XXHighEnd

What you see below first are a few pictures of the switch board at the back of the speaker which allows to set Fletcher-Munson curves. Here you see it in lab setup to determine the proper resistor values etc. before all goes on to a PCB built in the inside of the speaker.

A few examples of the settings (27 in total) :

(http://www.stordiau.nl/orelo/IMG_0825a.JPG)This is a flat curve and all is "green". On the graph below you can see what it does and it resembles this :
(http://www.stordiau.nl/orelo/Fletcher-Munson01.png)
As you can see per "loudness" the curves differ, and we need to look at the 80dBSPL / 90dBSPL curves which assumably resemble our playback levels.

Now, the green-green-green curve is just a flat line and normally loudspeakers will be tuned to such a straight line. The Orelino and Orelo do too, but only if you set it like that. ;)

These are snapshots of the picture above :

(http://www.stordiau.nl/orelo/Fletcher-Munson03.png)
(http://www.stordiau.nl/orelo/Fletcher-Munson02.png)
and it shows from the shadows on the dip switches how they are set. Also look at the arrow directions next to the switches;

Now think : While the normal loudspeaker will be tuned to green-green-green and the F-M (Fletcher-Munson) curves deviate from the horizontal line partly to under that and partly to above that, the deviation towards that F-M direction will be other than green. Look :

(http://www.stordiau.nl/orelo/IMG_0824a.JPG)
Here you see the switch for the (right) part above the horizontal line switched upwards (higher) and now it is orange.

(http://www.stordiau.nl/orelo/IMG_0826a.JPG)
Here we switched the left and middle part more "into F-M" and both switches are now in their middle position (orange). The right part is green again and the switch is in its most down position.
(the middle led for the right part (normally orange) catches the light of the bottom led in this lab setup).
 
(http://www.stordiau.nl/orelo/IMG_0827a.JPG)
And here you see that we set the middle part all into the red, or IOW into the most extreme F-M setting (again see the switch settings by means of the shadows).

This one again :

(http://www.stordiau.nl/orelo/Fletcher-Munson02.png)
but now look at the texts above the switches;
From left to right the switches are named Slope, Dip and High. And with this one repeated you see it happen :

(http://www.stordiau.nl/orelo/Fletcher-Munson04.png)
So there's a slope from 600Hz (look at the mouse pointer !) and which "600Hz" is also denoted under the switches :

(http://www.stordiau.nl/orelo/Fletcher-Munson05.png)
So see under the left hand switch. But what you also see is that this "Slope" is controlled up to 6800Hz (6K8). See the mouse arrow for that postiion :

(http://www.stordiau.nl/orelo/Fletcher-Munson06.png)
Now, the second switch controlls the Dip, and it reaches from 1200Hz (1K2) to 5000Hz (5K). Need to show another F-M plot with arrow pointers ? no ! because surely you got the idea by now.
Notice : The Dip as such is "in" the Slope, so when the slope is set more downwards the Dip automatically goes along with it. However, since the Dip is a separate setting just the same, that Dip setting determines the deepness of it. More below you will understand when we start showing measurements of this.

The rightmost switch controls the "highs" and therefore it is named "High". Its working range is from 6800Hz (6K8) and the 6800Hz is a lever point (you'll again understand this from below measurement plots).
It is no coincidence that the beginning of its working area matches the end of the Slope's working area because this too you see in the F-M curves might you look at the different levels of loudness.

Remark : If you again look at the F-M plots, you see that there's also working area for that under 600Hz. And if you look more close to our listening levels (the 80-90dBSPL) you see that nothing much happens under 600Hz up to 200Hz. From there (towards the left) things start to change again. And it is there where we can adjust all with DSP (Digital Signal Processing) ...

---

While this may all look nice to you, it obviously has a purpose. And this goes beyond the Fletcher-Munson ideas;
So notice that the idea of these two blokes (back in 1938 or so) was about our hearing system (mechanisms) and how we human are more receptable to certain areas of the frequency spectrum than others. For example, the "Dip" area consists of "danger" noises we humans from origine receive better (louder) than others. A snap of a branche for example. So, while we perceive that more loud, it may jump out in music when equally (for level) presented compared to the other audible frequencies. It may sound nasty ...
(Notice : with not too many words this is not very well solvable if music is to be sounding less "nasty", but still ...)
So what happens with the adjustable F-M curves is that any louder perceived frequency range is attenuated and the result of that as how we perceive it is now a flat response.

What we, as the owner of this loudspeaker can do with it goes beyond what F-M intended with their curves;
It occurred to me that for example the difference between Windows 7 and Windows 8 can very well be attacked with it. So, too much of uncoloured highs ? then ... put in some Dip. So yes, you might expect that some High should be "muffled" out, but it does not work like that. Not for the frequencies what this is all about.

People might think that this is all the expensive way of applying equalizers, but that is not so; first off it all happens in the high voltage domain (read : at loudspeaker levels) which in this case is a necessity and is all related to the sensitivity of the speakers (118dB). Meanwhile (with this as a given), the crossovers all stay in-tact. Not an easy task to accomplish but obviously the most crucial.
Foremost though, we don't want to have our sound touched by other means than "normal" (which is passive filtering in this case), assumed no DSP is possible at this moment, but while the speaker fully anticipates on that (future upgrade) and now our "switchboard" is even more complicated than it already was because it anticipates external amplifiers.
Can't follow ? it's not all that important for now ...

---

So then let's now start measuring this.

---

All measurements were performed at 88dBSPL. Is this important ? Well, actually Yes *if* we think the Fletcher-Munson curves really mean a thing (they are different for output levels).

(http://www.stordiau.nl/orelo/Orelo-MKII-Slopes.png)
What you see here is the Slope with all its three positions. Green, Orange and Red.
The dB scale is not shown but it is 5dB per vertical division.
In communication Bert and me got used to S(lope), D(ip) and H(high) positions. Above you see mixed in one picture :
S=1,D=1,H=1
S=2,D=1,H=1
S=3,D=1,H=1
So each switch has its default at position 1, the middle position at 2 and the extreme position at 3.


(http://www.stordiau.nl/orelo/Orelo-MKII-Dips.png)
S=1,D=1,H=1
S=1,D=2,H=1
S=1,D=3,H=1
Got it ?

Varying the the Dip means varying the "presence" (empirically found).


(http://www.stordiau.nl/orelo/Orelo-MKII-Highs.png)
S=1,D=1,H=1
S=1,D=1,H=2
S=1,D=1,H=3

If you payed attention during the introduction you will have notice with these measurements that the "green" for High does not comply to the promised description;
The reason is that the tweeter really needs burning in and now the fun : no problem during the burn-in period because you can just flip a switch for it. This really matters because the horn for the Orelino speaker I used for the past 7 monts really shows more tweeter output by now, and it is audible too.


(http://www.stordiau.nl/orelo/Orelo-MKII-Extremes.png)

Here the both extremes are set against each other;
S=1,D=1,H=1
S=3,D=3,H=3

The red line will make older mor bad recording (could be Rock) sound less grey while still detail is picked up and which makes them sound fresh. It is not the best for good balance, but you may like the detail coming forward which can sound really "on-par".


(http://www.stordiau.nl/orelo/Orelo-MKII-AllMid.png)
S=1,D=1,H=1
S=2,D=2,H=2


(http://www.stordiau.nl/orelo/Orelo-MKII-S2-D3-H2.png)
S=1,D=1,H=1
S=2,D=2,H=2
S=2,D=3,H=2

You can also see this as the difference with the previous picture, where the red curve shows the added Dip.


(http://www.stordiau.nl/orelo/Orelo-MKII-S2-D3-H3.png)
S=1,D=1,H=1
S=2,D=2,H=2
S=2,D=3,H=3

Same as the previous picture but now also the High is set in extreme position (3).


So these were 7 pictures which showed some of the combinations and you will see (hear) that most of them really make sense when it were about some means of wanted perception. Without telling the combined settings, you can expect this (of course each one mentioned also has its opposite) :

- Dial in presence.
- Dial in spaciousness;
- Dial out harshness;
- Dial in über-detail;
- Dial in Mid-bass;
- Dial in highs-colour.

With my own experience from the past 7 months, a lot of music really can be played with (for these settings) in order to get so much more out of it than "normal". So think of music so well recorded that you can set H=3 and all it does is create a very wide stage and sharpened detail while nothing starts to hiss. Things start to fly in the room. It could be your genre of the night and all it takes is setting some switches.

What crucial about it (to let it fluently work) is the levers you see; one at around 400Hz and one at around 7100Hz. So this keeps on connecting the frequency spectrum in a responsible fashion. And for example, setting up a Dip can create more highs. Yep. So what you will be doing in that case is making the 7KHz region more profound. You do this by making the - actually more profound by nature - 4KHz-5Khz *less* profound, so it allows the 7KHz to be heard. Just look at the last picture to understand this.
Similar can happen with making Mid-bass more profound, assumed this goes from 200Hz to 400Hz or so. Just set S=3 (and see the red line in the last picture).

All this is outside of DSP which operates on the bass (up to 400Hz, but can be more for the real tweakers). This is for another topic.

Peter

I remeasured everyhing (don't ask, but I made a stupid mistake as I found out) and I put new screenshots in the original post above. You can now better see how the tweeter extends with the notice that it still needs burning in so it drops off somewhat (~2.5dB at 21.5KHz for the intended straight green) line.
Maybe you need to apply (Ctrl-)F5 to your browser to see the new screenshots (the old screenshots extended to 15.2KHz, the new to 21.5KHz).

Peter

Impressive!

Am I the only one thinking of three new buttons in XXHigh end?

 :) :) :)

regards,
Coen

Exactlly what I was thinking Coen
And have you looked at the curves for the bass part. About 34 db higher at 16.5 hz compared to 1000 hz. That will be tough for your amp...
Leo

Hi Leo,

The BD design eq is within the horizontal 5dB line spacing of Peter's graphs. I see no eq below 600 Hz.... The only tough one would be the high, since that is a 'gain' and we need (imho) to avoid clipping.

Regards,  coen

Hi there Leo,


Coen is just ahead of me, but even after reading this 10 times I can't understand what you really meant to say !

Peter

Coen, surprisingly it is more smart than that, and it is not extra gained as you suggest. That is, not within the one powered frequency range (like 400Hz up to 22KHz could be one frequency range to power and one band in there is boosted).

I already see it is hard to explain without explaining all (I tried hard on the latter :)), because some things sure will be kept in secrecy.

;)
Peter

Probably my mistake, but:
If I read it correctly you take the FM curves for the correction for perceived loudness. The original FM curves and the present form ISO203-2013  go way up for lower frequencies. About 6db up starting from 1000 hz up to +34dB for 16,5 Hz.

Gain, attenuation whatever you use, I still see only moderate corrections that can be done digitally without a big loss of bits.

If the passive components are more 'transparent' than the 1 bit digital eq loss (for selected frequencies) the hassle would make sense. And of course for the speaker users that have no interest in computer audio/XXHE/dsp.

Anyway it is a perfect candidate for another unique feature of the XX player!

regards, Coen 

Oh, Coen, maybe only now I understand what you were saying. Including the clipping.

Well Yes. And so this is how the speaker contains that other dimension (in electronics) : how to switch off all the passive filtering when done from DSP. But this too is for an upcoming topic. :yes:

Peter

I prepared this somewhat longer post longer ago, but never got time to finish it until today.


Aha ...

I have talked about this previously (in the Ambience topic I think) where I said that first of all the modern version of the F-M curves you refer to are not correct. Well, they just can-not-be;

Maybe this gives me the opportunity to explain how the F-M curves for the Orelino/Orelo MKII emerged in the first place :
When I was finished with my "tuning" of the Orelino and which was based on my own filter I used for many years with the same top horn (above 270Hz there) it appeared I was using the F-M curves. I didn't know that, and while I just told one of the due customers of the MKII some numbers, he came back with "but that's exactly Fletcher-Munson !".
Coincidence ?
Can't be eh ?

One thing : For the bass this is not true. No-way.
But why ?

I tried to sort that out with the same guy who came up with the "That's F-M !" and we need to be speculative ...
Notice : I forgot the sequence of things somewhat, but this is not so important;

Say that back in the days (1938) headphones where used. That's cool, but I don't see the really low frequencies come out of there while at the same time the official SPL (Sound Pressure Level) is to be mapped on to that. Maybe it can be done, but I'm sure we all feel that it won't be reality much for a one square inch driver to move air like in real open air (with the notice that headphones are closed and which does a few things too).

Say that in 2013 or whatever recent, normal loudspeakers were used, then I want to see which and how the real lower frequencies were squeezed out because already something will have gone wrong on the distortion levels. On this part : Try to create a threshold of pain (see official F-M curves) for 20Hz without not first implying 40Hz for a 2nd harmonic. It is to-tal-ly impossible - I tell you. Now I'd say that whoever applied that test(s) will have known about 20Hz not being allowed to turn into 40Hz because it is so obviously audible but *if* it did, it would blow out the windows because of that 40Hz, while 20Hz only will shake the room.
So what the heck did they really do ?

I know, and I guess I am amongst the very very few, what happens when a fair amount of SPL is put out distortionless. So, this is about 20Hz not being audible at all, while the meter still reads (the reference) 88dBSPL. Same for 23Hz (also not audible), same for 19Hz. Also same for 18Hz and beyond (I stopped at 8Hz), but from there (18Hz) the output first has to be squeezed or otherwise the 18Hz (and beyond) becomes audible (thus too much distortion). Now :

What is not obvious at all is how a normal speaker will just not put out anything much onder 27Hz because of the, say, acoustical roll off (mechanical also OK). Notice that in normal practice the limit is 100Hz for a fine woofer.
Now some smart guys thought that pressure chambers and ports and what not could help that. Well, let's say that is correct. The same smart guys thought that rooms would help too. This is already more doubtful for me, but alas.
But it does not work. It does not work because you can easily measure it is not true if you only pay attention. Actually it is quite simple : Grab such a fine speaker with specs like "straight to 27Hz !" and measure. It just won't happen. And oh, you will perceive a nice bass alright, but it is at way lower output than e.g. 100Hz (say 10dB for the discussion).

Now we buy speakers according those specs. We also *use* them according those specs. And if you now don't pay enough attention, before you know it you use them to apply such tests. And what happens ? you need to apply 10dB more "power" to them in order to hear the same level (level of pain or whatever level).

So what am I saying ? well, that this speaker 100% really goes straight to 19Hz and if I now apply 10dB more to 27Hz it already goes very very bad and if I'd apply 34dB or whatever (F-M) more to 20Hz, well, ... do we know how much 34dB more is in that low regions ? So, all tuned nicely in balance and then add 34dB there ? I'd say you would die.

Even if I am half-correct only, it is completely clear that those tests for the lower regions must have totally failed. So I am 50% correct but test still failed 100%. It is just too obvious.

Here is another kind of proof, but from another angle :
So we have subwoofers (like I had them). 700W Hypex in there and such, and the 700W still is critical when fed with line level (same line level the MKII etc. is fed with). So how is it possible that e.g. the MKII need something like 40 Watts only ?
This seems directly related to efficiency, but this really can not be all. I mean, there's one tiny 12" woofer in that sub, and although the driver may not be as efficient it just can not be so that it needs to wildly excurt and vibrate the cabinet while our 3x15" hardly move. Right ?
So in my view, all that cabinetted speakers have just fighting-itself drivers for the chamber pressure they create and which works counter productive. So much so that not even real waves come out (distorted ones yes). From there can be reasoned that the frequency gets lower (a sort of sub-harmonics) but not at the same SPL as a "full wave" (sine) would do it nicely.
How this measures in comparison I don't know (I never tried) but I do know that standing waves form easily and nothing will be good.

There is some more to it, like lower frequency waves spreading more easily (are not directional) from which can be calculated that what you perceive from it should be 1/4 easily compared to when it would have been directional. A microphone would not capture that differently. But outside (anechoically) would be different from inside, because inside reflections play a role and add again.
Now how come that when I tune the lot with the microphone from a fair distance (few meters) the bass will blast the windows out again when tuned so. This just won't go. So it has to be done the other way around : from way close. And now all is in balance.
For net result at listening distance the bass rolls off heavily while I think that everybody would expect it the other way around.

And so it has to be my conclusion that the F-M curves for the low spectrum are the exact other way around; we humans should be very much more sensitive to the low frequencies than e.g. 1KHz. But I think I can tell you that all is related to how well these frequencies are formed and how an e.g. 32Hz sine is really that instead of a wobbling thing with spread energy (!) to lower and higher frequencies than the 32Hz intended.

Sorry to be so fuzzy but where science is lacking somewhat we need to start expressing empirical finding. That can easily fail and it is only my thinking as of now. Those receiving the speaker soon : you will see. It is a whole new world ...

Peter

Hello Peter,

That makes a lot of sense (instead of nonsense science with weak bass speakers)!

Thank you!

Leo

Interesting the old Fletcher-Munson curves.

For those interested in some background information, read http://www.lindos.co.uk/cgi-bin/FlexiData.cgi?SOURCE=Articles&VIEW=full&id=2.

The tests were repeated by Robinson-Dadson in the fifties and more recent for the ISO-226 standard. The discrepancies indeed appear in the low frequencies.

Regards,
Stanley

Edit: also see http://www.lindos.co.uk/test_and_measurement/SOURCE=Articles/SOURCE=Articles|VIEW=full|id=10

and

http://www.lindos.co.uk/test_and_measurement/SOURCE=Articles/SOURCE=Articles|VIEW=full|id=17

So,
I've been playing with these curves for a while, and they are indeed very very interesting! Some of the design concepts Peter/Bert came up with make this design a landmark in speaker technology, and that I would not say lightly.

Since these speakers play "realistically" one can just use real music as a guide. My experience so far is that these curves help compensating for all the small changes from recording process. One can see how a poor recording was done, or how a good recording/mixing artist shines.

This is quite apparent, when plays a particular genre of music for a while and adjusts the FM curves, forgets about the curves. Let's say play electronic music, FM curves are adjusted for spaciousness, and then switch over to a live concert, suddenly things sound not quite right, image density very poor. Tighten up the curves, and things fall into place.
Play classical music for a while, then switch to a the girl with the guitar, suddenly girl seems out of size (not that it's whacked out or anything, but not real), switch curves to change image size, back to authentic proportion. There are many smaller paths one can take, right size, push the image a bit back, for those too forward recordings or vice versa.

In many ways one can get an idea of what went on with the recording/mixing. Maybe even comment on the speaker he/she used for playback, because certain things were not solved or heard, and let thro?

Many of the things Peter talks about "dialing in" I'm yet to experience with.

It even might be used to compensate for some hearing loss.

All in all, it's is refreshing Peter and Bert did not just with the "flat" response, but gave an option to manipulate the sound for ones own benefit.

 :yes: :yes:

Hey VJ, nice that it works out !

Ok, let's unveil a small secret;

VJ (as how Mr Rao prefers to be called) knows a bit more about our auditorial system than the average us. It is actually his profession.

It was him whom I invited to do a nice seminar in the US about these things, mainly knowing about how my own perception of this all (think "Phasure" name for now) would nicely match the human anatomy VJ knows all about. Envision an email from me to MD VJ just asking (and I did not know him other than from being a NOS1 customer).
We prepared for the seminar totally indepently from each other. And it was fun (we recorded it all and should upload to YouTube).

At some stage VJ ordered the MKII's.

After that I pointed out to VJ what my personal filter settings were. So, on the Orelino of back then (say October 2013) I had my own longer ago "set" filter response and I explained it by words.
VJ came back to me and said "but that is Fletcher-Munson curves !"

Here "MD's theories" started to match practice beyond ideas put forward on a nice seminar and what I did was making that F-M curves really practice. Well, Bert had to do it of course and I can guarantee you that this is NOT an easy thing for a passive filter (because this all happens in that passive domain).
VJ did not even know this ...
Or not at first, because it first had to work out and that took maybe 6 months apart from quite some hairs.

Actually all is the other way around; say that I am responsible for the filter response with the notice that this is all about one pair of ears plus maybe another few pair in the house over here, but still; once you are used to XXHighEnd "dials" you know that it can go all sorts of directions, including not yours (read : findings by others for the better). So it is quite a responsibiity to apply a not-flat response curve in a not-so-cheap speaker only because *you* like it that way.
So this is how it was made set-able and now I could not be blamed for applying some strange curve ...

That curve though appears to be "science" (I knew about it but never realized I had been doing *that*) and diving into it more it was obvious that this is not the same for everybody. Now the settings became a fairly explicit thing or better, a goodie which is not to be underestimated.

This long lead-in actually leads to one small message I wanted to put out and the reason of this post :
It is caused by VJ.

Thanks man,
Peter

Hey Peter,

Thanks for the shout!  ;)

The practicality of this is astounding. Though we know all the theories on the FM curve, it's application actually has not been performed to this level. And in practice, this actually brings in "unique" results, unlike anything on the market. There is more to be discovered! These speakers open new possibilities.

VJ

Let me reemphasize the importance of FM curves. It is nice in a way Peter/Bert have made it happen. On the fly one can change the settings, long behold, flat frequency response actually sounds bad on most recordings. Dialing it around FM helps with a more meaningful recreation of the signal. Meaning, the way these speakers play music, one can easily say somethings wrong with the recording, hum .... manipulate the curve, and yes one can recreate the recording event.
The speakers, amps FM curves and crossover work quite seamlessly. The different permutations on the FM curves makes it a very powerful tool for plaback. This would be impossible in seperate or different speaker/amp matching, where there is only one result, which is not calibrated for.

And yes, I've yet to discuss "Bass"

To some extend this FM "correction" can be done in the digital domain too. This opens up the possibility of coupling the intensity of the FM correction to the volume control position...

Not that this is my idea. Like many things we just have to look back a couple of decades for the great ideas (and poor execution). I've got old Philips tube amps form the late 50s and they all have a "physiological" volume control which intends to compensate at least the bass for FM. Actually the engineers of those days saw the tone control (Baxandall) and later the "loudness"  button as a means to correct for FM rather than to compensate for the loudspeaker tonality or disco boosts. Quite crude and ineffective measures compared to the sophistication that the Orelo provides, yet a digital solution may hold even more potential.

regards, Coen

Peter
Just checking with you about ongoing development of the NOS1(a) if you only use the Orelo MkII (with FM adjustments) for your listening tests. Effectively you will be seen to be using FM "EQ" type adjustments to make the music sound as best as you can in your listening environment at your listening levels.

While this may be a great option for the few Orelo owners, it will not necessary give you the same insight into how your NOS1 sounds compared to other DAC's using "ordinary" loudspeakers.

However, knowing how you operate, I assume you have this covered.

Cheers
Frank

Hi CoenP/XP,
You are indeed correct, there are many ways to skin the cat.
However the direction Peter took makes in unique, since it is accurately calibrated on the speaker, it is reproducible.
Otherwise one has to work with the specific transducer, it's characteristics, crossovers, speaker design, and the results may vary so much that it might might not make sense. There is much to add to this, all in due time. :)
Kind regards,
VJ

Hi there Frank,

I think it is the other way around. I't like I said two days back : with the again better speaker I can hear better what can be improved. Notice though that I put that a bit differently : "I needed to improve the NOS1" and "after that happened I needed to improve the Arc Prediction Filter". So improve one and hear how the next now needs improvement.

I know how unreal this may come across but it is also about the level we're at these days. Like VJ said "you can only compare with real music(al instruments)" (similar). Or how it was clear right from the start with the NOS1 that not only me, but people in general could far more easily hear where the next culprit is (assumed there is one and there always is one of course). But all is related to a reference and in this case that reference could use some upgrade. It is just audible but needs experience of course (way much actually).

And so in my view everybody will benefit. A better DAC is a better DAC. But thinking in the same realm, now people may be able to hear (better) that an amp needs an upgrade. And that's not a bad thing per se. Usually the only probem is "how do I tell my wife", right ?
:)
Peter


PS: That last remark can come across as sour to some, so I know that. But in this hobby I think it is the general idea ...

Hi VeeJay,

Thanks for the reply. Looking forward in quiet anticipation!

regards, Coen

[/quote]
I think it is the other way around. I't like I said two days back : with the again better speaker I can hear better what can be improved. Notice though that I put that a bit differently : "I needed to improve the NOS1" and "after that happened I needed to improve the Arc Prediction Filter". So improve one and hear how the next now needs improvement.
[/quote]

Peter, Perhaps I am not expressing myself well. And no criticism is intended in my questioning.

I have no doubt that a better speaker will help you develop a better DAC. I have no doubt that the highly transparent and revealing Orelo MkII have made the evaluation of any deficiencies in the NOS1 easier to identify and help you in your quest. I have no doubt you are achieving extraordinary SQ results through the Orelo MII speakers with the NOS1a. Absolutely trust you there!

However, I still have a question relating to 'balance and neutrality' for the DAC.
Take this example (made up case only): How do you know you are achieving the right bass weight/drive from the DAC, if you are using speakers, no matter how revealing and transparent, that are themselves not voiced neutrally?  That is, a perceived bass lightness (no matter how detailed and dynamic) from the DAC could be compensated by (and potentially masked by) FM adjustments that lift the Orelo's bass response. The result, a fabulously weighted & dynamic bass presentation in your room at your critical evaluation listening levels.
Nevertheless, your extraordinarily revealing DAC might be perceived by others to be bass light with other speakers/systems - no matter how detailed/revealing the DAC.

Hence my question about you using another recognised "neutral" speaker in your development evaluations for cross-checking the NOS1a's performance against other DAC's.

Final comparison: Your NOS1a/Amps/Orelo MkII "System" might give the best "system performance & quality" available anywhere. That doesn't automatically guarantee that you will have voiced the NOS1a DAC to be recognised as providing the best "DAC performance & quality" available anywhere. You probably have achieved that though, and I look forward to future feedback from the many new NOS1a users - without Orelo MKII speakers.

Cheers
Frank

Hey Frank,

I suppose I have an answer to that too, but it really needs some thorough work out and of course longer post.
So I'll be back !

Regads,
Peter


PS: Question is fully justified of course !

Frank, an attempt :

As you may have read (topic elsewhere) I even dare talk about a neutral speaker. I recall even adding "100%" to that. Yes, super dangerous and maybe not realistic to begin with.

You are one of the happy few that sort of dares to pose the question; as you will know, elsewhere this goes differently and some (often extensive) justification is needed. Not much in my own forum. But I guess this is because of that other happy more "believe" better to begin with because they hear it (coming) themselves. So elsewhere I have to explain a 1000 times that any DAC serving the frequency domain is killing the time domain while it is all about that (killing). And no matter you should have been the very very first with a NOS1, it never happened and now you can't "see".
Side note : If anyone with a NOS1 disagrees the slightest, let him speak up. No problem !

So my point is : The DAC is neutral and with a bit of luck "100%". It also measures that (all *is* about measuring) and the amps used do as well. So no reason to not tune a speaker to that while a speaker is not much inherently neutral at all. But the THD figures also easily show that ...

Side note again : I know of no amps that measure neutral as well, but one (the Sauermann). Of course I haven't seen all, but it may be an indication that all what we listen through is not neutral for measurements to begin with.

This was the easy part; The FAR more difficult part is how to explain that a DAC which is regarded completely neutral world wide (this just is so and otherwise waiting for the first to come up with some sort of "complaint" regarding this) ... how such a DAC can be improved upon. So just telling : I am not good at that because it requieres wording which may not be in my vocubalaire. But I think "timbre". Now, timbre IMO will hardly change while in the mean time jitter causes all sorts of "anomalies" elsewhere. Like sound being too harsh (after all and after all we try with XXHighEnd for example). We could say that the more harsh sound implies a nature to the sound, but we don't recognize it as that. We only do once the culprit is out of the way. Like W7 was so good. But not any more since finally W8 started to be good. Get what I mean ?

So a typical example of less jitter (for me) would be "more real again". Timbre did not change, harshness probably got better, but things play together better. But is still neutral.

Before I go too wild on this, it now is crucial to understand that a combo DAC/Amp - but actually also playback software, can be regarded as neutral, or at least neutral enough to right away hear where a speaker (last part in the chain left that can influence) is not neutral at all. This really is easy.

The last part of the story is that when a speaker changes from 115dB sensitive to 118dB sensitive, this implies such a huge change on the HF output (all just flows out so much more easy) that while this with 115dB was hold back, it now does less and e.g. harshness becomes more apparent. It is just louder.
Remember : A piano plays at 90dBSPL live and a cymbal does 110dB. With an 80dB sensitive speaker the cymbal may sound at 60dB while the piano still plays at 90dB (this could be exaggerated). But when the speaker gets more senstive (all moves more easily) the distance of the piano and the cymbal gets larger; the cymbal travels towards reality level outputs (and let's be happy when it never happens really - a cymbal is LOUD). Last thing to understand - which is not difficult - is that when the cymbal sounds louder the implied distortions from the rest of the chain also sound louder. This has been the problem forever (for me). So what can be wrong when things now don't sound right ? the speaker ? no, in this realm not. So it is the remainder. DAC is first. Why ? because we KNOW it is subject to all the noise sh*t we ever so much talk about in this forum. So let's solve that.
Done ? now it becomes apparent that serving the time domain for 100% is again nog te best for hearing distortions. Why *this* conclusion ? well, it can't be the PC any more, it can't be the playback software, so ot has to be the filtering (which is in the playback software, but in another dimension of it).

It is almost simple math. So :


Yes, and sadly it is just the truth. No bass can be too light when it measures well. But also and merely : No bass can be extended when the speaker can't do it. Read that topic/posts about me explaining what woofer surface must be in order to represent a kick drum for reality ? That. It can't be done otherwise unless with super heavy distortion (which is in all of our speakers) and so this too is easy.
So NOS1 *is* bass light because speakers are. Change that into a normal DAC and ... NOT neutral. Bass-light is also not neutral but personal me looks through that. Again remember W7 ? more bass. who can stand it ? nobody Why ? because now it distorts.

Still too short for all the real answers (and many typos I'm afraid) but this is how I "work".
All is also how I dare to call the MKII neutral, because my ears tune it like that in the first place (this is the DSP part and btw not the F-M Curves) BUT I also measure THD. And *that* now is crucial.

Sorry if this is all too confusing - I know it is without the implied references. But what to do.
Best regards,
Peter

There is more involved:

http://stereos.about.com/od/stereoscience/a/amppower.htm
http://electronics.howstuffworks.com/how-to-explain-speaker-sensitivity.htm

People,

I don't even dare to read back, but I'm sure that last post must be quite "odd" to say it mildly. I wanted to do it later but saw 5 minutes od spare time yesterday and I wrote that in really 5 minutes. It *is* what's in my mind though, but this was shouting from a high mountain to you. Too fast.

I will still keep it brief now, but I'll try another way.

DAC is still neutral. It was created on totally flat measuring amplifiers (look at noise line) and the process of "tuning" the speakers I used back at the time is a process of years and learning what comes from the speaker and what from the DAC. But, DAC is also measured all the time and this is with the focus on the time domain - no ringing. This is crucial because all what rings also colours (not neutral). What remains from anomalies in the frequency domain can be harshness. This expresses better when the speakers are more fast. The more sensitive, the more fast they are. DAC is as fast as possible (this is time domain stuff again).

When this is all in balance all should sound neutral. Neutral is : Play for a week all types of different music and hear no similarities all over. This is my main tuning means, also with XXHighEnd changes.

It is NOT so that e.g. too much bass in a speaker is explictly compensated by e.g. XXHighEnd means. Also not DAC means. This is just not possible because of the pure technical means used for DAC and software. These means for example "as fast as possible" which often is a deriviate of as lean as possible. If now more or less bass springs from the speaker, so be it. But please remember, it is YOU ALL who determine this. Me only as a very small part. So if at some stage one says "better bass" you can almost bet everything on it that all will say better bass. This is how "better" is determined. One big A-B company.

I am not tweaking my NOS1 all the time; I keep it the same as yours. This is on purpose or all would be moot.

With my reference of neutral, which is all confirmed by you out there, I may proceed. So you don't complain and I don't either. Now, new speaker. Does not sound neutral at all and why would it. It wasn't tuned for that. But I know the rest of my gear and that it is able to do it.
... now envision the process ...

Nothing works, but the speaker is quite similar to what I had and the filtering in there which has been a process of years can be applied to the new speaker. Aha, that works. But only as a base. There's lots more of tuning possibilities so in many months time it starts to be good. Better.

While the above can be seen as all audibly tuning, now measurement comes in. This is related to it being a commercial speaker and no "my room" or "my ears" are allowed to play a role. So measurements explicitly to not incorporate room response. It becomes a very technical thing now.

Major part of this is how the bass response is tuned, which merely means "how it is measured". That needed something odd but in combination with listening to the resulting tuning I found a way how to measure so that what I measure for the better also sounds for the better. However, this is super simple because it is not about tweaking out anomalies - it is only about tweaking *in* all for the better. Like how to go straight to 19Hz. One thing : the whole bass section has to match the mid and *that* only goes audibly. Why ? because they can't be measured together which in itself is related to not wanting to measure the room.
Now another technicallity was applied : making the XOver so that the bass section allows for linear volume up/down while the XOver keeps on "working".

This latter tuning of the bass against the mid(/high) also takes weeks. That ends with maybe once in 3 days another 1dB more and later one half less and more again, until nothing disturbs any more and all remains neutral. Remember : nothing sounding the same. Nothing bass heavy, nothing too lean. What remains is that one recoding just *contains* more bass than the other, but no way you have the idea to tune in more bass. It is just not in there and you know it.

Then the next round comes and it is about knowing that the speaker is OK while you can also hear it can sound better. Nothing much different from sitting down for another XXHighEnd tweak with one difference : I don't know of any more XXHighEnd tweaks at this moment. So what to do ? improve the DAC because we all KNOW it is there why and how XXHighEnd can influence. So eliminate that influence finally.
And I could even do it.

If *now* something sounds wrong and still knowing it can't be the DAC, it must be the filtering. And easy, filtering makes the sound and is the reason why the NOS1 sounds as it does in the first place. So it is the last thing to improve upon, thus now that.


I said the same as yesterday but now maybe in a less shouting form and is is the truth.
It "just" can work like that but remember it is actually a process of many many years *why* it can now work like that. And it becomes more and more easy.

To hopefuly better answer the question which was key I think : No DAC is made more or less bass heavy. It just does not work like that. It is only made "better" with assumptions like more speed = better and less jitter = better. Stupidly that.
Speaker almost the same; find out what THD is audible plus find out a means how to check that (all written out somewhere) and make it so that THD stays under that. Super technical things and nothing subjective. The only subjectivity in it is that I say it should work like that. This is for example different from someone who says "but it should be tubes".

And in the end all is based upon not attacking distortion with more distortion (which smears the first) but getting ALL right.
I did not say it is easy and I especially did not say it is easy to understand.

Regards,
Peter

I am measuring the first production unit of the NOS1a anyway, so this is what I mean with a straight noise line.

(http://www.stordiau.nl/Phasure NOS1/Phasure NOS1a Noise.png)
In this (NOS1a) case also without any anomaly anywhere. Only some 50Hz mains pickup.
USB is streaming here.

An amplifier should add nothing to this (now new peaks); only gain.

This could not be done before and what it should tell us is that this now is a reference. Meaning : When any next NOS1a shows something here, it is not right. Simply : because it can be done apparently.
And FYI : Yesterday this was not the case (didn't make screenshots of it). So *now* pure theory comes about : It should be like this because of the total isolation and when all soldering is right nothing can show. Of course it takes more like the components used and PCB layout etc., but we shouldn't underestimate the reference this now brings. I know, it is a technical reference but in the end all is; My previous posts were all about references although from different angle and in different realms/domains. But once you have such a reference it will be the base for the next development. So for example, a next development may bring down the noise more. Nice. But now peaks show again. And thus the next development will be about bringing down those peaks. And so on.

Low noise so what ? well, think about how 24 bits can resolve. This can only be done when first the noise is lower than that (-144dB).

And so we continue ...
Peter

Playing with these speakers for about a week, I'm getting more and more acclimatized to its capabilities. I can probably see where the NOS-1a can enhance playback, there is a very small margin, ;-) (but probably an important milestone, considering Peter's response).
Mani, let me know your thoughts when you have your NOS-1a

I know Peter talks about cymbals all the the time, it actually took this setup to understand where he is coming from, this probably would relate to the high sensitivity. Considering cymbal clashes require enormous power requirements for ultra short period of time, I think this would be the forte of high sensitivity speakers for playback at realistic levels. Recording mistakes with enhanced hi frequency is easily seen, such as cymbals, which all of a sudden stand out in a recording, probably because the speakers used in the process did not have necessary dynamic intensity for realistic playback.

In general the SQ of these speakers must be amazing, but there must be other explanations than the high sensitivity only.

I came across a film report about Gauder Akustik, and Dr. Gauders argument about a good SQ is different.

http://www.youtube.com/watch?v=rnYa83UOvG4

Start the film at 3.30 minutes and watch ……… .

Joachim

I allowed myself to watch all of it and well, what to say.
Maybe Dr. Gauder didn'thave time enough to go into details he wanted, but this really makes no sense;

If I put a very same horn to my mouth like the top horn we use (or even the bass horn from my previous speaker) then it sounds the same like Dr. Gauder showed with his hands around his mouth. But of course nothing sounds like that in reality which is why a horn speaker is supplied with "corrections" to not let sound it like that.

His argument like "we have digital amplifiers" etc. to make clear that no high efficiency speaker is needed to today (opposed to the 1960's) is ... BS ?

Yes we need reflections (like I always say that side curtains need to be *open*) but it doesn't mean that 180 degree radiators are thus the best all the way.
So such an explanation is just 100 times too simple and only makes me think that these speakers will be 85dB sensitive and some argument had to be found.

Sorry ...
Peter

VJ,


I see you try your best to seek explanations for the unexpected. And I personally think you do very well on it ...

It will be too hard to dig out, but somewhere along the lines of the "NOS1 development topic" I came to the strict conclusion that no passive I/V (current to voltage) conversion could work because of a lack of current for the HIGH frequencies.
A year later I even forgot about it and retried something with passive I/V conversion, came to the same conclusion and ... only then thought I already came to that conclusion earlier on.

Although we all think that the most current is needed for the lower frequencies it really is the other way around and math easily shows it. So when I/V conversion is properly set up no less bass comes from it but way less highs yes. This is how the NOS1 ended up with active I/V conversion, no matter I held back from that for ... 2 years I think (passive is inherently better because of less components in the signal).

So your conclusion is correct that the more easy the high frequencies are rendered (higher efficiency) the less current is needed to do that; current which might not be available (think required speed (rise time)) - in any amplifier.

Maybe it is not known much, but the tweeter in the Orelo MKII is controlled by its own amplifier.

Peter

Adding to Peters explanation ...
I've had low sensitivity speakers, and I know their performance level.
This is for the fellow readers, amp requirements

http://www.crownaudio.com/elect-pwr-req.htm

One can spend an hour with different scenarios. Let's say 85dB speakers listening 3m away with 85dB listening level with 3dB headroom, 18W ... No biggie right, now let put cymbals with dynamic range to 115-120dB .... Then it's possibility is uncertain, until unless you have your own power generator. (Thanks to Peter, he alerted to me on this a year ago)

A particular amps power rating from my understanding is for a single note, and it will drop for complex passages. Power supply is a big limiting factor for most amps. Class D can do a better job on this with the switchmode supply, but they themselves are limited by the technology.

 Peter, I think you are spot on ... multi-amping is mandatory for realistic audio playback.

Because my previous post simplified things to some extend, notice the secondary effect :
When tweeter and mid are controlled by the same amplifier, the life is sucked out of the mid and the tweeter "does" that (the real high frequencies do).

At least that was one of the ideas behind the fourth amplifier in there.

I have heard these speakers and must say they are probably ‘best in class’.
What particularly struck me was the level of detail they produce. If you like the sound of diamond tweeters then you will probably love these with their diamond mid-range.

I have also heard the Orelo however, MKI mind you, and although I think the RC 11 is a fantastic speaker, I personally prefer the Orelo’s.

Why? Well because if you would do a blind test between live music and both speakers I’m sure you’d pick the Orelo as the one that comes closest to reproducing the live event, and in the end that’s what it is all about for me.

Although I couldn’t identify anything particular wrong with the RC 11, somehow they didn’t quite sound natural to me. When listening to live music you wouldn’t say ‘Wow, what a great level of detail!', it would just be there. With the RC11 it seems perhaps over-accentuated (like looking through binoculars), and with long listening sessions this can become fatiguing.

What I love about the Orelo/NOS1 system is its neutrality, transparency and finesse, a three dimensional soundstage, detailed accuracy, and speed & dynamics, but always in a very natural and musical way.

Obviously both speakers are great and I would be very happy to live with either one of them. At this level it also becomes a matter of taste, but for now I can’t wait to hear the Orelo MKII/NOS1a  :)

Diede 

Follow up on the the link to the video, and Peter's comment:

First: ....... I was warned about the horn shout, in speakers, and told to avoid,.... this was prior to even purchasing my second speakers (yeons ago), and that was forever engraved in my brain, and the first thing I did when these speakers came over, was... And every day it's still in the back of my mind, in every tune trying to point the issue out, not only that, the concentric horn, I though I could point out a design issue ... Alas it's not so at this "point" of time ... Seamless, well kudos to Peter/Bert.

I have/had speakers which are wide dispersion .... Without getting into the technicality (implied by this post) 360deg dispersion is a is a big issue. The sound from 0° to approximately 45° cancel each other out, on my prior speakers there was a huge drop from 5000 to 20,000Hz to approximately 20 dB. Room becomes increasingly important. The design which Peter/Bert took actually is quite brilliant, the horn is wide dispersion, but not too wide, the bass is OB, but with a horn dispersion.

Perhaps line arrays speakers may be another answer, with 180 or 120 deg cancellations can be overwhelmed by the forward wave. This would be a design challenge. All the line arrays I've heard have issues with imaging.

One not does need ambient information with exactly 180 deg as the guy in the vid mention, my experience, this was recently re-evaluated with the Oreleo MK-II, and that is not a simple statement coming from me. This "idea" needs to be reevaluated.

Secondly if we uses the amp cal from my prior link, we would see that increasing the power from 200 to 300w or to even 400w is actually irrelevant, even from 100W.

Total BS ...

 :(

http://www.phasure.com/index.php?topic=2910.0

Now it dawns to me that this are full ACTIVE  speakers, and that tells a lot and should be more highlighted. This is a different world in sound reproduction.

I heard in Berlin at Max Schlundt last year the Dutch speakers Grimm LS1, an active speaker too, and the SQ was totally different to my system and the sound was ......
incredible.
A full 3D image with an amazing resolution. I sat there with an "open mouth" and my question was, how is this possible.

The active driven speakers seems to be a superior design with an amazing result.

We do not see many reviews in the HiFi press about active speakers, but when, the critics are overwhelming positive.

Joachim

True Joachim. But this is a bit of my fault;
I had prepared another topic about the amplification but never got round to that because of waiting for better pictures than the ones I took (could take at the time). I now have these pictures (from a full production version) already for a couple of weeks but never had the time ...

Will look into that now !
:sorry:
Peter

Ok, here : Active Speakers - Why ? (http://www.phasure.com/index.php?topic=2991.0)