XXHighEnd

I've just taken receipt of a pair of Sauermann class-A monoblocks.

You think you know what an amplifier should sound like? Well I did, until Gerd Sauermann (in person!) brought a pair of these over from Germany to my place in the UK.

My following posts in this thread will no doubt come across as unadulterated advertising for Sauermann. And you know what? I couldn't give a damn. When someone follows their passion and produces something like this (like someone else we all know), it needs to be shouted about. Loudly.

More later...

(Meanwhile you can read more here: http://www.sauermann-audio.eu/en/technologie.html)

Mani.

PS. Gerd, if you read this post, I hope you made it back to Germany OK.

You know what ? I don't either.
Don't blame me for the results though. Haha.

Now, tell us more !

Intriguing this amp looks to be somewhat along the lines of the First Watt F1 which provides current to the speaker in proportion to the voltage input (and so the amp delivers the required current in to the speaker regardless of what impedance it sees and impedance also changes dynamically). But it looks as though it goes further as the F1 has special requirements of the speaker (particularly the crossover) to work properly.

http://www.firstwatt.com/f1.html 

Looking forward to more news with anticipation.

P

Hi Paul, my electronics knowledge is hazy at best but I agree, there seems to be some similarity between the two approaches. This isn't surprising as Gerd Sauermann is quite familiar with the designs of Nelson Pass.

Also, if I draw from my own experience of owning and listening to many different amplifiers, the closest I've gotten in the past to the sound I'm getting now is with my old Pass Labs Aleph 4 (now sadly no longer with me).

Mani.

The Sauermann shouldn't work well with my Quads. So how come it seems to?

Well firstly, let's consider its bass. It's fuller and more extended than I've ever heard from my large Quads (the 2905s). What's interesting is that there is a 'family resemblance' to the bass with my old Pass Aleph 4 amp and also my Berning 300B amp. If we look at the 'on paper' power outputs at 4 Ohms (the impedance of the Quads below 100Hz) of these class-A amps, they're approx the following:

- Pass Aleph 4: 100W
- Sauermann: 45W
- Berning 300B: 6W

Could it be that this 'full and extended' bass is actually due to distortion (harmonic, or whatever else)? I mean, how could these amps be better than my Sanders Magtech amp (900W into 4 Ohms) in the bass? If this 'full and extended bass' is caused by distortion of some kind, I don't understand why it remains, even at low listening volumes.

My hypothesis is that it has something to do with an amplifier's ability to 'keep phase'. In order of bass quality, I would put these three amps in this order:

1. Sauermann: single gain stage, zero feedback
2. Pass Aleph 4: double gain stage, no global feedback
3. Berning 300B: triple gain stage, zero feedback

And I would contend that all three amps have better 'quality' bass (though only at low listening volumes for the Berning) than any of the mega-power amps I've tried with my Quads.

Phase is important, and so too an amp's ability to maintain it. The Sauermann seems to excel in this respect.

Mani.

PS. There must have been a reason why Peter chose the name 'Phase Sure' for his audio company...

'Phase Sure' ... ha ha

Phasure emerged from my before-XXHighEnd 2 years of developing a Local Positioning System which worked with 0.1mm accuracy. This is million$ military stuff ...

But what it really is about is the reverse of audio;
It works with the phase difference of antennas (GPS works the same) which in this case works with the speed of light. So, with only two antennas but both working at a difference frequency, positioning is enabled ... in 3D and for most positions unique. All 100% unique needs two antennas more.

So ...
Most positions unique with two antennas - and in the 3D space.
Two speakers and such ...
At some stage I will have this working. But accuracy is the subject.

This is where the Sauermann comes in.
Or where the NOS1 comes in. Gerd really found me first;
For a more than good reason as it appeared later ...

Hi Mani - usually the power rating of an amp will not directly have any effect on the bass performance of the speakers. Across the frequency band the speakers will take what they want from the amp driving them i.e. according to the impedance they present. The difference between amps is their ability to provide current where it is needed which is not really a function of their power rating.

But that would not be the case with a "current drive" amplifier. And this is where I am not sure if I am right that Sauermann amp delivers current into the speaker regardless of what is happening with the impedance of the speaker. But lets say for a moment that it does. It means that a big problem that most speakers suffer (i.e. that impedance varies considerably with input current) no longer matters because the amp forces in current (in direct relation to input voltage) regardless. And of course that would potentially have a dramatic improvement on sound quality because it would mean that the speaker would no longer suffer for example power compression.

But there would be problems because lets say there is an impedance increase across certain frequencies then the Sauermann would force in current which would increase volume at those frequencies (ie bass) which would not be the case with a "normal" voltage amp. So a special kind of speaker may be required to work with that type of amplifier. 

And so therefore I have to think that I MUST be WRONG and that Sauermann is not a current drive amplifier because there would potentially be problems with many / most speaker designs.

So forget I said anything and we should maybe continue to wonder what is happening. BUT it would be great if Gerd reads this and could provide a little more insight into the design of his amp. Because we are a curious lot around here!!!!

Confused

Paul

As i read the design it is about a way to decouple the amplifiers current path from its voltage path. Current is kept constant at the (bottom) voltage part so it appears unloaded. This has the advantage that there is no interaction between the voltage gain and load wich is a source of distortion.

Unmentioned is the setup of the current sources. These are usually -100%- feedback powerfollowers and have very low intrinsic distortion provided the powersupply can deliver. These supplies probably float wrt the voltage maintaining a constant voltage to the currentsources. At least that would make sense given the voltage amp designprinciple.

So it is more about minimising the interaction between loads IN the amplifier than an alternative way to put current in the speaker. If I understand it well it is designed as a class A bridged voltage source.

Given the feedbackless design I expect it to have a higher output impedance than the megawatt amps. This means that any LF resonance will be audibly enforced.

The whole interplay between amps, speakers and rooms requires that it should be considered as a system. Oh and Peter just added the source component with the 0.7 magic button... ;). That is a whole lot of variables to take on consideration!

My 2cents on this topic,

Regards, Coen

Hi Paul, hi Coen, thanks for your thoughts. Well, what ever's happening, my feeling is that the Sauermann is accentuating the bass and attenuating the treble, compared to my Sanders Magtech amp (which was designed with electrostatics in mind). Let's see if we can work through this.


I think you may actually be right (sort of). If you look at the impedance curve of the Quads, you'll see that:
1. below 100Hz, they're 4 Ohms or so
2. between 100Hz and 10KHz, they're pretty much 8 Ohms
3. between 10-18KHz, they rise gradually to a peak of 20 Ohms or so
4. above 18KHz, they dip sharply to 2 Ohms or so

It really sounds like the Sauermann is accentuating region 1 and attenuating region 3. So my only 'correction' to your post would then be that there would be a rise in volume with an impedance decrease, as opposed to increase. Does this make sense?


Yep. But Gerd quotes the output impedance as 50 milliOhms across the whole audio bandwidth! Hmmm...

BTW, I'm not criticizing the Sauermann amp here - it makes music in a superb way. But at this point, I'm just trying to understand why/how it could sound so different to any other amp I've heard with my Quads, and especially my Sanders amp.

Mani.

My perspective ...
Is a relative one.

I (or we) compared them to my GainClones. I consider the GainClones to be completely neutral. This is about bass, mid, high, everything. A bit dangerous within itself to state so, because first the source has to be neutral too. Well, I consider the NOS1 to be even more neutral, which I can derive from all what I feed to it, comes trough.
Well, as it does that through the GainClones.

Mind you, being neutral is subjective to some extend, but let's say that the definition of it comes from no single track sounding the same. At least this is how I judge it.

Neutrality is not equal to how beautiful something can sound. However, it is dangerous to find examples where "more beautiful" would not lead to the loss of neutrality, or otherwise how the oppononent may not have been neutral after all.

In the case of the Sauermann I found no difference, except for in one area : the highs. Cymbals have more body and sing more.

The area of "highs" is dangerous within itself, because we may consider highs from digital to be wrong anyway. Or they are smeared, or they imply false harmonics, and no one way to do it 100% right exists. Still it would be so that when all tracks sound the same in this area, something along the lines is not rendered how it should be (optimally).
I don't perceive such from the GainClones, and I don't think I perceived so from the Sauermanns.
I must leave it to both being neutral, but the Sauermann makes a better job of it.

I did not - no, not at all perceive any decrease in highs output. However, it should be "softer" for the attacks. It "should be" because of the better colouring I perceived. But I dit NOT notice any softness.

A few days back I described how synthesized drums suddenly became apparent because of a XXHighEnd tweak. I described it as sliky in the positive sense. Beautiful. But what I added to it, was that the lot being synthesized became apparent because -as it only now appeared !!- the edge applied to those high frequency built up synth sound, had gone. This edge is not spread evenly over the whole frequency range concerned, and it comes along "here and there". Call it intermodular harmonic distortion. Things interact. This now, funnily enough, makes that synthesized drums sound more realistic; if you'd look how a cymbal disforms (in slow motion) you can imagine how often its sound changes during its envelop of the sound, and decay.
With this kind of distortion out of the way, the sound becomes more refined. Ehm, more realistic, because distortion is not - no matter it may contribute to "good sound".

Nothing is more dangerous than freshness because of distortion. Sometimes hard to recognize so.

The Sauermann seems to lack the distortion that may be present in the GainClones. Well, officially GainClones *are* full of distortion. Reletively spoken of course, but this THD stuff *is* important, and what I recall from the Sauermann is that its THD figures equal that of the NOS1 (the GainClones as I have them are 6dB worse than the NOS1).

All 'n all you should try to perceive the "less highs" as just less distortion. If you can not do this, only then start digging in impedance stuff and further secrets.

2c
Peter

Hi Peter, thanks for the very insightful post.

I'm going to talk about the Sauermann's HF performance later, but you've summed it up well here.



The first thing that struck me on listening to the Sauermann was the lack of any 'lines' delineating the individual instruments. Edges have totally gone. Instruments that used to irritate me (like loud trumpets) now sound OK... pleasant even. The initial attack, the fine detail and pin-point spatial stability are all there, but the music comes to you as a coherent whole with no edges. When a guitarist plucks a string, it's the whole guitar you hear, and not predominantly the string with a bit of disjointed body thrown in. And I don't think I've ever heard this before.

It's taking me quite a while to get used to lack of 'freshness' that I'm so used to.

Mani.

Reading this, I am happy not having bought a new amp yet. Peter, have you meassured the high power distortion level of the amp or is there any information? I would need the whole 50W for sure.

Here is a review of this amp...in German:
http://www.fairaudio.de/test/endstufe/2011/test-sauermann-verstaerker-stereo-endstufe-1.html

And there is another German amplifier builder following a similar path but is more affordable:
http://www.valvet.de/blocks_A35%20gb.html

Hi to all,

I'm very pleased to see that this discussionis starting in this forum. I hope I'll find some time to give replies and comments in time.
Regarding the amps circuit design: On the first look it might have some similarities to Pass' circuits but in the end it differes in total due to one mayor difference which is, that there is constant current flowing through the voltage gain stage.

Hi Coen, You are pretty close but not totally right.
Since there is constant current flowing through the voltage gain stage this stage has an output impedance of some kilo ohms. This stage can not deliver any power or AC current at all.
The current source that is keeping the current through the voltage gain stage constant has a control loop circuit at a current gain of more then 10 million. This control circuit delivers any current the speacer draws in addition to the constant current flowing through the voltage gain stage. Due to the extremly high current gain factor of that controller loop the output impedance of a single current source drops down to as little as 6 to 7 milli ohms. So the symmetric amplifiers has an output impedance of about 15 milli ohms (which no other single stage single transistor amp can deliver).
One big advantage of this circuit design is that the distortion produced is independent of frequency and load impedance. It is just related to the output voltage. The distortion spectrum remains homogenious at nearly any load and frequency change.
But harmonic distortion is not the only aspect that matters. Group delay, phase shift, bandwidth etc. the entire behaviour within the time domain maters much more. And since my amps are very "fast" also at low impedance loads (1 ohm or less) - the current control loop operates at a bandwith of about 1 mega Herz, the amplifiers bandwith is above 300 kilo Herz - the behaviour within the time domain on complex (low) impedence is best.  
But all these electronic technical aspects are not really the main thing. We talk about reproducing the illusion of music. That is what matters. In the end you have to have the feeling that someone real is making music in your room just for you. You have to be able to feel the "puls" of the music the attitude of the artist, his "message".  
I tried to fine tune my amp by various means to give that illusion best. The circuit is only the basis. The next thing is the quality of the parts used and their exact values within the circuit (determined in listening sessions). Then the thermal stability and at last but not least the mechanical attributes. The cabinets of my amps and power supplies are assambled the way to be not sensitive to vibration. Every screw for instance is treated at least twice bevor being mounted. Plates are damped so that no vibration can cause any microphonic effect ....

All of this is responsible for the entire sonic result.

Gerd

Hi Peter,
thanks for posting that link, but fairaudio did not test the latest revision of my amplifier. There are some anhancements made to eleminate vibrations that have a really significant impact on the sonic quality. The bass gained strenght and dynamic and the entire resolution capabilities of the amp did get better (even though fairaudio rated them extremly good in their test).
There will be another test about the monos comming at "Hörerlebnis" in a couple of month. They will get the amps tomorrow.

Gerd

Hey, great to 'see' you here Gerd!

I know you're busy, but if you have a few spare minutes could you share your thoughts as to why my mono amps sound so much 'fuller and extended' in the bass (and yet utterly taut, fast and tuneful) than even my 900W amp?

I'll be posting more of my purely subjective thoughts on the sound of the amplifier this evening...

Mani.

Hello Gerd,

Thanks for taking the time to clarify on your design!


Very desirable for a true voltage source. I remotely associate it with a similar strategy that was done in the Halcro amplifiers that included a high gain feedbackloop in the outputbuffer. Their basic design was a pimped up version of a standard Douglas Self bipolar though (containing a multiple bootstrapped currentsourced gainstage).


The excellent complex impedance performance should be very favourable to electrostatic loudspeakers!


I cannot agree more on this philosophy. Attention to all details seperates the good from the excellent products. You have got my attention!

Regards, Coen

Hi,
I have put some plots into my white paper (just in german, but for the plots that does not matter):
http://www.sauermann-audio.de/dokumente/sauermann_white_paper.pdf
But remember music usually has a dynamic range of about 10.
If the amp is fully driven, so that the pulses are at 1% distortion (45 Watt) the RMS level is at about 5 Watt where the distortion is at less than 0,1%.
The amp can be easily overdriven by at least 3dB without audible distortion or compression!
This is due to the circuit design and its' dynamic behaviour.
I would be pleased to give a demonstration.

Gerd

WARNING: If anyone takes Gerd up on this, you're probably going to buy the amp. Just saying...

Mani.

Hello Gerd,

As I look again at the diagram, I start to worry about DC over the speakers and failure of one of the legs. How do you adress a safe operation?

Regards, Coen

Hi Mani,

there are (at least) 2 aspects about bass performance:
1. Low output impedence of the amp to control the energy of the fundamental. Even though your Quad speakers do not have large (cone and coil) mass to move there is energy stored as magnetic energy in the speakers transformer which has to be kept under tight control. This is the same as for dynamic speakers where a lot of people say that the discussion about damping factor is senseless since there is a coil of 0.3 Ohm resistance or more within the dividing network for the bass. And compared to this 0.3 Ohm resistance a difference in cable resistance or output impedance of some 50 milli Ohm or less does not matter. It does, because the magnetic energy within this coil has to be kept under control as well. Its about energy and not only about resistance.  
2. The illusion of a bass as an instrument is the sum of the fundamental and all the other frequencies that makes the natural sound of the acoustic bass or electronic bass guitar or whatever instrument. As long as we are not talking about a single tone generated by a synthesizer the sum of all these frequencies have to be radiated by the speaker accuratly in phase at the right amount of energy. If the amp does not do this job well and is not keeping the speaker well under control in the mid- and high frequency domain, the "bass" woun't be really good. I can demonstate to you the improvement of the bass performance of a small 2-way speaker system in exchanging the capacitor in the dividing network of the tweeter!

I the case of your setup I think it is a combination of both aspects.

As I wrote in an earlier post I stated that the damping of mechanical vibrations improved the bass performance of my amp. This has something to do with aspect 2. The energy of all frequencies of a tone played on a bass a consistend "at the point" at the right time.

This has nothing to do with rated output power, not even with the frequency response plot of an amp. It is again a subject of energy handling in the time domain at every energy level. The amp has to deal with milliwatt or microwatt as accurate as with one, ten or more watt.
And I'm sorry to say but most of the amps can't.

Gerd

Hi Coen,

As with all symmetric (bridged) amplifiers with unipolar power supply there is a DC offset on the output. Between the output terminals the DC is less than 200mV when fully warmed up 40mV. This setting is long term stable since resistors of 0,01% tolerance are used which will have a long term drift of 0,01% in 10 years (Vishay S102C). Even the potentiometer to adjust the DC offset on the output is a military grade thick film metal poti (Vishay 1280G) which I buy at more than 15,00€ excl. VAT per piece.
The amp is long term short circuit proof by design. If one leg fails there will be always a huge overload in this leg and the power supply DC fuse will blow (special fast blow fuse). So in this case there will be only a short low frequency square wave puls of 1/4 of the maximum peak output rating on the output (about 25 Watt).
That should not harm any type of speaker.

Regards, Gerd

Hello Gerd

Again thanks for the elaborate answer. I find your design intriguing, since it resembles familiar designs (totem poles) so much but is actually quite different in operation. In the end it is all about how it performs, and it seems it does very well on this critical forum.

Selecting and matching components to keep DC offset under control is a much more elegant way than a adding a servo. It keeps circuits simple but is also much more costly.

Furthermore I wondered why you chose for a balanced/bridged design. Save for the impressive reduction of even harmonics, it is a way to be able to use a single supply. Are the outputs floating or are they in some way referenced to the power "ground"?

Regards, Coen

Well it's midnight here and I've just listened to 3 albums straight through. I think I'm getting used to the sound of the Sauermann with the Quads. I wouldn't say the sound is totally transparent, but I'm pretty certain this has more to do with the way I currently have things set up than anything the amplifier is doing. What's amazing though (for me at least) is the total elimination of a 'metalic glaze' that I think I've pretty much always had, and something I've always mistaken for 'clarity' and 'hyper-resolution'. This metalic glaze has now gone and been replaced by a much, much more pleasant sort of 'woody' sound. Again, I don't think this is a characteristic of the amp - perhaps this woody sound has always been there but just been obscured by the metalic glaze... who knows?

I've had loads of amps in the past, and I can see now that pretty much all of them have, to varying degrees, had this metalic glaze. It seems remarkable to me that they could all have sounded wrong. But maybe that really is the case. I mean, I now know that all the DACs I've had in the past sounded wrong.

I really am beginning to think that the Sauermann will prove as big a revelation in my system as the NOS1 did.

Mani.

Mani,

What you should explicitly look for, is whether the detail is still there; you might need to play the stuff I use for that though (you know). So, glaze gone, but the very high transient "ticks" etc. still there. *Now* you won't have false detail.

... And I didn't notice any of that lacking ...

Btw FYI: I measured the impulse response as I do it for myself to test the filtering and which I normally do behind the DAC. All was followed as expected (this would come down to 29KHz square wave response, but now impulses (only one side of the voltage rails). You can't have it better when THD is OK (I didn't look at that).
What amazed me most is the perfectly straight noise line; This is indication #1 for something good - at least how I look at it. Remember the other pictures you came up with yourself ...

Maybe even more important is that I didn't expect anything else than "just very good", in advance of meeting Gerd. Of course, this "very good" is (to) my standards only, but since I have them anyway ... Some people can let know that their product is good, just by talking about it;
I use to have no conficence at all in whatever comes by. This time it was the complete other way around. And without further thinking about it, it just worked.

Whatever it is you further find, I hope you can map it onto real merits of things. It should (IMO) be similar to the NOS1; nothing is perfect, but at some stage you can see that whatever it is what's not good, is encouraged by something else. In this case, *if* you come to the conclusion not all is 100% transparant (difficult to achieve to begin with), maybe try your other speakers. Reason further from there.
What I like to see (for you, for maybe many more) is that this amplifier *is* the most transparent (because I judged it so), and thus when it doesn't show that, something else might be going on. And oh, not that you so much talked about this, but aren't we hunting for the best playback ever ?
That's why. The more "sources" can be taken as close to 100% OK, the more easy it will be to blame cables etc. by just exchanging what's left to be unsure.

Peter


PS: Might it help someone ... Gerd Sauermann uses the exact same approach for his designs as I do. This is up to the software, although Gerd didn't go that route.
Since this approach is not the most common one, it was really strange (for me) to find this out, which was only afterwards ...

Oh there's plenty of detail... but no glaze now. And what's interesting is that this detail is throughout the whole bandwidth, not just in the treble.


Probably the best description of the sound I can give is of it being 'totally coherent'. Nothing, absolutely nothing 'jumps out'. Everything seems in it's place and is a pleasure to listen to. A great impulse response no doubt contributes towards this.


All I can say is that this is easily the 'quietest' amp I've ever heard. And I mean quiet through the speakers, not mechanically (although you can put your ears right next to the outboard power supplies and you won't hear a thing). The background really is totally black. This is best heard on classical music during really, really quiet pieces. The level of detail coming through even at these low volumes is very impressive.


Not sure which 'other speakers' you're talking about. My smaller Quad 2805s? Well they have a similar sonic character to my bigger 2905s, though it would be easy to set the amps up in my study/office and take a listen. The only other speakers I have here are my old Celestion SL600s - great speakers, but not really in the same league as my Quads, I'd say.


Yep, I'm sure there are a number of other things going on. Once I have some more time, I'll do some experimenting. Right now, I can certainly live with the sound I have.

What's interesting is that if anyone were to come and visit to take a listen to my system, I would be almost embarassed. There is nothing 'amazing' about the sound at all. But that's why I like it. The system gets out of the way and allows you to focus on the music.

Mani.

Hello Coen,

if you drive the amps with a balanced input signal the outputs are floating.
Correction: The output is allways floating, no matter what the input is. There is no relation to the internal ground (- pole of the power supply) nor to earth ground. So you could even hook one output pin to earth ground without problems (what some messurement equipment do when testing on complex load).   

This are the two most important reasons for a balanced design:
1. Symmetry as such to ensure the best protection against any influence from outside.
Think of asymmetric circuit designs with a direct relation to ground. The signal on interconnection cables is directly related to ground since one wire is the grounded shield as well as the ground pin on the mains plug. So the signal might use (and is to a certain extend) the ground wire of the mains as the interconnection between the two connected pieces of equipment. To get best results most expensive mains cables have to be used.
The reduction of even harmonics is a nice side effect that helps a lot to get the best puls response (in my opinion).
2. The circuit I developed in this simplest form is only working on an unipolar power supply. Therefor the output has DC offset. To get rid of the DC a balanced design is the best in terms of sonic result.

Regards, Gerd

Hey Gerd,


I would never do this !
This assumes earth ground to be equal to the (created) potential of the amplifier ... which it will not be.
And there you will have DC offset ...

Or ?

Regards,
Peter

A quick update on how I'm getting on with my Sauermann monoblock amps.


Well, I was wrong and right at the same time!

These are the most transparent amps I've ever heard. This is most obvious with complex music, especially large orchestral pieces. It's as if each instrument has it's own amplifier/speaker channel. Nothing gets muddled up - each instrument remains totally pure irrespective of what's happening around it. I've never heard this before. Actually, with the Sauermann, I think it's a case of 'you don't know what you don't know' - I didn't know that all the other amps I'd heard were 'distorting'... until now.

But I was right about something: the way I had things set up wasn't optimum. And I'm talking mainly about my PC(s) now. I've been a long-time advocate of totally silent PCs. And for good reason I think - I just abhor even the slightest background noise when I'm listening to music (usually late evening once the kids are asleep). So my music PCs have always used fanless cases, fanless PSs and also SSDs. But I did an experiment a short while back and although it pained me no end, I had to concur with Peter that HDDs sound better than SSDs. The music sounds more 'open' and full of life. All the SSDs I have really sound 'filtered' in comparison. So a few weeks ago I decided to build a 'monster' PC and stick it in my basement, as close under my listening room as possible. The PC ('Le Monster') was completed yesterday. It has a total of eight 120mm fans and although it is much, much quieter than I had expected, it certainly isn't something I would want in any domestic enviroment. Nevertheless, I temporarily placed it on the floor next to the NOS1 and connected everything up, just to take a listen. And my God, what a difference to my 'totally silent' PC. Actually, the sound was a bit too forward for my liking so I played around with the SFS and settled on 475 in the end. I'm only now hearing the gorgeous 'sweetness' of the Sauermann amps. They sound truly outstanding.

I suppose I should also mention the quality of the Quad 2905 speakers. They just seem to spit out what they were given. Oh the sound won't be to everyone's liking - the LF dynamics are still somewhat lacking - but hell, I can't imagine a faster and more cohesive sound from a pair of speakers. And now there's a beautiful 'sweetness' thrown into the mix which is intoxicating.

I remember writing a post here immediately after I'd heard the original NOS1, saying something like, "beg, borrow or steal the money for this DAC" - the first time I felt so sure about any component. Well here it is again. If you're in the market for an amplifier [and can cope with its class A operation], beg, borrow or steal the money for the Sauermann. It's on-par with the NOS1... and sorry Peter/Ciska, but much better looking and better built. But just like the NOS1, it's been born out of sheer passion for the perfect sound.

Mani.

(PS. Yes, I know this sounds like advertising. But just for the record, I'd like to say that I paid the full retail price for these amps. It's not because I have a vested interest that I started this thread... rather, it's because I share Gerd's passion for reproducing music faithfully, but just don't have the capability to create something like this myself.)

Hi Mani,

interesting to read. I was just on the road to buy a silent PC.
So I owe you some buckets of beer.

Joachim

Mani

Le Monster - CaseLabs W8, Asus x79, Intel i7 3960X is certainly a huge machine.
No wonder you can't have it in the lounge!
Nevertheless it must look very impressive. Any pictures for us?

Cheers
Frank

Hi Frank, well there's really not a lot to see, but here's a low-res pic taken with my phone. In the background you can see one of my totally silent Zalman TNN300 PCs. This itself is not a small machine, but is dwarfed by Le Monster.

To keep some semblance of being on-topic, notice the gorgeous Sauermann amps on the racks. I asked Gerd to finish them off in white to match my NOS1 ('The White Sheep'). Not even a high-res pic would do them justice - they are simply the best finished hifi components I've ever had.

Cheers, Mani.

Mani
Yes, it is a bit of a monster (and I suspect at a monster price), and hard to hide, but not as big as I had imagined! I understand there are even bigger Caselabs cases?

Your system looks great. I see one of the advantages of having an open baffle design is that their radiation pattern allows you to position closer to side walls without degrading the sound. Less imposing on the living environment as well.

The Suaermanns are somewhat bigger than I imagined for what is a relatively low powered mono amp - which is good of course.

Cheers
Frank

Hi Frank.

The biggest cost was the CPU and of course the RAM (64GB is of course totally crazy, but I just felt anything less would not be worthy of the 'Le Monster' name).

Thanks. The Quad 2905s use a 'concentric ring, time delay' system to create a point source effect. And as you say, they tend to 'beam' quite which means that they can indeed be placed fairly close to the side walls without any ill effect. But you need to have enough space behind them so that the sound reflected from the back wall remains benign. I like the big Quads and in a large enough room, with them placed near the side walls, they're not as imposing as their 'on paper' size might suggest.

Yep, even 40-50W of pure class A amplification requires reasonably large heat sinks. Fortunately they don't get stupidly hot - even after being on for a while, I can easily keep my hands on their heatsinks.

Mani.

Hello Peter,

no!
Since no part of the circuit has any relation to ground you can connect any SINGLE point of the circuit to ground without changing anything. Except that now any other point of the circuit shows a voltage to ground what does not matter, doesn't it?

Regards,
Gerd   

Two things IMHO :

1. The ground you refer to is absolute "zero" with SE all right. ... Within your device.

2. It will influence that same ground reference. Something like the average (RMS) positive voltage will tear it down. Now PE goes down = neutral will go down.

Now, what happens with other devices on that same neutral ? They fight with it. An inherently lower voltage device can't fight back. It will have an offset compared to the stronger device. But merely, your mains will not be a sine anymore.

Gerd, I don't say I am completely correct on this reasoning, but I just see it happening. The max offset I saw was 109V. All it needs is this independent other "PE" reference. I have that here. The more devices you tear out (all SE of course) the lower the offset gets. Mighty difficult to prove in an always consistent setup, because groundloops may undo it and next create problems else where.

So what I think is that the only way to avoid this for 100% sure, is create your own ground reference throughout.
This is what you do with your amplifiers, and this is what I do with the NOS1. Nothing to destroy it, if only PE is left out.
Have one device in there (like my bass amps in "our setup") and the offset is there, even when that bass amp is not connected to PE. The neutral does it, because it balances to PE. The net output will be lower (cancels out) and the noise rejection less.

Again, my reasoning can be wrong (because most of it is just that only), but just too many things add up; Saw Mani's transformer supply ? looks good eh ? well, at the moment he put that in, the DC offset went from near 0mv to 120mV everywhere. And no, I don't know how. And yes, 3x 120 = 360. These kind of relations are in there too (I'm, obsessive now).
These things you see happening when you try to let nature do its work, while nature is different per kind of "mains" (like at some stage I had it working for all, but 110V worked nowhere).
Today I have it working everywhere, but don't ask me what it took.

Kind regards and call me crazy where it's needed !
Peter

A quick update on how I'm getting on with my amps...

They're great, and I love 'em! Every change in the music (different recordings, XX settings, etc) comes through loud and clear. It's a cliché for sure, but I really don't think these amps have much of a sonic character... and yet they sound like no other amp I've ever heard. Ergo, all other amps I've heard do have a sonic character...

Attached are some measurements of the amps' THD performance (taken from Gerd's white paper):

1. Be careful here because I'm pretty sure K1 denotes the fundamental and K2 the 1st order hamonic, etc. This being the case, it's clear that the amp is dominated by even-order harmonics (K3 and K5), with the 2nd order harmonic rising linearly with power.

2. I'm not sure how common this is, but the THD just doesn't change much with output impedance.

3. Finally, THD remains pretty constant throughout a large bandwidth.

I'm no electronics engineer, but I think these results are impressive considering the design of the amp.

Mani.

Hi Mani,
1.
K2 is even order Harmonics and k3 and k5 is odd order. That means K2 is two times the basic frequency, k3 three times the basic frequency and so on.

Here is a site where you can calcualte THD from Klirrdämpfung
http://www.sengpielaudio.com/Rechner-klirr.htm

and here is a review of the amp I use which is a similar design (class a, no negative feedback, third order dominated) and delivers similar distortion values.
http://www.jungson.com/files/reviews/ja99c_uk_review.pdf

In the box on the bottom there is a technical interpretation of the measurements, which could be interesting for you.

In comparison to a design which uses negative feedback, k3 is high. But I compared my "old" class a amp to negative feedback designs with lower distortion and finally I kept my old amp. It shows more details and more of the original colors of the instruments. But it can also sound a little hard on high listening levels. I like it.

Greetings
Flecko

Hi Flecko thanks for this. You're right of course. I don't know what I was on when I wrote what I did... I suppose I just thought that the 2nd harmonic couldn't be so low for the type of amplifier the Sauermann is.


Yes, they suggest the Jungson gets a little 'sharp' at higher output levels due to that 3rd harmonic. I'm not sure if I listen to the Sauermann at high enough volumes, but I'll explore this a little more. It makes me think that an amp like the Sauermann should really only be used with high sensitivity speakers. Mine certainly aren't. However, because they 'beam', you can listen to them at pretty low volumes.

Mani.

The question of the "right" sensitivity of a speaker to fit a certain amplifier is a question I ask myself at the moment. Once I wanted to have very high sensitivity speakers (~100db/w) and planed to use very low powered amps (10w). But the problem with this is, you also amplify the noise of the electronics. Also, the distortions from 1w watt to 1mW normaly increase in a normal amplifier design. So with a very high efficiency speakers you will listen in this "bad" region of your amp. From this point of view, it could be better to have a very high powered amp (500w) and a very low efficiency speakers (80db/w). And all should be matched ideally in a way, that at 0 db at the DACs out, the amp should be short before reaching the region, where it starts to produce an higher level of distiortion. That way you would use the digital volume control at its best. Noise should be very low and distortions at normal listening levels too.

PS.: Just after i read my post I looked again at the sauermanns distortion figures. The distortion does not increase from 1W to 1mW as far as you can see the figures. It would be nice to see the distortion for lower watt and distortion levels. As far as I know this is unusual and could be the reason for its good sound. This amp should match very well with high efficiency speaker.

Although true, easier said than done ! And careful now, because a 1W tube amplifier doesn't amplify all that much. I mean, the gain is relatively (very) low, so the possible noise will be just the same (very low). But ... the gain will be sufficient for the 115dB sensitive speaker of course (even for larger rooms).

The real sh*t happens when you have a higher gain amplifier and run *that* into this 115dB speaker. Now, funnily enough I just have that. And I can tell you, it won't be a much normal thing when no noise is audible at full gain of, for example, my GainClones (20dB gain or so). But since this is topic is about the Sauermann amplifier (with somewhat more gain IIRC) ... they too are dead silent through these same speakers.

But is it about the amplifiers ? ...

Partly yes of course, but for the largest part it is about the environment; the remainder of the chain. How grounds are connected. Ehm, how much noise the DAC produces ...
... whether there's an analogue volume control in order ... Just saying.

Let me put it differently, and remember, all is about the (very !) high efficiency speaker :

5-6 years ago I obtained these same speakers and same gainclones. I used "a DAC" and a passive preamp (TVC). Played with that for years.
I better did not turn up the volume fully (no playback going on) or otherwise a blast of noise (and mains rattle) would be my part. Of course I did all to get rid of it as good as possible.

I could use a FireFace800 in my chain, and I found out I had to shut off all the unused channels (there are 56 in there) because they all produced their own share of noise. Took me maybe a year to find that out.

If I measured the noise through microphone it was around -70dB. Remember, at full gain. But who cares, because who plays at full gain.
Well, all of us today, because the first thing to get rid of is the preamp. Aha.

Long story short, when the environment is made right, only *then* there's only noise from the main amps and the DAC. And the DAC is the first thing to look at. Not only because it may produce nosie herself, but merely because of the incurred for groundloops, EMI and that sort of problems. Pickup from the mains.

You want noise ? just put in that one good resistor (and think about it as attenuaton). It is totally impossible with 115dB spakers and a gain of 20dB or so. Only that resistor ...
(this is how all analogue attenuations on the NOS1 failed)

At this moment it is fair to say I have "no noise"; The NOS1 produces a tad better than -140dB and measured at the output of the amp that becomes -120dB. It can't be much better with a gain of 20dB, right ?
Well, this is not 100% practice, because my GainClones are single ended (RCA) and it very much depends on how the interlinks go to the amps, and in my situation this varies per day (testing stuff). So, as a result I can hear some noise sometimes when I'm with my ear very much into the horn.

But there you have it : the environment again. When I hear noise (ear in the horn) you can bet it will be at -90dB or worse. Just pickup from the interlinks, the connectors or poor soldering of them. This is why balanced interlinks are a must. This is why an "all the way" balanced (differential is how it's called) setup is a must. By a bit of coincidence for a chain with the NOS1 and the Sauermann this is the case. Nothing faked and all based upon a self created ground (which is what happens in a true differential setup). Have the DC offset under control at both ends (DAC/Amp) and there you have the ultimate (for noise rejection, also "in the box").

So let me conclude that when someone perceives noise from an amplifier (outside a "lab" ituation), chances are close to 100% it is not that amplifier alone, or maybe even not at all.

Peter

No, no, triple No. And Greg, your first sentence here seems to show that you don't believe in it much yourself. I mean, who talked about poor electronics ? you. So, you use those poor electronics ? I guess so. Thus, get rid of them (you implied that yourself too, haha).

But following my previous post, it appears easy to turn things upside down. So, we put in an attanuator to surpress the noise from ... itself. Now *that*'s nice.


How is that related ? IMHO nowhere. Well, if you are saying that a higher level recording is allowed to mask the noise which would be audible otherwise, then I understand what you are saying. But never think I will agree with the "solution";

What people often seem to think is that as long as noise is not audible, all is right. But it is not, because noise, at audible levels or not, is in all of the signal, and it creates the signature of it. Take that attenuator again;
It's noise signature already shows that it's totally uneven in the frequency band to begin with. Oh, what ? it's at -120dB and thus inaudible ? NO, because and again, it is in the signal everywhere. This is how any means of analogue attenuation immediately kills the sound. Makes it dead. Dead bird I call it.
But of course you first need the reference to compare with.

So Greg, go hunting for the dead silent system and don't start with replacing your amplifiers.
(for fun : Ask Scroobius (Paul) whom I "guided" to a noisless system, only because I told him it could be done -> took 30 emails, so don't ask me :))

When we play 16/44.1 Redbook material (onto a 24 bit DAC) we can attenuate to 48dB and nothing is lost. It needs decent digital (losless) attenuation, and XXHighEnd just contains that (ehm, for the explicit purpose).

There is much more going on, but this can't be explained in here, and some of the things I might have difficulties with myself. But for example, when I use the NOS1 and a test signal at -3dBFS the THD+N is around 0.0018% (never mind this figure in absolute sense). Now, when I digitally attenuate with -21dBFS, THD+N is around 0.0035%. Now think : Attenuation 21dB while THD+N increased only 6dB (2 times). This, while the signal came 21dB closer to the noise (THD+N) - is what you think.
But it ain't so, because thermal noise in the gain stage got down too (less output from the D/A chips). Actually with that same 21dB.

You could of course say that THD+N got 6dB worse anyway, and thus it is not a good thing to digitally attenuate, net. But for starters it is 6dB and not the 21dB you'd expect.
Now stuff in that analogue attenuator and see the noise floor RISE (instead of drop). With what ? depends on the means. But 20dB is the most easy ... Now 0.0018% becomes ~ 0.0054% only because of the N part. And if that were all ... But because of the uneven characteristic of the attenuator (thermal stuff again) it will be worse, because harmonics will appear all over the place too.

The story is infinitely longer, but this is to get the idea. Or something to think about at least.

Regards,
Peter

Ha,  thanks for the replies.  Maybe I was trolling a bit there :-)



Ah, I see where we are misunderstanding.  No, that is not what I am saying.  Not sound masking, that is not at all a good thing.  I'm just talking about electronic noise generated after the DAC.  If there is excess gain, then there is excess noise.  The only solution for such noise is to reduce the gain by removing (or improving) devices or through attenuation.  A volume control after the noise source reduces both signal and noise.  A volume control before the noise source just reduces signal and reduces the signal to noise ratio.   


So very painfully true.  Just when I think I finally understand the big picture, there is always more.   :scratching:

One thing I'm not understanding is the dislike for analog attenuation.  I'll bet every user on this site uses analog attenuation.  Even if you have a Gainclone amplifier driven directly off a NOS1 you are using a great deal of analog attenuation in the chip feedback loop.

I could even go so far as to claim that "digital" volume is no different than an analog voltage divider.  It is simply selecting different resistors per word in the PCM1704.  After all, the DAC is mostly a bunch of switched nichrome resistors as a current divider.  Music signal currents still must flow through these resistors and all the effects of analog theory apply.  It's just that by careful design BB keeps any filtering to a higher frequency than we will ever worry about.

For the record, my "analog" volume control is a constant impedance network of switched nichrome resistors.     

Analog attenuation is the lowest distortion building block available to the designer.  I agree that using it stupidly, such as driving an interconnect cable through a pot sounds bad.  But, that is just an accidental analog filter.

Often, there is no good spot in the signal chain to place a volume control.  I will certainly agree that if using a chip amp, there is no benefit to adding anything in the chain.  Putting a volume on that amp front end makes no sense because it would effectively be at the same spot in the signal gain stages as the digital volume. 

However, I must still stick by my statement that an analog attenuator placed as close to the last device as possible has benefits over digital volume control that is upstream of noise generation.

Many amplifiers do generate internal noise that is easily audible if you are direct driving horn compression drivers in a multiamped system.

Cheers,
Greg 

Haha Greg, you seem to know more about my system than I myself. So, all still 100% correct ... :) :)
(and still no noise)

Anyway, very nice reasoning about the "attenuation" in the PCM1704. Maybe we should talk adding / subtracting currents, but okay.

Without analysers at hand, or pictures of it, all is theory of course. Or at least maybe it is for you. I mean (and for example only), just try to pull a plot of the noise line from an amp from somewhere (internet) that is 100% straight (up to say 96KHz). So there it starts. Btw, I think I already said that the Sauermann does this. Same for the GainClone.

That is one part only. A next is your attenuator, or preamp for the worst "solution".

Along with this I can only try to tell you (listening would be so much better ;)) that this noise line tells all. When that turns into something uneven (lumps, bumps, slopes up / down) the music is killed. It is just a measure for that. And not to forget, I worked for months on it until I had to give up (if only the PCM1704 could be voltage regulated).

It is a bit more difficult to relate this to the "analogue attenuation anyway" theory you applied so nicely, but let's say I don't see that happening.


That neither, but that's another story (what "filter" ... where ?).

So for example, when I attenuate digitally with 141dB, I have the (btw 16/44.1 !) signal just sticking out the noise, and this noise does not look differerent from no signal being there at all. Notice though that this could be the opposite situation of "attenuating" and I never really wondered what comes first in the 1704 (full signal which is divided, or no signal which is added, but the bi-polar etc. stuff should tell a few things).

Now, with my mentioned -21dBFS signal, the noise line still does not change, but, a few harmonics are there of course (that's just the THD from the chip, and losing bits if you like).
Only when I attenuate less than 21dBFS the noise line starts to rise linearly which in my case will be caused by the means of (Arc Prediction) filtering. But, this noise is still completely "horizontal" (straight). This latter only applies to the audio band, or more when higher resolution is in order (meaning : after fs/2 the noise level rolls off). Well, look here :

(http://www.stordiau.nl/Phasure NOS1/Idle Noise FFT 256K.png)
To keep in mind : The FFT depth used here (and further pictures) makes the noise look 20dB lower than realilty. So it's here at -140dB really.
Anyway, this is what I call a straight noise line. Measure this behind my GainClones and nothing changes except for the gain which is added plus a small 3rd harmonic (sticking out with 9dB IIRC).

(http://www.stordiau.nl/Phasure NOS1/FFT 256K 0-10K.png)
This is 1000Hz at -0dBFS signal at -3dbFS (16/48 Arc Prediction upsampled to 24/192 (and *not* to 24/768 which the NOS1 can digest). All still straight.
(notice this is up to 10KHz (just grabbed an existing picture from somewhere)).

(http://www.stordiau.nl/Phasure NOS1/FFT 256K 0-96K.png)
Here you see what I talked about regarding the fs/2 thing. The 20 dB added must be from my filtering (I see this as random HD resulting in white noise) with possible reactance in the gain stage.
Each dB of attenuation will drop the noise floor linearly, until it gets under the inherent noise (this is what I talked about in my earlier post).

(http://www.stordiau.nl/Phasure NOS1/FFT 256K 0-96K 96KHz-a.png)
Here you see the proof of the fs/2 thing, since this is a 24/96 signal.


Ok. As you can see, noise lines are not sooo straight as I implied they should be, but this is software impeded. To my ideas this can't harm because it "goes on with the flow" and further more it is as straight in the audio band intended (fs/2).
Now add any attenuation means (of which I don't have pictures, or at least not at hand). On of the means I obviously tried was your type of attenuator (I won't name the brand to not discredit well respected people), relais regulated. It was the worst of all I tried. Look at the first picture again, but now imagine lumps all over like you see beyond my fs/2 pictures. But then in the audio band of course. Spikes all over the place, and of course varying per chosen attenuation (btw, digitally attenuating with the ladder DAC shows different harmonics per attenuation level just the same - just saying, but in normal "expected" fashion).
Maybe you can dig up plots from your attenuator, or otherwise go out on Google and find nice straight noise lines which are not 100 times smoothened (look at the thickness in my plots).

Anyway, this is measuring. But before that comes listening. Ad then what's audible is visble at measuring. It just is so. In all situations I tried (up to one-value fixed resistors in all kind of setups, yes, also right in front of the amps).
Any means of transformer (I/V) ? it just can't work (with a TVC ahead as a rough one). I went as far as buying a most official LDR for USD 450 or so, and then to think this was for two "ch" only (so not even able to work in differential mode which would add another 200 or so).

In my case you can also say that I chose the wrong chips of course. But in the end not, because doing it digitally is just the most okay and really creates the best sound by far.
And not to forget, in the 1704 situation 90% of the work in this context goes to the I/V itself, which is a pain ... (but very much "volume" related).

Aaanywayy ... no noise here with dual GainClones and compression drivers. Haha.
Oh, this was about the Sauermann. Well, as said, same story. Same noise line too (and I say this is RARE).

Thanks for sparring a bit. :)
Peter

Nice to read Peter and Greg.

Peter, you told your system has "no noise". Does that mean you cannot hear anything even if you put your ear directly at the tweeter? With 110db/W speakers this would be an incredible achievement. I finally got rid of my preamp (...I must find it out myself) and now also the noise is very low but not unhearable. I can hear a noise comming from my tweeter if my ear is 5cm away from it and if just my amplifier is turned on. I can hear a noise up to 15cm away from the tweeter if my dac is turned on too. As it seems to be very important to get the noise down in the system, it would be nice to have a to do list to reduce the noise. Maybe you or Paul could start a thread about this?

One thing disproving one argument of my own: The increase of what I thought was distortion at very low power, smaler than 1W, could be very likely caused by the noise in the THD+N plot. The pure distortion should be smaler with lower output power. So no better performance with low sensitivity speakers here.
The noise is at least less hearable. As the hearable noise should be normaly come from the DAC/Preamp section, the amplifier should be matched by choosing the right amount of gain to fit to the speakers and the DAC.

Noise, noise, noise.

Great posts

The whole xxhe pc playback is in the end about noise. Are you able to see the effect of different xx settings? Is this plot made with v07?

So this is where the Byebees come in, and everything else. Even vacuum tubes  ;), at least on my side!

Regards, Coen

Almost correct Adrian;
Speakers are 115dB/W @1m.
So, gain of the amps is around 21dB and the base is some 8uVRMS coming from the DAC (idle but On of course).

(http://www.stordiau.nl/Phasure NOS1/Idle Noise XLR-a.png)
It now depends a bit how my interlinks ar routed whether you can hear totally nothing or just that little bit. Coen here is witness of how he could hear that little bit the other day (ear completely in the horn) and maybe he can also tell you how my interlinks (SE) went about. I forgot. Haha.


Edit :
To be honest, this picture shows the output noise with balanced interlinks; with SE (RCA) the noise is ~17uVRMS (say 6dB of difference). Since this is what I use (SE), it is this what I will hear (after the amplification of the amps of course).

No, this is from much longer ago.

The effect sure can be seen but this is merely in the amount of upsampling. So remember, those pictures were about 16/48 upsampled to 24/192 (32/192 actually) and upsampling to 24/768 changes the picture more or less dramatically on where the harmonics fall. So, think about (false) harmonics being pretty much the same for interval (like each 1000 Hz distance for a 1000HZ signal) but some are higher, some are lower (like you see the 3rd is higher in the 2nd picture above). For "nasty" frequencies and nasty harmonics this shifts to outside the audible frequencies (say beyond 20KHz) but this only happens at this 16x upsampling (not at 8x yet).

So, this is not really XXHE version related, but let's say that this is only not the case because I never changed a thing about the filtering (Arc Prediction) means from when it was introduced. It could (be improved) though.

Regards,
Peter

 :o

It's true that I don't know Peter's system.  I was using examples that I thought most readers could relate to.  My reference to 115db 1watt goes back to Altec(and other) horn drivers I've used.   I'm sure Peter has talked about his speakers, but I haven't read that yet.  As far as chip amps go, I heard a few and repaired a few.  I'm neutral on that topic.  They are what they are and seem to be extremely good for the effort and parts cost.  I have never tried running a chip amp directly into a high efficiency horn driver.  It's interesting that the GainClones are so quiet.

Nice plots by the way, thanks for sharing that stuff  :clapping:

Maybe I should take this to another thread, because it's going far off topic.  I am interested in the Sauermann, honest!  Sorry for the hijack......

Read datasheets at night to cure insomnia......., I'm actually quite familiar with the internal diagram and function of the PCM1704 via the Japanese application notes.  It's true that the differential bit switches are operating more or less as current switch only, but the impedance of the rest of the R2R ladder is certainly not zero and voltage is swinging(making capacitive reactances matter).  Hence my statement about it being filtered.  Also, if you have succeeded in making a perfect current sink for the I/V conversion, then my hat is off to you Peter! 

I can't think how the I/V can be made a perfect current sink, so I'll assume a small voltage swing on the current output as well.  Therefore, the summed bit current output is limited in the rise time.  This is a filter, it's just so far out of band that it doesn't matter.  Sorry to be argumentative on that point. :)

It's all just reactances and no different than what happens when you insert an analog attenuator followed by an interconnect, amplifier, etc. 

If a simple voltage divider sounds poor compared to digital attenuation into the input of the GainClones, that says more to me about the input impedance and reactances of the chip amp than about the divider resistors.  Another thing to consider is the sonic effect of the output levels on the I/V.  Different levels might sound different and internally on the DAC chip. From what I can see of PCM1704 there may be sonic advantages to running at a sweet spot in the ladder with regards to major bit crossings etc.  I know the colinear arrangement fixes the zero crossing, but other major crossings exist. 

I don't have a clue what modern commercially available attenuators are out there.  I build my own stuff and also have some lab grade General Radio decades (T and H networks).  I've listened to TVC at a friends house and they just sound like transformers to me.  A bit of dynamics and air lost.  Not my choice.  They did seem to help filter the output a Sabre32 (Buffalo II) as a plus.  BTW, as far as I can tell, this old 16bit Phillips chip beats the Buffalo II when using XX arc prediction.

I'm using transformer I/V right now because I'm lazy and it's very safe for a 4 year old girl to play ABBA and dance :)  The gain is limited so that 0db cannot arc the Quad ESL63's.  Trying to teach her how to run XX.........

Regardless of my current system, I'm completely in favor of active device I/V.

Back to digital attenuation.......  How can we use the same bits for two different jobs?  If you attenuate 48db, are you not using all 8 extra bits below 16?  At the same time, don't you need those bits to interpolate 16x FS in the NOS1?

48db is a lot more attenuation that anyone needs.  But realistically, 24db is not.  If you use -24db attenuation (gives up 4 lowest bits (6db per bit) and also use 16x oversample, where are the discrete data points to create an arc?  If two words are 1 LSB apart, then having just 4 bits (16 levels) does not allow an arc (only linear), or am I missing something?  Isn't 16 integer points plotted over 16 samples a straight line?

Also, it seems practical experience shows that your digital attenuation sounds musical (many NOS1 users).  Maybe we don't need so many extra bits to make the arc prediction idea work?  I know that on my humble 16bit dac, arc prediction at 4FS seems to work surprisingly well.

More food for thought.  Has anyone analyzed how often two adjacent words in redbook pcm happen to be within a few LSB (least significant bits) of each other.  If that doesn't happen often, it would help explain how arc prediction can work well with less than ideal bit depth (as when using 24db of attenuation).

Yes, I am building a PCM1704 dac (as promised), just very busy right now!

Thanks for the entertaining conversation.  I hope I'm not irritating anyone and again sorry for the topic hijack (I'll move if it continues).  My mind tends to wander around with this stuff.  It's all so interconnected.

Greg


 

Hi Greg - Two small remarks for now, and I'll be back with some others probably tomorrow :

1. Gerd Sauermann is as interested in the NOS1 than others are hopefully in his amplifiers. So I don't think we need to worry much about hijacking the topic. It is all about a perfect match (of our both ideas) here - by accident (or not).

2. You are 100% correct on not being able to use the same bits for the different jobs (you know quite some stuff here). I didn't want to dive into it too deep because there's more. But this is why I mentioned the 16 bit (!) signal still sticking out its neck at -141dBFS, which includes the digital attenuation *and* 16x upsampling (onviously). No dither used ...

More later. And thanks for some great posts ...
Peter


PS: I have that Japanese (extended) datasheet too. Can't read it all though. Haha.

Ok, a couple of more responses ...


In band (sort of) it goes like this :

(http://www.stordiau.nl/Phasure NOS1/Pulse Arc Prediction (XXHE) 44.1 176.4.png)
IIRC this is something like 27KHz Dirac pulses. This is measured from the NOS1 output. This is 16/44.1 upsampled to 24/176.4 which gives the pulses some width (of 4 samples instead of 1). Upsample to 705.6 makes them wider again, which obviously is for the better. So, supposed these kind of "transients" are in the music data, then the upsampling (better : filtering) makes them smooth enough not to harm the remainder of the chain. Or at least softens it.

Since this topic is still about the Sauermann amplifier ...
I don't recall I measured this at the output of the GainClones, but I did measure the Sauermann. It shows exactly the same picture as the DAC output shows (but with more output voltage of course). So, all still "perfect" ... (but make your speakers follow this, or otherwise we did wrong :yes:).


Ok, let's be careful that this is not too much theory only. I mean, yes, okay, true perhaps, but sadly all I hear I measure at the output of the DAC already. Now, of course, we can say that the analyser suffers from the same, but if so I rather think towards the inavoidability of this all. For example, I myself use the most poor (well, $1) stuff on cabling and I do this because I can't expect others to mimic my super duper noise rejecting etc. (if true at all) own stuff. So I approach this the other way around (and even ship the NOS1 with the same poor cables).
Anyway, I can't agree. :whistle:


$300++ interlinks also filter nicely. And is also no solution.
Solve it at the source.
(I'm sure you think the same, but I want to emphasize how in 100% of cases things are solved the wrong way (for the worse). Ok, make that 99.99%.


This is for fun and may not show much common sense ... :
What I think is that these kind of "gain" solutions can not exist. Theories may be nice, but practice is that there is no free lunch and whatever is "gained" is taken from somewhere else. So, (I) think like the gain in the audio band is taken from outside of it as a rough example. Or, that the gain for a 80Hz frequency is gained from the reduction of 15KHz. Something like that, and it can be seen in plots.
As I said, the worse.

Hi Greg - good to see you on the forum and joining in with the spirit of things. As my name is mentioned above I thought I had better add my few pennies worth.

First of all let me tell you that I have achieved a level of noise that is vanishingly small and that I did not think possible. My speakers are 98db OK not as sensitive as Peters but still pretty sensitive. How much noise do I get now? well I have to place my ear right inside the cone of the speaker and right up against the dust cap to hear anything at all - even then it has to be very quiet in the room to hear anything. Noise levels are tiny - most people just cannot hear any noise at all. Now that is quiet and way better than I have ever achieved before (and I do not have any balanced connections). And boy did it make a difference to sound quality. As Peter has pointed out elsewhere noise does not just sit harmlessly under the signal it modulates it. So what most people think as innocuous noise is actually very very harmful because it imposes its character on the signal - a very bad thing as I now know.

All my systems generated noise in the past some more than others but all produced hugely more than my system does now. And how did I get the noise level down? - by sorting out grounding and earthing (they are separate) properly and in a way that is just not possible with any shop built system IMHO. So the totally unforeseen advantage about building Gainclone amps is that (provided you know how your other equipment is built - and provided it is built with the care and attention paid to NOS1) you can sort the grounding arrangements to almost totally eliminate noise which is in turn mostly caused by earth loops and RF. After all GC's are based on cheap mass produced chips so they must produce more noise than high quality purpose built hi fi but I hear virtually no noise at all from my humble GC's. So where does most of the noise come from? not the amp from what I have heard.

Of course with the incredible NOS1 at the front end changes in sound quality are very obvious.

All the best

Paul

One small additional remark :


Although several GC chips exist and yours will not be the same as mine, the little trick is a sort of other way around :
With this one-chip design, there's virtually nothing else that can produce noise. There's maybe one feedback resistor only and that's all. The remainder is in the chip (comes from the chip). No outboard voltage regulator impeding noise stuff and such.

Your beautiful "hi fi" component may have dozens or more noise creating parts in the signal path. These too are mass produced cheap thingies (in general). Now, choosing those components for the best performance (on THD+N) is that other trick. But if I tell that to an EE, he tells me I must be nuts because any spec will do hence will produce inaudible noise ...

Peter

Hi Paul,  thanks for the greeting.  Congrats on getting your system quiet.  I couldn't agree more that it's essential to getting good sound.
You can kinda see my point about filtering in the dac chip when looking at the rise time of the transients.  Are the peaks full scale?  It looks better than I expected.  Your I/V stage is certainly doing well!!


LOL, me either.  We may have to agree to disagree :)

Ultimately, all solutions either sound good or they don't, right?  Obviously, exclusive digital volume control is working for many people.  However, I'm having a hard time wrapping my mind around the idea that it is best in all cases.  I hate the idea of using up much bit depth for attenuation, since it's a function that I believe is done very well in the analog domain (I know that is the point, lol).

Various combinations of speakers and amplifiers can have such drastically different gains that I wonder how a fixed output DAC could correctly interface with them all.  I personally own speakers that are nearly 30db apart in apparent sensitivity (horns and electrostats).  At what amount of digital attenuation would you be better off just padding the analog signal?

When I finish my 24bit dac, I'll give it a try and see, maybe I'm wrong!   



Ha ha, not touching that one. 
javascript:void(0);


Yes, I agree that nothing is free.   Except in the case of my little transformer I/V experiment.  These Altec A256C amplifiers happen to have low Z input transformers as an integral part of the design (phase splitter and balanced feedback mixing).  It was just too easy to hook it all up.  The transformers are already there and I didn't add them in circuit.  Now you see why I said "lazy". 

The amps are old 1950's stuff.  I finally restored them to operation after 20 years on the shelf in my basement.  I felt like I had to use them a bit after such neglect.  I don't have pictures to post, but here's a link for a pair.  http://cafe995.daum.net/_c21_/bbs_search_read?grpid=fDQc&fldid=7foC&contentval=0009lzzzzzzzzzzzzzzzzzzzzzzzzz&nenc=y8McCQFicrtqJ1TisjO-_g00&fenc=&q=&nil_profile=cafetop&nil_menu=sch_updw

Using these amps is kinda like driving an ugly car that just keeps running.  I haven't had the heart to put them back into retirement.  Soon though!

Greg 

Hi Greg, I never used XX's built-in vol control for a long, long time. But (quite a while ago now) I did a comparison of the following three different attenuation techniques:
- XX's digital vol
- Audio Synthesis Pro Passion passive (Teflon insulated high purity silver conductors and precision bulk-foil Vishay resistors) with a very short IC to power amp
- Pass Labs X1 preamp (certainly not the last word in preamps but considered good value for the £5K I paid for it ~12 years ago)

I still have the Pro Passion and the X1 and could re-conduct the comparison if necessary. I can't find the thread where I posted my findings, but I do remember my ranking. To my utter dismay (as a strong analogue attenuation advocate up to that point) it was:
1st) XX digital vol control
2nd) Audio Synthesis Pro Passion
3rd) Pass Labs X1

IIRC, the Pro Passion was very close to XX's vol control, but lost some of the dynamics. The X1, even with it's pretty advanced vol control was waaaay behind the other two.

Mani.

Greg, I wasn't even aware of that this indeed is the whole point inyour view. But - or because ... this is a completely different subject. So, to my serious ideas this is not done well at all in the analogue domain. At least it is so different that you might wonder how the both can be compared at all.
But let's not put this to the debate, and actually I never elaborated about it (although I asked the question many times in this forum with nobody ever coming up with an answer - let's stick to that :)).


You are very correct here. Still nature depicts (so to speak) that this turns out quite all right actually always. I mean, when you have e.g. 115dB speakers you don't use 700W Hypexes at the same time, or ?
But nothing says you really won't, although possibly it may tell that the setup isn't optimal when combined so indeed (outside that DAC connection I mean).

With my own genuine experience about the NOS1 and its customers I don't know of any single situation where people are in lack of gain, or have too much. However, it for sure is true that most will have more gain than they like - theoretically. I myself play in between something like -33dBFS and -12BFS, with the normal range between -33 and -21 or so; that I sometimes "need" -12 is because of very soft albums (Crime of the Century is always my example). These very soft albums are not the best to begin with of course, because they lack some 9dB of dynamic range, and that in the 16 bit domain (this is just poor engineering IMO). So counting those out, I am unnecessarily attenuating 21dB at least. But no way it harms or can ever cope with analogue attenuation (oh, I said something like that already ?). And as a matter of fact, there is also no way that I ever ever perceived degration per attenuation level or something, and this while we can fairly say that I am about everything which could make a small difference (be that software or DAC).
... And I also have to meet the first one who tells me such a thing ...

Btw, you won't know it I think, but the NOS1 is to be used without pre amplifier or other means of analogue attenuation. The text about this comes close to "forbidden".

Peter

Yes, full scale (2.25VRMS).

I can not only kinda see your point about the filtering, but sadly you will "lose" here. So yes, it officially is necessary, but no, it can't be done; It can't be done analoguely because you will or heavily influence the normal audio band, or it isn't effective at all. It can't be done digitally because that violates the whole design principle for starters (ok, that's a moot thing by itself of course) and next will kill the sound more than whatever analogue attenuation. This latter is not a moot thing at all, which makes the design theories also not much moot. Of course you can only know when you listen and compare, which in my (NOS1) case is the most easy by just applying the different filters in-software.

In between the lines : you can't guess how much this is Sauermann related ...
(but Gerd better tells that himself if he likes to)

Of course you could only refer to analogue filtering (because what you see in the picture is the result of digital filtering to begin with (Arc Prediction), but as I said : it can't work effectively unless you want a huge roll off of the high frequencies. And it is this you should not want, because it is this where something like the NOS1 excels.

Notice that the picture doesn't show music and it can only be a test signal. Still it is the resemblance of super high transients which exist in music 100 times more than you could ever expect. Not that a hit on a snare rim would incur for it, but at the micro level it does like it can be in women voices. Of course we could say that this "one sample" rise (time) is because the resolution (sampling rate) is too low to spread it over more, but in the end this doesn't matter because it is about the time the transient spreads over, and these transients (spreading over 1/44100 of a second) do exist. Not all at full scale, but thousands in a track, exceeding half of it easily. So yes, that should be filtered by itself, but this happens and this is why that one pulse spreads over 16 samples to begin with and why the higher upsample rate is important. Still it will alias, but this is beyond the audible range, and flattening it with normal (sinc) filtering will flatten all to nice sines. The best idea to turn violines into flutes, but also the reason why you (Greg) work with 1704's instead of some SDM (Sigma Delta Modulator).

Peter

Thanks for the responses.

I haven't had time to follow up lately as I wished (too much to do)!


No, not at all.  But, it is a complex topic and I got off on a side issue.

My original post was regarding proper use of gain structure on the analog side and the correct location (in my opinion) for attenuation and also using minimum gains and so on to control noise.  We really agree on this stuff so much Peter, that I feel bad arguing.  But, as a little aside, I'm finding that the very best amplifiers at my home are also the best for both horns and electrostats.  My favorites are Class A, unconditional stabile, ultra low distortion, etc.  Sometimes this creates a gain problem with certain sources.  So, while I agree that most people with high sensitivity speakers are using lower gain amplifiers (talking voltage here).  It doesn't always work out. 

I will summarize my complaints about advising use of digital volume as an absolute truth.

1)  Not enough evidence.  Yes, the NOS1 w/digital attenuation is heard to sound better than using an analog pad following.  But, that does not prove that the general idea is correct, only that it is correct with the NOS1.

2)  It's not hard to build an analog pad that far exceeds the bandwidth/distortion performance of any other audio equipment.  For instance, I have a pad on the output of a waveform synthesizer to feed the external sync on my scope.  It's flat to at least 50mhz when properly terminated.  This is not a special or rare device.  Resistors with low noise and excellent voltage coefficients are available.  I'm having a hard time believing that using a different range in the PCM1704 ladder and running the I/V at a different signal voltage has less sonic "character" than using a analog pad.  Also, if you don't like inserting a  voltage pad at the amplifier, why not have different fixed voltage outputs in the I/V.  You are already dropping the DAC current across a load resistance at some point to convert to voltage and using a different load results in a different output voltage (can be attenuation).  No need to discuss the  I/V design, that's not anyone's business by yours :) IMHO.  I just want to point out that there are spots to pad that introduce no extra components at all.     

3)  Also, I do have a problem with the loss of bit depth.  It seems like you are asking for two incompatible things to be true:  Either the "filtering" action of upsampling and arc prediction is needed or not.  If we use the bit depth for attenuation, then we are simultaneously changing the character of the sound by reducing the number of bits for upsampling.  Are we now saying that changing the number of bits for upsampling does not make an audible difference?  I was believing that one of the main advantages to XX/NOS1 was the method of filtering.


More later....happy listening!

Greg

Ok Greg ... This post contains a few words which was actually at the end of this post : http://www.phasure.com/index.php?topic=1976.msg20913#msg20913 but it got too long and I cut it out. But saved it though. Here you have it after all and it hopefully implies some answers or shows my thinking. Implies, because I made it before your last post obviously.

---

Nah, that is not how I ever looked at things, although there may be some truth in it. Anyway, it will be tough to meet adjacent samples with about the same amplitude because it would imply a lower frequency wave without any higher frequencies (modulated on it). That's merely something for an MP3 approach I guess. :)
So no, there are no such secrets in there. Thus, when we were allowed to say that Arc Prediction is no real DSP within itself, then at least no DSP is added to make it work better. On that matter it's a "linear" algorithm (and if we'd compare it with normal filtering, then *that* for sure is pure DSP).

But think a bit different;
There won't be any way that a real 24 bit depth will be met. Not anywhere I think. In between lines : notice that I can see what's really happening because of the nature of the NOS1 (which does nothing itself to the sound).
Now, try to think from the other end. That end where I said that at -141dBFS there's still a nice little signal (and no other harmonics to be seen). How can it be ? The most officially it can't, and certainly not when I see that 24 bits can never be met.
Still it does ...

Of course, that 16 bit signal is made 24 (32) by Arc Prediction, but this looks quite contradictionary if we also take into account that each upsampling step takes out one bit to begin with. So, 16x takes 8 bits and 20 are left. Fine.
Then we attenuate 141dB and another 23 are taken. So we took 31 out from 24. Hmm ... I'm starting to think perpetuum mobile now.
This at least proves that when we attenuate e.g. 24dB it shouldn't be as worse as having lost 4 bits.

All I can think of (and this for sure is no law) is that the inherent noise of the system is a perfect dither means. But a couple of things are in order here for good (?) interpretation.
So, first off, higher noise will carry more "patterns" than lower noise. This is just because noise, generally, never is just "white noise" and it is creaeted by something. That something implies a pattern. Could be a voltage regulator, your USB transfer, anything. The more sources of this, the more complex the pattern will look and the more random it will be again, but there still will be a pattern. The more sources (at different frequencies) the longer it takes for the whole pattern to repeat. Nice.
At the same time, the more noise, the less we can get the signal to stick out (of that noise). So, although to some extend we can say that many noise sources will "randomize" more hence create a better dither means, it is useless because of the noise itself (which gets profound).

When all noise sources are perceivedbly eliminated, it should be a complete random thing again (of which I don't believe it exists), then thinking of molecule noise (though in electronics) which will be fairly random. When this happens, we'd have and the perfect dither and the sufficiently low level to utilize it. The signal will still be there, as in my example (sorry I don't have a picture of it).

It is still more complicated when we see that nobody is going to hear that -141dBFS signal, but still we could say that -somehow- it must be so that 24 bits are resolved, while nothing in the chain really did it (also not the PCM1704).
And no, even in my system (or better, the NOS1) no way 24 bits can be resolved when we see that the FS output creates the noise at -120dB to begin with. Too bad.
Ah ... oh ... but when I attenuate 20dB the noise goes down as well with 20dB. Ok, doesn't help because SNR is and remains 120dB (actually 117 because the signal itself is at -3dBFS and now suddenly we recognize the specs of the 1704U-K ...)

And thus the dynamic range (which is that 117dB) depicts that 24 bits can not be resolved. Too bad again.
Ehm, yea, but at attenuating 141dB the signal is still visible.
Oh.

From another angle (but maybe I already told this), when I attenuate 21dB digitally, THD specs go down with 6dB only. Hmm .. Now what.
This indeed partly is about the 1704 not being optimal at Full Scale, which merely is in between the -10 / -22 DBFS range (there it's quite flat). So that matters too. But furthermore it is a bit tricky to look at THD specs, because they are always THD+N and I am actually not much interested in how it relates to noise, while I am only interested in creating a nice "shaped wave" (eliminate stepping distortion and such). So ...
When I attenuate 21dBFS and THD+N goes doen by 6dB only, then I must gave GAINED on THD. How ? well, the closer noise floor says 21dB thus when it's 6dB worse in practice, the wave shape (THD part) must have become 15dB better.
In the mean time I must have lost 4 bits (or 3) but I guess now it is important that we look at 16 bit sources. So there we didn't lose a thing.

Nice story eh ? well, again too bad, because the story is wrong. Remember, at attenuating 21dB the noise floor also drops with that (well, 20dB max). Now what ?
Now nothing, and all what happened is that the chips perform relatively better at -21dB compared to FS. So, somewhere deep down I *am* using 24 bits, which is just because I use them all (and not only 1 more for one upsampling stage), and thus -again deep down- it must be so that attenuating with 21dB must lose 3 bits. That's 18dB while the figures tell 6dB. Fine, then the chips behave 12dB better, relatively. In absolute sense though it is still 6dB worse compared to Full Scale, and my problem will be that I used all the 24 bits to begin with. Had I used 4 only (for 16x) then -if all is right- output at -21dBFS would have (12dB) been better for THD+N compared to FS. Too bad once more that it would be worse to begin with.

If I am not stuck myself in this little story then for sure you out there will be, and STILL the whole story is wrong (I seem to like wrong stories). Again ? yes;
What I suggest all the time is 4x upsampling only. How ? well, indirectly via the analyser's capacity which limits to 24/192 (for its ADC). So ... when I engage 4x and compare that to 16x no difference will be seen in THD+N because the analyser's resolution can't go beyond 4x. Sadly this means that my given specs of 0.0018% THD+N are also not right and in fact they are way better than that. Now think ...

We drop bits (by attenuating) but we don't drop sampling speed. This means that at some stage the analyser will meet a kind of balanced situation between the number of samples it can take (192000/s) and the resolution in the bit depth.
In order to understand this we must first see that a bit depth of 24 is way too much for the slow sampling speed of 192000. So, we take 192000 samples per second, and could theoretically register 192000 adjacent (1 step difference) volume levels. But 24 bits has 16,777,216 volume steps. So that's a way too high granularity compared to what the sampling speed can deal with. Until the number of bits drop low enough to have it even (and each sample could show an adjacent (1 step) level change).

With 768000 samples per second this works a kind of other wat around; Still the bit depth is way too high, but at attenuating we will meet the balance more early (at attenuating, thus at less attenuation). Without real math, 4 times more early and I am not sure whether I must say that this will thus be "24dB more early" with which I try to say that the attenuation of 21dBFs plus the test signal which is at -3dBFS is exactly that 24dB where I see that flipping point of relative better THD. So then it would shift to FS ...
(and the chips are the best at FS afterall)

If this were true (which it undoubtedly is not) then the chips (but measured at the output of the NOS1) should show 0.00011% THD+N which seems feasable. The official TI specs are 0.00136% and it only needs to be ~18dB better to meet this 0.00011% while the differential + parallelled setup (4 chips/ch) may go in that direction.
But hand me 60-90K for a good analyser and I can just look ...

FYI: The way I measure is 100% comparable with the way TI measures (I've been through some nice projects regarding this with them).


Where were we ? ah, that "there is more";
The above ridiculous story is just to give the hunch that many things and phenomena are to be taken into account for the real interpretation of what is happening. The dither thing is almost like voodoo, but this is just what dither is to begin with. If one next feels a bit at home with what these chips do, you can even go further. So, where I suggested 0.0036% THD+N at -21dBFS, I am as far as being able to make that 0.0028% or so. Just by means of software, and I still wouldn't call that DSP.

---

Peter

And then more explicitly ...




Eh, wait a minute. This is not (meant to be) a stupid theoretical debate like we see them everywhere (were it about preamp / no preamp stuff), but real practice from all angles. At this side at least it is.


Which is a spectrum analyser ? If so, you talk practice too and we somehow seem to be even. Agree to disagree is vocabulary I never liked in my life. I am right or I am wrong, and never leave myself in the dark. Conclusion here ?
Something must be wrong.
You could show a picture of that flat 50MHz, but better make that the audio band first. It may unveil some things I predict and you never saw ? Besides that I am not so sure that a waveform synthesizer depicts reality. Just for fun think about how we should measure (for example !) loudspeakers; Do we take the software made for that and which includes the generation of waves / pules erc, or do we maybe need to used the software which performs best in its environment (which is computers !) anyway (think what XXHE is made for) ?


Sorry, you are too fast. Not that you can help that Greg (you just can't know I suppose), but XXHighEnd was there for the public in 2007, and the lossless digital volume was introduced in that same year probably. Totally unrelated to the NOS1.
I am not saying that everybody unconditionally used the digital volume only, and merely the contrary. To this regard we must recognize that posts like in this Sauermann topic from my hand are the most rare on this forum. Not on others, but here they sure are (actually, sadly). With this I want to say that people choose for themselves and I never told to get rid of preamps outside the NOS1 (but people could have read it elsewhere of course). So ... what people do from nature (and some hints from me indeed) is applying a tad of digital attenuation (often 6dB) because it works out for the better. How ? the same thing as I said before ... because the analogue volume works (out !) so much differently than the digital one. This *is* what I told in this forum, and people start experimenting with it. Same thing with Arc Prediction which is a better example : everybody uses it, while I explicitly told not to use it because *that* was for the NOS1.
And why would people prefer a preamp ? because it nicely works as a huge filter for all what is wrong in front of it (DAC).

The above summarized a little : all what emerges for good sound is created from user experience and most often only afterwards I try to find some reasoning behind how things work out. Btw, I am such a user too. As is my wife. And yes, it sure happens that theories (in advance) don't work out. So, I listen. Not only to music, but also to you and my wife. Next comes the reasoning of the why, and I always manage to fiddle that reasoning my way. Not that you find much of it in this forum, as said. But see the previous post as such an example of a half-cooked reasoning (which I had to cut the other day because I can go on and on before it's really done).


I see this as theory in a wrong context. Haha;
I count 3 active components in my whole chain which includes the 1704 (which latter I count for one) and when I count the 3 amps I use per channel as one too. For discrete resistors I come to 3 in the signal path, again in my whole chain. In *that* context I am talking. If you (can) do that too, we talk about the same. So :


See my before post, and I think I already admitted that. Not that it is audible ... which only tells that this should be harmless. And :


Pray that you will never meet the day that you will be working on your I/V with the 1704 *and* can measure it ! (so, better listen only ... you will sleep better).
IOW, that doesn't work either, and people will know I certainly tried (hard). In at least two occasions (the one a year after the other) I tried this for a customer. The first time it couldn'e be done, and the second time I forgot what I actually tried the first time, but after hearing it I recalled it by the sound. Really.

So about that context a paragraph back, ... please listen or measure in such a context first, or otherwise it is one big mess to begin with, and nothing will matter anymore;
The "probem" I have is that I started out with only one resistor in the signal path (were it for the DAC) and no single other component in there (passive I/V). Better than that it won't get, if you can bear the low gain (which I can/could). That is, for distortion (and not so much for overall (buffer) performance). So, with all further developments to give it a decent output level, I only wanted to mimic that "pure" non distorted behaviour. So I know, it can't be done other than I did it (with this D/A chip of course).

Would you be playing with, say that native environment of a passive I/V only, you will learn soon (if you can measure) that there's one optimum and one only. Remember, we are talking about that same 0.0018% THD+N ever, be that passively or how I do it now. So, this is the base resistance (or impedance) the chip wants (empirically found). Make that higher or lower and THD gets worse.
Today ? today I indeed can change the output level to some extend without hurting the figures too much (changing that particular resistor), but the figures will still always get worse because of SNR already (assumed all is tuned to max output already, so I can only get lower).
And then the fun : Attenuating digitally instead of applying that lower output, always gives better figures (don't ask me whether *that* difference is audible, with a theoretical Yes because -as I said earlier- digitally attenuating is not audible (not to me, and not from anyone that I heard of even a single time), so then the figures start to speak (better use the good THD base).


Superfluously : not that I could find. But hopefully it is your time soon ! hehe


Besides what I said in my before post playing a role, I also said earlier on that not all the bits are utilized anyway. So, attenuating a reasonable 30 bits takes out 5 of them, while upsampling x16 needs 4. So, this example (with too much attenuation for my own liking) loses one bit on purpose. Now, I can refer to my -141dBFS story (with leaving it to you to explain *that*) for enough reason not to worry, but I could also say that by the time you will be able to peceive a dynamic range of something like 90dB something else very strange must be going on. Same like the -141dBFS story ... "not that you will be able to perceive that (of course highly distorted signal)". This losing one bit is different, because a. the decreased DR isn't utilized by your ears in the first place and b. the possible sticking out additional false harmonics are way more down (than 90dB) anyway.

Now Greg, you may get less sleep because of thinking that losing this one bit is harmful, but if so, consider two things :

1.
I say that possibly (!) HD rides on the music ("signal") itself, which thus would make the distortion audible at the playing level (say that the level is just "loud" in your room). I say (that I can imagine) this, but don't even know it (like from noise I sure do), so maybe it isn't even true.

2.
This is to be compared with not losing bits at all, and listening (or measure for that matter) to that nice analogue attanuator of which I know the result and which is completely audible and totally measurable.

And as a bonus :
This is all in the midst of using something like Minimize OS from XXHighEnd which is 100% totally completely audible but which is measurable by no comparable (FFT) means. What does this say ?

Well, that *when* things are measureable by means of FFT (outside of what can be (linearly) expected), it must be really really bad (different) for the audible result.
But remember my context, and you better show me your "context" by means of an FFT (noise line of the whole chain) before I guarantee you that all is moot in your situation because it is a big mess to begin with - and no additional poor resistor will harm (could even improve :yes:).

Ok, long enough again.
Peter


PS:
I read that ! Therefore I see this as sparring and I can always be influenced by good argument (which leaves my ears to convince of course :)).

Lol, I deserved that.  I was trying to figure out how to say what I wanted to without making it very long.


Fair enough, I was referring to sine waves on a 50 ohm termination.  As far as flat, not really of course, but reasonably so.  There are always variations in the signal source (a DAC!).  The attenuated and unattenuated signals into the scope channels will track each other.   You could say (and be correct) that I'm just comparing two attenuators (digital in the scope preamp and analog in the 50 ohm line).  I think this ignores your major point about noise, hence spectrum analyser.

I really need to get my shop computer set up as a test instrument.  Those photo captures are pretty handy.  So, no photos to back up my possibly defeated argument......:)


You have me there again.  I can't remember, did I mention that I haven't used digital in my home until the past year.  I am trying to catch up now!  I did occasionally listen to pro gear and mixed a few shows on digital consoles/studio work or whatever.  But it's very hard to hear in those situations.  Too many variables.  Anyway, I did not know that lossless volume control is that recent for home playback.   


Amen!  I think this is the most significant statement you have made here.

I do believe that everything we hear can be explained by accurate theory.  I have yet to hear anything that didn't eventually make sense once I had all the information.  Can I measure everything I hear?  No, not very often.  I think also, that without theory, it's easy to get lost down the wrong road, following our ears down the easy path with immediate gratification only to miss out on much better sound by solving parallel, interrelated problems.  I know that seems non-specific theory :), so for instance:  I've noticed many times people equalizing with component choices to overcome harmonic content causing what they perceive as a frequency response problem.  Don't even get me started on speaker design.  I would say that maybe "ears don't lie, but they also don't think for you".

 

Yes, the same here.  A TDA1545 and two amplifiers.  But I do not want to defend my current setup.  It's still just for testing ideas.



Yes, I notice the same issues on the 1545 with far less range!

See, I made another incorrect assumption.  I had expected your load (to the 1704) to be active (an emitter or a source) for lowest Z and the translation to voltage happen across an isolated resistor.  I totally agree that with passive I/V, there are no choices when you find the right value.  Noise lies in one direction and distortion in the other.  At least that is what my limited experience is showing.

Thank you for the longish post on bit depth, noise and dynamic range.  I haven't had time to digest it yet.

When you speak of full scale not being ideal on the 1704, are you referring to how it's loaded or to intrinsic properties of the chip itself?  I had meant to ask earlier if there is a sweet spot in the output level and does that give an advantage to digital attenuation in this case?

All for now,
Greg

   
   






 

Let me merely thank you for a very nice post which was a joy to read to begin with (I won't be able to do that :nea:).


It is amazing how much one can work on e.g. crossovers and have one thing right but the other not. I did so for more than a year at finalizing the crossovers in my own horns, together with the manufacturer of these fine speakers. Not that the credit goes to me here, but the point I want to make is that after more than a year of balancing out things, I (we) came to the conclusion that all we tried to solve could be solved right in the player software - by means of having that better. So, just better. Nothing equalized or anything - better. This is why it appeared useless to measure speakers outside of this "better" playback software. It matters too much.


I hope that the correct answer (regarding the question) is : both. So, first the maximum signal (always -3dBFS in my case), and next load it such that the highest output (in Voltage) comes from it. In that case the FS (-3) is not the best (IIRC this shows just in the curves in the datasheet). Attenuate this digitally with 10dB (up to 22 or so) and you are better off. The same can be achieved by a lower resistance to the chip (so it will have less output) but all what happens is that the curve becomes linear - hence at FS it then it as its best. Funnily enough (or logically I think) the effect is nearly the same, because you will have less output, and thus can use less digital attennuation. So, this (also) balances out for THD. But you can't be rough about it, because all is a very fine tuning thing. Here you see something of it (open this post in another browser instance and put the pictures next to eachother so you can do alt-tab to easily compare). Watch the yellow lines only :

(http://www.stordiau.nl/Images/NOIV THDN vs Level xxxR DiffIn.png)

(http://www.stordiau.nl/Images/Rather optimal with 314.9mV output.png)

The first one ends (FS, at the right side) at something like 0.015%. The curve is linear as you can see; each more digital attenuation will degrade THD linearly. Now compare this to the second picture, which doesn't show linear at all. THD at FS is now a tad worse only, but as you can see it is better in quite some attenuated range (some -4dBS to -20dBFS for sure). FS is still quite OK too, but attenuating is better. And what you also see is that the second picture is better all the way, up to -60dBFS or so, where things collapse. This is okay with me, assuming we can perceive a DR of 70dB max only anyway, but, of course this is relative to the attenuation to begin with (notice that this -60dBFS is not only happening because of attenuation, but already with softer music (or the more micro detail of it if you want). But this is the chip, and remember, this is passive and only tried to squeeze out the most of it, which was 315mV in this case (in the end I achieved 630mV by another setup).

Now also look at the purple line in the first picture, which clearly does better at the low end. Way better in fact. What does it tell you ? that this isn't alowed to be used with less than -10dBFS of attenuation, and when that is applied THD is at its optimum from -10dBFS to -30dBFS. Attenuate more is allowed, but less sure is not.

In fact this is what I was talking about all the time and here you see it happening, once the load is so that this emerges indeed. And of course you start to feel how complex this is because in the end any curve is possible and it really can be so that a worse looking curve performs way better if only some guidelines are followed (like that purple trace would need a guide clearly).

Below you see two more where the first one reaches almost 0.0012% at Full Scale. The second one makes ~0.00125 of that but can sustain it to -6dBFS, which the first one can not. So, better than the before pictures. Small problem : this is 115mV output only and even for me unusable.

(http://www.stordiau.nl/Images/3 and a bit, 115mV.png)

(http://www.stordiau.nl/Images/3.25.png)
I showed you the last two pictures, because it should make clear that this is the base of it all, and once we are going to amplify this, the base - hence those curves stay. Still it is not said that I use one of the latter two for the base, because the overall THD (at more attenuation) is worse again. There is no free lunch anywhere (we say it must come from the length or the width, which is almost literal here), but we still can chose the best and apply some rules more down the line.

Peter

Hey Peter, nice explanation. Looking forward to reading Greg's response.

But just to dumb things down to my level, I have a question. In my main room, I use XX's vol control usually between -18dB and -9dB (although -6dB for very, very quiet pieces). And 'Peak Extend' gives me a further -3dB on top of this, right? So it looks like I'm usually the 1704's optimum range. But in my office, the NOS1 is connected to a preamp (the Pass X1), which although far from ideal is useful because I like switching between different sources as I work (NOS1 for PCM, Mytek for DSD and my FM tuner for radio). Peak Extend is always on (off for HDCD of course), but should I set XX's vol control to say -9dB anyway, rather than the 0dB that it's currently set to?

Like I said, dumb question because the real answer is "get rid of the preamp". But this would just make my life a lot more difficult.

Mani.

Hi Mani,

No dumb question at all;
This is related to what I talked about a few posts back, and users by themselves (without me telling them) coming to that 6dB or so digital attenuation while using a preamp. So, a. this should be better, but merely b. those applying it found out by themselves.

Notice though that this was a kind of "hot" when the digital volume was introduced and for psychological reasons people start to play / test with it. Today it's always there already, and people just don't bother when they use a preamp (etc.). But the same "guideline" obviously still applies.

Please notice also that I myself don't bother about it too much, already because I myself don't use that preamp (etc.), meaning : all I could be bothered about is whether my gain matches the output of the DAC and next whether I play in that better range. But the latter is a stupid thing of course, and the former applied (ever back).

Next notice that indeed this ever was not about the NOS1 at all, although for sure more D/A chips will be subject to this matter. A next thing could be that when the DAC as a whole is not all that good at the slew rate (merely back down - hence preventing overshoot) it would be possible to go beyond the allowed voltage (what the voltage rails are supposed to deal with) which will incur for clipping. This is a DAC thing of course and shouldn't happen, but now look at our little nice Weiss experiment ... So, here something *else* incurred for the too high voltage, and it would just clip because the DAC isn't made for the thing (bad input).
(others : allow me to not elaborate further here)

More interesting (for testing the merits of the difference between analogue and digital attenuation) would be to play very soft (like people in the house sleeping already) while using a preamp. You can go either way of a balance between huge attenuation in digital vs. analogue, and I sort of claim that when the balance is towards "more digital" you will perceive way more detail at the soft levels. A better balance in the sound too.
But just try it.

Lastly, everybody with an NOS1 will use it in its optimum range. Why ?

1. Because I used the "guidelines" reversed, meaning :
2. People should allow themselves to play the rare very soft albums too while at the same time they won't incur for huge digital attenuation (like 48dBFS would be huge, and by now hoping that people will understand this, hence adjust their gain where possible).

... With the result of people playing in exactly that range, except for this way soft album, which is rare anyway.

Sneaky ...

I envy your capability to post those analyser pics Peter.  I see what you mean about the distortion at FS.  A lot of things make more sense to me now.

I'll try to attach a file.......

Maybe other readers, even if you aren't technical will find this interesting. 

Here is the inside of the PCM1704 (see attached jpeg).  I thought it might be helpful to see where the business of producing the sound is happening.  The diagram shows only two bits due to space considerations.  I circled the MSB (no, should be highest current bit) section in red.  The other bits are represented by the single section circled in magenta and are typical for the remaining 22 bits.

You can see that the highest current bit switch (in red) is connected directly to the output. With a low Z load, nearly all of it's current will sink into the load.  The other bits are summed progressively through the rungs of the R2R resistor ladder and summed to the output.  The rungs of the ladder divide the currents from the lower bits to give the decreasingly less current for each lower bit. 

I think it's interesting that the NOS1 sounds better with some attenuation.  If you digitally attenuate, each -6db, abandons the use of another significant bit.  First the highest current bit is unused and then the adjacent sections as further attenuation in 6db steps.  For each additional -6db, the summing point for the bit currents is moved further down the ladder.

The lower bit switches that connect into the R2R ladder load are already working into a higher impedance than the highest current bit section by design.  They already swing voltage regardless of the load on the DAC, unlike the highest current bit.  (back to the HF filter arguments again, lol) 

Thanks BTW Mani for the report on your past volume control tests. 

I wonder if the slight advantage in dynamics (for digital attenuation) you heard was due to voltage swing issues in the MSB.  Having the load non-ideal (for completely understandable reasons) might make that bit current slightly out of calibration and perhaps affect adjacent bits as well.  It depends on operating conditions of those switch transistors and we can't easily know that.   

On another related topic:

You might see my point now why I tried to say that the resistors in part of the ladder are working in a similar way to an analog attenuator. 

When you are not using the higher bits, those switches are left in a static condition and that unswitched section of the R2R ladder is just a resistor acting as one leg of a voltage/current divider out to the load.  The bit currents have all been summed previously in earlier steps of the ladder and the signal flowing is the complete analog signal.  The resistive divider is attenuating this signal voltage.  That is the definition of an analog volume control. :)   All IMHO of course and I'll wait the response.

Back into theory heh.  With an ideal load (maybe not possible at this sound quality) then the advantage of digital volume should disappear, yes?

Sorry Peter, but I couldn't resist having one more go at this attenuator sparring (as you say)!  It was just too tempting.  

BTW, I do have 4 of the K spec 1704 chips, but I thought I'd try damaging the regular ones first :)  I would tell you guys the I/V design, except it might sound like cr*p and I'd prefer at least some of my failures be private!

Anyway, another can of worms is open :).

(post edited to correct misuse of term MSB by me)

Cheers,
Greg


 

I thought the msb is allways involved no matter how much you attennuate since it is determining the dc value of the output. Iow you only attennuate the ac component made by the other bits in the r2r ladder.

Am I missing something here?

Regards, Coen

Hi Coen,

You are right, in a regular ladder DAC that is how it would be.  However, look at the bit switches and think about the Colinear feature.  In this chip, the MSB in the word determines which bank of switches are used (positive or negative) and depending on which side of zero all bits are processed on either upper or lower section.  If the total value of the word is more than 50% of full scale then the highest current bit of that "bank" is switched on.  This way you never cross zero with MSB currents.  You can see the two banks in that every bit section contains two differential pairs of switches.  Very clever, these guys :)  At least that is how I think it works.  I have to be careful here in case someone who actually knows this stuff chimes in.

Greg

I should not be using the term MSB to talk about this.  Yeah, no attenuation on that bit as it always must exist in the word.  Inside the dac is different.  I'm not a digital guy, so I struggle for the words on this.  I should maybe say, the highest current bit instead?  I'll edit my posts so they make more sense.  Thanks for pointing that out.

Greg

Yo Greg, thanks for that post, and adjusting it on that "MSB" part. But maybe it wasn't so much wrong after all. And maybe it isn't all *that* important ...

In your perception of how "analoguely" it works, you may forget about something, and this is the continuous change of that "analogue attenuator". So, it seems that you approach this as a static set of switches, while they switch all the time (let's say 44100 times per second (or 705600 times per second in the case of the NOS1)).
This is already important when you look at the bipolar setup of the chip, and your mentioned "zero crossing". So, zero crossing in a virtual fahsion all right, but it merely is the switch from positive to negative from one digital sample to the other, and what that implies for current change normally (current goes from zero to "full scale") - and how this is solved in the chip (a bit of another subject, but you seemed to touch it).


Better turn this into : ... we measure is due to the voltage swing issues in the MSB.
Because remember, I measured it first (depending on the created "slope") and next people came up with it, without me pointing it out.
But it also goes beyond just "stupid DAC output" since there is so much more going on. Not qualifying it really, but consider this :

Something like the NOS1, and especially its means of filtering, allows for relatively super transients towards the analogue part of the chain (starts in the gain stage of the DAC itself of course), which theoretically could be too high for whatever is behind it. For example, imagine that pulse train to reach the speaker diaphragm, and maybe wish the "dynamic range" would be somewhat lower. So, distance of the pulses remain, but tops are lower. The "sharpness" of them has gone, and next you will be amplifying it with a (more) noisy amplifier. Could work out for the better ...
Remember, just one angle, while many more can be thought of.

But this was an example of how less DR can work out for the better, while I merely point out that there just *is* more DR at that little attenuation (see that -6 to -22 or so in one of the pictures (slopes)). And never to forget, in this case (NOS1 specific) the noise goes down linearly.

But still from the same angle, think about what too high micro dynamics may do (with emphasis on "too high") : they will imply distortion and that by itself will be looking as "feshness". It is a pitfall so easy to fall in. It may even color things like cymbals a nice natural way (the transients of them - which are gradients - riding along a slope which nicely works out in "sibilance" which is natural to the cymbal itself).
So much dangerous stuff here ...

Anyway :


So, if a normal analogue volume control would be changing itself 10s of thousands of times per second, then maybe. Haha.
But on this part an R2R D/A chip wil only make things worse, because now there's also the "glitch distortion" (there's a more formal pehenomenon for that, but I forgot it at this time). But let's say this emerges by the current changes themselves, a bit similar to the "zero crossing" you taked about, though way smaller. You can (also) well say that the current the chip needs is changed by herself to it deteriorates herself.
But this is only pro normal analogue volume, so never mind. :) :)

We can all sum it up by stating that there's no way to compare the both with easy theory because too many parameters play a role and we always have to look at the whole chain. Or think of how the passive I/V suddenly is subject to the reactance of the amplifier which is out of our control (thinking random apmplifiers). So many parameters ...
This is why (endless) measuring tells all. That is, once you believe in better figures sounding better, and while at first (long ago) I myself thought that better sound not necessarily measured better, that indeed is a long time ago now. It just does. Always.
Did I say always ?

Regards and thanks,
Peter

Please bear with me, I am going somewhere with these posts :) and thanks for the reply Peter. 

Still not sure if I should use MSB and LSB for the ends of words that get truncated.  I ended up confusing myself with that.  From now on I'll use the bit numbers as shown on the datasheet.  MSB of the original word is 1 and the LSB is 24 (for 24 bit audio).  Sorry all for the sloppy way I worded that post.  I'll try to clear it up now :).

One of the reasons I posted the schematic is I never saw a good explanation of the 1704 internals and what things there affect the sound in which ways.  I did quite a few searches and only found that Japanese application note and I can't read it either BTW :)  At least the schematic is clear.  It has been posted elsewhere on the net, but with little or no explanation. 

I think I understand the point you are making about dynamic range.  However, I'm still having trouble with valuing bit depth based on the noise floor.

I'm also still wondering about filtering benefits if using more than 4 bits for upsampling.  I'd love to see a spectrum analyser, but maybe it wouldn't resolve.  I suppose you have done listening tests and I cannot do that yet.  I also question if small transients and noise riding up near the high frequency limit, might get imperfectly filtered at 4 bits.  I mentioned in another post that I couldn't understand how adjacent words that only differ by one LSB (at 16bit) can be filtered to an arc with 4bit upsampling at x16.  That is still confusing to me, since the math doesn't work. 


I don't think we are seeing it the same way.

I'll try to explain my mental image of it again.  This is what I see when I look at the schematic:

At zero, the two switches for each bit are in opposite states.  Each controls a separate current for that bit.  One is normally open and the other is normally closed. 

Depending on the state of bit #1 MSB, closed switches can open or open switches can close, for either positive or negative waveforms (from zero).

Starting from silence, as the signal level builds, increasing large bit currents are used starting at bit #24(LSB) and working towards #2 (max current).  The state of the #2 bit switches never changes unless the signal exceeds -3db.  The state of the #3 bit switches never change unless the peak signal exceeds -9db, with bit #4 at -15db and so on.

As TI / Burr Brown says in the datasheet "The sign-magnitude architecture, which steps away from
zero with small steps in both directions, avoids any glitching
or large linearity errors"   (edit:  They call this Colinear in the PCM63 datasheet)

The high level bit switches don't do anything at all unless there is high level signal.  When you digitally attenuate, you are removing the possibility the high level signals can exist, so the high level bit switches sit idle.  The output end of the ladder is just sitting there as passive as can be.  Even the closed switches are a very high impedance current source and have no effect.  There is no switching or glitching, just analog attenuation.

This is very different than my lowly TDA1545 or other dac designs.

Each time the digital attenuation reaches another -6db boundary, another bit switch section can no longer be actuated since those word values cannot possibly exist.  So if I'm right about this :).  We have the first boundary at -3db (since there is a possible +3db for peak extend).  Any digital attenuation of -3db makes it impossible for data to exist that could actuate the bit #2 switches, so that section of the ladder then becomes passive.  When digital attenuation of -9 is called for bit switch #3 is likewise idle and another section becomes passive.

Those now passive sections of the ladder each attenuate 6db (when properly terminated) The signal is already analog at this point inside the dac.  It is analog because it is complete and contains all the bit currents.  The bits have been summed (1 bit per rung) through the R2R ladder.  By digitally attenuating, you are progressively moving the active digital portion back up the ladder away from output end in 6db steps (1 ladder rung at a time).  I'm guessing that the rungs are composed of 500 and 1000 ohm resistances, since they claim 1000 ohm output impedance.  As you digitally attenuate, those resistances are still dividing current/voltage, even if there are no bit currents being added at that ladder rung. 

If you took that same resistor network and put it outside the DAC what would you call it?   

I now view this digital attenuation as an analog stepped pad inside the dac with 6db steps, in series with a 0db to -5.999999......db digital pad.  This is inspired by our debate :)

The "analog" steps should be slightly wrong (not 6db) for the first couple steps since the ladder is intentionally terminated out of spec(for passive I/V), if I understand correctly the NOS1.  This also explains the distortion graphs you gave since the current division in the R-2R should develop small errors as the output end is approached.  The last couple bit currents in the ladder would not get divided as accurately.

The distortion graphs are actually strong evidence that my explanation is correct, I hope, haha.

Maybe this view seems crazy and If I'm wrong I would certainly appreciate a better explanation. 

Best regards,
Greg

Haha, say that !

Your logic about the bipolar (as how it's called for the 1704 to my best knowledge) seems correct to me, although I have problems with its working anyway. So, not with your outlay, but with its working. But, as it still seems to me, you let guide yourself by its good intentions (of its working) to derive other things which now seem wrong. Seem - to me;

Your perfect example is the Peak Extend (smart digging btw) which points me to how you actually think. So, with Peak Extend - and 3dB attenuation always for the necessary headroom - the whole bipolar feature can't work. That's what you say, right ? Ok. I agree. But about there the agreement stops.

You talk about the signal slowly rising, and thus the bipolar feature won't be active in that stage. Correct again in my view. But where it goes wrong is that somehow you pose that at any wave cycle or something, bipolar gets active and then ... then what ? then only helps distortion for that highest possible output level ? that by itself is something I have problems were it about the design itself (that is, what I logically derive from it for its merits), but merely : that highest output level may not happen at all. So, play a nice classical piece and only a relative few seconds of the whole piece the highest MSB may be "active"; all the other minutes - not.

So, while you say "when the signal starts to rise" you somehow seem to imply that at some stage it will reach the maximum digital level, while this just is not true.
For a test signal maybe. But even there, only for a fraction of the wave cycle.

We better say that our both perception of how the bipolar feature works is wrong. As a matter of fact I already know that at least I am wrong (so you too hehe) because I can prove that by some means of measuring. But that doesn't imply I know how it works. Merely that it's always active, and isn't hung op to the MSB.
A clue could be that servo part (see your datasheet) which IIRC is described differently (in words) in the normal datasheet; combine the two and your insight migyht be different (I didn't look for ages, but this is what I recall).


In between the lines and maybe funny :
At this time there is no single person within TI to be found who really knows how all is working, and whether it's at it best or could be better. Maybe the person exists, but he can't be found. I mean, by the TI organization herself. It also seems that I may know more about the stupid chip than anyone else within TI. As I said in between the lines earlier : I know how they test the chips, what they are tested for *and* what is wrong with it. This latter comes from myself only, because no feedback about it is available, or maybe I should learn Japanese. So, we spent some nice posts on it by now, but imagine it to be ten times more/longer when the chips are really down. :)



The introduction you have in your last post, where this is part of that, shows some great deal of confusement, while something like this quote doesn't make it better. ;)
At least in that other post it was clear to me what your thinking was. WAS, because now you seem to explain it better.
Oh.

What you say, I think, is that we have a stream of samples and each over the other differ one LSB, BUT, going up and down. So, the general level stays the same.
Now think practically ...
If this really would be the case, it would be about a frequency of 44100 - ehm, no, 22050 (up *and* down). So, 22050 indeed is resolved by two samples only, and here you'd have it. Okay, at -90/-96dB.
When this is upsampled 16x, we have 32 samples instead of 2. No problem. The wave gets virtually 16 times longer, but, it's also passed 16 times faster. Same result (for frequency).
(btw, count out where the sample points are for the 2 sample base, so this is *not* a square but a wild monster (urging for sinc filtering).

Referring to my earlier little talk about this kind of stuff ...
the 32 samples have sharp edges now. The 2 also had sharp egdes, but it was in balance with the (time) length of it. So, analogue could smoothen the 2 samples, while with 32 the story will be different (never mind this, and it may be hard to justify, but it makes clear the nest better, hopefully);
We need to smoothen those edges, which happens by adding more level resolution. So, adding 4 bits (2 x 2 x 2 x 2) just does that. Less is still out of balance, and too many is a kind of useless (also not in balance).
The edges now have slopes, where btw the slopes are "eaten" from the samples in time length. It is here where your sine emerges, while at first it was a square (but don't forget the monster because of where the sample points are).

If you call this math, then I don't see where it's wrong.
If this isn't math at all, it's also no wrong math. :swoon:

We now utilize 20 of the 24 bits. 4 more to go for 24dBFS of attenuation without losing anything;
The only thing where the "algorithm" could be off somewhat, is -like I said earlier- that I just use all the bits there are for new level calculation, and with 4 less because of the filtering itself, that level will be less accurate. However, think about that balance which has some truth in it. So, you can have 64 bits for the level if you want, it is still the sampling speed which determines where those levels fall, and although more accurate, no "adjacent" line of level steps of 1 will be found anywhere. So, you could say that 64 bits (or whatever) leads to a more accurate level for where the samples are, but it is a bit of a moot thing because the samples themselves are too few to do it right (to justify the granularity in the levels now).
Filtering is always the product of sampling speed and bit depth but the one with the least granularity determines the accuracy of the product.

To make you feel better, maybe turn this the other way around. So, we skip the filtering but have 4 more bits for better level accuracy. Below you see two sets of pictures; for each set the first unfiltered, and the second filtered/upsampled 8x. First set is 5KHz, second set is 2KHz.
It's a no-brainer what to use of course - but which doesn't imply that digital attenuation is or is not to be used; it only tells that we always want to trade the 4 bits for the filtering. What's left (another 4 bits) and what to do with them, is something else.

Peter

PS: I grabbed these older pictures from somewhere, but I think the first of the sets are not right for how the one step emerges into the other. The steps are sure there though, and it is about that.

Thanks for the explaination again.  I think my confusion came from expecting an advantage from a more precise interpolation (out of band of course).

For the example I asked about (two adjacent samples differing by one LSB), 4 bits at 16x upsampling does not give the same shape you could get with more bits(before analog filtering).  I was thinking if you used more than 4 bits that the amplitude of any sampling artifacts should go down and the effective digital filter frequency should go up.  4 bits seems a rather arbitrary choice.  Why not 3 or 5 or 8?  You seem to already have proved that x16 sounds better than x8, so why not go for even more smoothing?

It all goes back to my inclination to preserve as many bits as possible for "something".  I'm not sure what the use would or should be and you make a pretty convincing argument that there is no reason to avoid use of bits for digital volume control.

I can see that your design choices for XX and the NOS1 dac are mutually supporting.  I respect that thinking.  It seems to me that your plan is based around attenuating the "loud" rather than boosting the "soft" recordings.

Would it be an equally valid choice to load the PCM1704 for linear full scale, implement lossless digital "gain" and use more bits for upsampling?  That would preserve the minimum parts count you have achieved.  I realize the gain question is tricky.  I guess it all depends on the usefulness of extra bits.

Dunno, just thinkin.
Greg   

The digital gain is already in there (auto - and limited to what it can take).
Sorry ...

:)


PS: But max what I ever encountered was 4.5dBFS (possible boost).

For a discussion I'm having with Gerd right now I thought I'd open the ProPassion passive attenuators up to see what's inside. I bought these over ten years ago and paid ~£700 back then for a fully balanced setup. But from the picture (only one channel shown) it looks like I was short-changed - they don't look 'fully balanced' to me - but anyone, please correct me if I'm wrong. Maybe this explains why I prefer the sound of XX's internal digital volume control?

Mani.

Hi Mani,

Before I start thinking about what I actually see - how can it NOT be balanced while you will be having some balanced interlink connectors in it, them ending up in a balanced amplifier ?

The only thing I can think of is that in that case the sound is 6dB less without you knowing it.

But further ? You will recall how you measured DC Offset from the NOS1 (the version without the metering). Well, find yourself a 50 Hz test signal (but 200 or so will still work, 1000 I'm not sure) and do the same at the end of your balanced interlinks. One change : set the Voltmeter to AC;
With your posted picture I indeed imagine one of the pins not showing a voltage ... (and notice that what you see should be reversed for plus and minus on the both pins)

Peter (hoping he did not overlook something or made a mistake otherwise)

Start laughing - I am just trying to be creative ...

http://www.audiosynthesis.co.uk/propassion.htm

From in there, and it seems to be a key phrase :


If that means that it physically can have one XLR input (like from one channel) but it outputs in "dual mono" up to doing that by means of one XLR connector not changing the phase underway (all far sought !!), you'd have a balanced control.
Use two of these boxes for balanced stereo. :swoon:

But I am over-creative here.
Peter

Haha, I think they added the 'dual-mono' line when I ordered mine. I was originally using them in a multi-amp system and I needed to match the gain betweeen the sub-amps and the rest of the system. But I needed one attenuator on each side, so they made a pair for me (see attached).

The more interesting line for me is "It can be supplied with top quality WBT RCA sockets for unbalanced use or with Neutrik XLR plugs, sockets and additional attenuators for fully balanced operation."

So where are my additional attenuators for fully balanced operation? Surely there should be two rows of resisters per channel, one for each phase, no?

Mani.

I was looking at that line too, but I envisioned that since you have XLR connectors that extra set of resistors was virtually assembled in your case. Virtually, because apparently that is not the case.

But now assumed that one of your pins will be dead, there's not much balanced operation in order. So, no noise rejection plus an unnecessary low output (6dB less).

Or ?

Mani hi,

I have a ProPassion silver of the same era, mine is a single ended version (RCA inputs and outputs). The shalco switch internally has two layers in mine (left and right but otherwise looks identical to your picture. I cannot see how the version you have could be balanced. Room for improvement there if you are looking for balanced end to end connectivity.

Best,

Nick.

Hi,

This looks like a fully balanced control to me. Two 3k5 resistors from input to output for each phase and a "pinch" resistor on the (very nice) switch.

Off course neither the input nor the output impedance is constant. It varies with each volume setting. But that being constant was not claimed i presume.

Regards, Coen

Well, I always was terrible at electronics. So could someone explain how a single 'pinch' resistor could be used for both phases?

Mani.

Well,

I'll give it a shot.

The pinch resistor kind of short circuits at the output.

 When volume is completely down, plus and minus are tied together at the output. Input "sees" 3k5 to ground on each phase, impedance at the output is zero ohm.
When completely up, the driving device (DAC) "sees" the 3k5 in series with the input impedance of the amplifier (600 ohm per phase?). This will allready act as a voltage devider.
In between the "pinch" resistor is parallell to the 2x amplifier impedance. The smaller the resistor value the more both phases are short circuited, the more the signal is attennuated. The same thing but in different wordings is saying that the more attennuation, the more the plus and minus are summed at the output pins. Complete summing leads to zero signal at the pins!

Attenuation range is probably limited and the pinch resistors will be small in value (which is advantagious since these are dominant for the output impedance of the attennuator).

I can't think of a simpler balanced attennuation device.

Regards, Coen

P.s in the single ended community this circuit is known as a "shunt" attennuator with a fixed value resistor in series with the signal. (my personal preferred way of building an attennuator dispite some obvious theoretical drawbacks).

Hi Coen, thanks for the explanation. Really appreciated, and I think I get it. So apart from a varying output impedance, this seems like a pretty neat little 'trick'.

I would have used these passives in the upper 1/3 of their range (from -18dB to -6dB) and maybe here the output impedance of the passives and the input impedance of the poweramp played a part in the difference in sound I was hearing. I can't even remember which poweramp I originally used but whichever it was, maybe its input impedance was lower than ideal for the passives. But I do know that all interconnects were 1m or less in length.

In any event, the digital volume control in XX sounded the best... for whatever reason.

Mani.

[this post has been on the screen for the whole day because I hardly dared to post it ... ]


Answer : Because the signal now goes through that "fine" 3K5 resistor(s) ...

All I did before was concentrating on all the other resistors, thinking "would those be the best for audio ?" - and next I never saw that this other XLR would be the output. Or actually, that this other again would be the input.

So, the resistors on the pot won't matter a thing (they can be lousy ones), but that 3K5 does. Now all is relative ...

Suppose you are looking at a pre-amplifier, and this would be part of it. Envision a preamp built in discrete fashion. There will be these parts (resistors) here and there and everywhere. This is one more. They all could be of the best type, and this is just one more of it. Now the theoretical (or mathematical) problem : In the NOS1 is exactly one resistor like this (per phase and channel) and it should be the best I could find (for thermical noise, deviation). I assume it is better than we see here, but let's say it is of equal quality.
... Now you added as much "distortion" as there already was. So, twice as much now.
Keep in mind : all is relative.

Additionally that 3K5 resistor can't be the best for passing on the higher frequencies.

I am not sure anymore whether I tried exactly this setup in-DAC but I imagine not because I don't recall using one (mono) pot only for one channel. But then I don't think this is a shunting attenuator (as you seem to suggest Coen) because I see no ground involved. So, just (slowly) shortcutting the both phases is what's happening here (but correct me where you can !).
Heck, I just don't know anymore in what type of (volume) setup I applied this kind of thing; maybe it was behind a transformer gain and that sure didn't work because of the transformers. Point is : I recall this as the most easy "attenuation" which I didn't see myself at first, but it was not this situation. IOW ... maybe I should try this now ...
I just don't see how it can harm the signal and all I need to do is change that one resistor into a somewhat higher value so the one resistor there just remains that ...

:smirk:

Hi,

On a more technical level, the currents from the inpot voltage are flowing to both phases to the outputs because of the pinch resistor.

So current on the plus flows also through the pinch to the minus output (and input throught the 3k5) and current on the minus via the pinch to the plus output (and input). These currents are summed at the output, hence the blending of positive and negative currents (iaw attennuation of the phases).

Note: to clarify the above, this is when either the plus or minus has the positive voltage. The increased current through the series resistor will cause the attenuative voltage drop at the amplifier input impedance. On the negative leg this positive current will cancel some of the voltage developed across the amplifier input impedance.

No groud is involved. It is not nessessary and is a lot harder to make symmetric (maintaining the balanced nature of the signal).

Since signal current flows through the pinch resistor, i would presume that quality would matter here. Such is also my experience with the se shunt (to ground) attennuator. Since the current will be less than through the series resitor(s) one might argue that is is less important (but not unimportant).

Imho you are trading of dac liniarity for resistor noise, parasitics, hf cutoff and thermal issues. The constant low output impedance of the dac also will do no harm. Furthermore resitors do distort measurably, as you can ee one of the linear audio magazins.

Regards, Coen

Thanks Coen.
This is what I have been looking into yesterday;
So, indeed the current will flow through the pinch resistor, but IMO that can't do much. Where it sure can matter is at the other side : the output which is feeding this. This is the DAC in our case, the NOS1 in particular. So, if you envision something like a gain stage in there, now current flows through there. Can still be harmless, if only the device(s) in question can have it. And, with a proper balance of the series resistor and until where attanuation is allowed, it looks like it can be totally harmless. Key here is this attenuation, which I personally already never thought to be from full to zero. So, in the NOS1's particular case I'd say that something like 20-24dB of attenuation is just perfect; it will allow to dial in the optimum THD figures coming from the digital attenuation as discussed heavily earlier on in this topic. And otherwise, of course, something like 24dB will be a very convient range of operation.

But it has to be tested (measured). The shunting to ground at least never worked, I think partly (?) because of influencing the whole grounding scheme with it. This, while the whole lot is a pure differential setup. I think that by attenuating one of the phases only the slightiest off, the whole balance (zero point) is drawn to somewhere, and this with four of these points that can go off (2 phases per channel). All this stuff is super tricky because all *depends* on its virtual ground, while there's also DC corrective stuff which could be called "mechanical". Not that I saw it, but it can easily oscillate I think, and I wonder how that's seen on the analyser when the frequency is sufficiently high. Think about the data itself too (the voltage of it) and what happens when one phase is exponentially higher than the other (I'm not even sure whether this is linear or not) when the level of the music changes. So, already that can bring things out of balance with nasty results. Just thinking out loud.
And for that matter, if you'd look at the output level of the both channels of the NOS1, you'd see that both will not be 100% equal. It may differ 0.02dB or so. This not only can't be achieved in equal fashion, but also not consistently per DAC. Thus, "nothing in the signal path" is quite true, but from the beginning to the end there's a lot of circuitry per phase (starting at the transformers). So, we can't say there's nothing on the PSU for components, and that too is differentially setup (with 0.1% resistors, but still). On that matter we'd be better off if the differential signal would have been created with some OpAmps, but *that* wouldn't comply much to "nothing in the signal path" ...

But this is why I now like this means of balanced attenuation. It just can't do something else to the plus opposed to the minus, right ? But keep that current under control. It just should work, if it's only dedicated to the outputting source. And this, of course, is under my control here.
So I guess I will have *another* round of attempting an analogue volume control. But later.

Thanks again,
Peter