's modules.
 
Have asked Tom about mounting buffer inside power amp enviroment plus the voltage levels for conversion, so lets see what he say about it.

The big questions is always its not low cost get 3 or 4 of them at computer side of system, its 372€ plus vat for 3x raw boards then comes PSU plus box and plug terminals, so probably in € half a kilo. That Okto DAC was in € one kilo some months ago and it have four of them differential out build in, whatever analog output level per channel can be ordered and user can change it down the road solder few smd resistor with other value that set the gain level and then comes each channel or channel pair can in units display meny be set to whatever lower levels via chips digital 32bit.
 
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With meters for each of the stages, it's pretty easy to set optimum gain staging, and solve level matching issues.

First, a meter for the DAC, and set the DAC to pass peak signal just below clipping, 0dBFS.

Then, a meter for the analog output of the DAC, set to where the body of the signal resides at about -16 to -18dBFS, if using a 'digital scale' meter.
This is a good level for both consumer and pro DACs, as of course both need the same headroom.

If the DAC output analog meters reference voltage and not dBFS,
use their nominal voltage output spec to set where body of music resides. (0.316V for consumer, 1.23V for pro). That will give the same approx 17dB headroom as before. Easy to tell these meters, because 0dB is somewhere towards the middle of the meter's range, not up at the top ;)

Then, if SPL isn't all you want....cause those settings should be the maximum DAC or line out levels you run...(there's no output left that doesn't eat up necessary headroom)...
You either turn up the gain on the amp (which really means get rid of any inline attenuation), or get a line stage amp (preamp in home audio), or get a pro DAC.

Is a buffer board just a line stage amp; or is it that, and the ability to use both bal and unbal?
Either way, I can't see spending much at all, or adding any complexity, to what appears to be a simple case of getting gain staging correct, and matching line and amp levels. Good line amps, mixers, preamps, whatever you want to call them, are cheap. Good pro DACS aren't necessarily cheap, but I'd buy one in a heartbeat before spending much money on a line stage device.

Just my 2 cents guys...
 
Have asked Tom about mounting buffer inside power amp enviroment plus the voltage levels for conversion, so lets see what he say about it.

The big questions is always its not low cost get 3 or 4 of them at computer side of system, its 372€ plus vat for 3x raw boards then comes PSU plus box and plug terminals, so probably in € half a kilo. That Okto DAC was in € one kilo some months ago and it have four of them differential out build in, whatever analog output level per channel can be ordered and user can change it down the road solder few smd resistor with other value that set the gain level and then comes each channel or channel pair can in units display meny be set to whatever lower levels via chips digital 32bit.

I'm not saying it doesn't make sense, but that Octo DAC would leave me with two questions.
- Feeding it an input signal through USB, I have had problems with PC noise with 2 generations PC's ago and using a different amplifier than I have now.
- I don't know what it sounds like, and I do know I can be happy with the M1 (for now) as that has been tested and tried.

With meters for each of the stages, it's pretty easy to set optimum gain staging, and solve level matching issues.

First, a meter for the DAC, and set the DAC to pass peak signal just below clipping, 0dBFS.

Then, a meter for the analog output of the DAC, set to where the body of the signal resides at about -16 to -18dBFS, if using a 'digital scale' meter.
This is a good level for both consumer and pro DACs, as of course both need the same headroom.

If the DAC output analog meters reference voltage and not dBFS,
use their nominal voltage output spec to set where body of music resides. (0.316V for consumer, 1.23V for pro). That will give the same approx 17dB headroom as before. Easy to tell these meters, because 0dB is somewhere towards the middle of the meter's range, not up at the top ;)

Then, if SPL isn't all you want....cause those settings should be the maximum DAC or line out levels you run...(there's no output left that doesn't eat up necessary headroom)...
You either turn up the gain on the amp (which really means get rid of any inline attenuation), or get a line stage amp (preamp in home audio), or get a pro DAC.

Is a buffer board just a line stage amp; or is it that, and the ability to use both bal and unbal?
Either way, I can't see spending much at all, or adding any complexity, to what appears to be a simple case of getting gain staging correct, and matching line and amp levels. Good line amps, mixers, preamps, whatever you want to call them, are cheap. Good pro DACS aren't necessarily cheap, but I'd buy one in a heartbeat before spending much money on a line stage device.

Just my 2 cents guys...

What is much money, it is all relative in the Audio world, isn't it? :)

The line driver we talk about here can be configured with gain, basically acting as a pré-amp. As I've heard one of Tom's devises do this job (The HP-1) and do it real well, that still is a no brainer, I put trust in his products and do not burn many bridges here in the proces.

I can buy the 3 I need right now, if I decide later on to switch to a pro DAC in the future I can re-use two of them.

One could build a killer pré-amp with these boards... Which is something that's also on my mind. As that could give me an option for introducing an analog 6 channel volume control to avoid using digital attenuation If I would choose that path.
I don't see it as needlessly complex because I've started with (all) home gear here. If that Goldmund hadn't impressed me as much as it did there would have been a big chance that I would have moved towards Tom's MOD686.

I still see that Goldmund as a pretty crazy amp, powerful in its simplicity and philosophy.

If I start anywhere, which I'm sure to do, the first step is probably known now. It may not end there :).

Unless I would get an Okto in the mail to try/see if I like it, it simply is too big of a gamble for me despite it's crazy good numbers. That DAC8 Stereo Okto is out of this world crazy :D.
 
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