The best amp and speaker setup for digital playback

[PietPara]

Hi everyone, hi Peter,

I've come across this site through another website referring to the best PC audio players available (I believe it was computer audiophile) and have been eagerly readin a lot of stuff here.
I like to consider myself a reasonably experienced audiophile (25 years of hobbying and hearing a wide variety of equipment) but only on the listening front, not on the in depth technical front.
I have built a HTPC and recentlzy putchased a Xonar Essence ST card for it with the extension board H6 (for making it a 5.1 system in the near future). For now I have been enjoying very much the sound quality of this card over my Dynaudios using my Duson A10 and C100 amplifiers (Class A).
I too am succombing to the virtues of PC-audio and have been on a quest to confirm, explain or rearrange my thoughts about this theme, but I find I understand too little of the technical side to make convincing judgements for myself.
I hope to find some answers/opinions here, seeing that there is so much experience at hand (or brains to pick at . But before doing that, let me first find out if XX will be a a good player for me.

1) I read here that the Xonar Essence does not support some bitrate formats that XX supports. Does this mean that XX is not any better than e.g. foobar?
2) Is it really true that you can get bit-perfect rips using EAC? Does that not then mean, without dispute, that buying e.g. a €20.000 Wadia is absolutely pointless, unless you just like the sound that the DAC in a Wadia produces? Just don't buy it because a better drive system makes any difference. Or is there more to a drive system?
3) I read some motivations that a Squeezebox is must have. I still don't understand why one needs this is one has a good soundcard and e.g. foobar or this XX player? What does a squeezebox add to the equation?

Anyway, I could go on, but it's a start.

cheers to all
PietPara

[Per]

Hi PietPara,

First a big welcome to this forum

XXHE (XXHIGHEND) might not be the easiest music player to setup and use - but to  many - me included - it is definately the best. So don't give up - no matter what happen 

To your questions:

1) I read here that the Xonar Essence does not support some bitrate formats that XX supports. Does this mean that XX is not any better than e.g. foobar?

Even at 16 bit 44.1 khz standard CD resolution I think XXHE is worth the effort / registration fee. Others might chime in here. I use an Audiotrak HD2 PCI card at 24 bit but have yet to explore higher sample rates than 44.1 khz.

2) Is it really true that you can get bit-perfect rips using EAC? Does that not then mean, without dispute, that buying e.g. a €20.000 Wadia is absolutely pointless, unless you just like the sound that the DAC in a Wadia produces? Just don't buy it because a better drive system makes any difference. Or is there more to a drive system?

I use dbPowerAmp as I get Acourate rips faster and easier than with EAC. But a lot sees EAC as the defacto software for bit perfect ripping. To the Wadia comment: I would always prefer a PC to an (expensive) proprietary setup - but of course you have to invest some time to get things right.

3) I read some motivations that a Squeezebox is must have. I still don't understand why one needs this is one has a good soundcard and e.g. foobar or this XX player? What does a squeezebox add to the equation?

I have considered a Squeezebox myself -and still do for other rooms than the living room. Remember that a Squeezebox musicplayer is in fact a tiny PC with a networtwork adapter (and for Classic and Touch also with a built-in display) running embedded Linux and some proprietary software. It is fanless, small and easy to set up as the choices are made for you. What it looses is flexibility and upgradebility. Still cute, though 

Hope this has answered some of your questions.

Per

[PeterSt]

1) I read here that the Xonar Essence does not support some bitrate formats that XX supports. Does this mean that XX is not any better than e.g. foobar?

The response Per already gave is one angle (sound quality), and this would be another :
Suppose I "translate" your remark to the auto-switching to the various sample rates and bit depths XXHighEnd does, while your soundcard does not (just an example), indeed you wouldn't be able to exploit that virtue and players would become "equal" because of that. But the comparison is a strange one, because it would be equal to a Wadia doing 16/44.1 only and a Weiss capable of 24/192, you telling to play normal CDs only (16/44.1) and *thus* the Weiss is not better *for you*. Yeah, read this, and you clearly see the comparison is not right.

So, although my example is not the same as your real life question, it is about the remaining virtues for your circumstance, and the Wadia should win because it is about sound quality (I am not talking about your Wadia, but about my example above only). So, a Weiss just is not better, but it is able to play hires. So what ? better find a "Wadia doing hires" ...

In the end it comes down to what Per said; If you can't do (or auto switch to) all the sample rates etc. XXHighEnd can do and your material will have ... so what ? it is about the better sound, and the remainder is psychological. Or buy another soundcard. Haha.

2) Is it really true that you can get bit-perfect rips using EAC? Does that not then mean, without dispute, that buying e.g. a €20.000 Wadia is absolutely pointless, unless you just like the sound that the DAC in a Wadia produces?

My take ? yes. I would say this is a well thought and correct statement. Additionally (and this is the mistake 99% of today's people make) : it is NOT about the conveniency computer playback brings. Indeed, you did not bring this up as an argument, and for that reason your statement is so good.

Of course your "bit perfect" is to be translated to the, say, 20K which is needed to get a Wadia (etc.) as far as doing quite allright, and still we wouldn't know how ultimately good it is. And here is the advantage of the PC, which inherently is 100% (at no cost really) unless the CD is garbage.

We must be careful, because while it was my own idea back then that this would be all, as long as we would be able to have good external DACs (which is a 1000 times more difficult than having the lot internally - in one cabinet), this appeared not to be all at all. Or worse, the error free reading may even not be a real subject, compared to what, well, XXHighEnd shows you. So, this is about all being bit perfect to begin with, while each player sounds different, and with the fun of XXHighEnd making that an explicit virtue (like with the Q sliders, apart from the indirect stuff). Thus, if we only look at the difference this brings - and what evolution (of the player) brought us over the last couple of years, well, this CAN NOT be about error free reading or anything related to bit perfect; it was just that in the first place !

It is the above where all normal hardware will loose. Yes, *will* loose. Why ? because it isn't incorporated. There are just other parameters, and besides it is very hard to apply them in hardware, nobody knows what they are. Ok, but one small person maybe ...
And maybe this is one of the main reasons I work on a DAC myself, which (work) I ever back anounced as applying the from software learned parameters in hardware, because doing them in hardware has a far far more impact on the sound.

3) I read some motivations that a Squeezebox is must have. I still don't understand why one needs this is one has a good soundcard and e.g. foobar or this XX player? What does a squeezebox add to the equation?

Well, with your "background" of the Duson amps (which I have myself, so it's a nice reference for both of us), it is a matter of "seeing through the hurdles", or actually, what the hurdles are;
Before I started the development of XXHE, I theoretically choose a Squeezebox-like application, and with the same (Audio Note) DAC it immediately sounded way way better than my TEAC P1 drive. The theory is good, because in the end all is eliminated what is there to eliminate, and besides the (virtual) story about error free reading, it eliminates incoming jitter. So, here again, where a company like Wadia may come up with *another* 20K for a good external DAC connection (I think they are really famous for that), something like a stupid Squeezebox just doesn't need such a thing at all. And this is the same with my ever bought "network player", really based on the exact same principle, but 5 years ahead of something like the Squeezebox, which is the explicit (audio) means. Notice that these players always use Ethernet for the transport, and this is always normal computer (error-checked) data (careful with "always", because the first applications for transporting audio over Ethernet as audio already emerge).
All the currently available "media players" work like that, or by means of Ethernet connection, or by means of just sticking in an USB pen. The principle they use come down to the same : the data is far away, but the DAC is in the same cabinet as were the "clocking" is. And this latter is 100% jitter related, and in the very end all is about that, and that only.



The problem with the media players, is that they're always bound to a dedicated playback means. So, you can't use Foobar or anything, you must use the withgoing user interface. For the most media players this even means you have to have your TV switched on.
So, no free lunches.

Is a Squeezebox a must-have ? Well, if you only understand what is happening for the various elements involved ... of course not. The point is, such an application has the head started of theoretically being able not to have incoming jitter. And, contrary what people (including Wadia et al) always thought : no PLL is going to eliminate it. But again be careful, at the time PLLs could eliminate jitter, jitter levels were at the micro second level to begin with (ok, maybe nano). But today ? today "we" tend to use clocks that operate with a few ps in its base, *and* applications exist to also output that in net fassion. So, if that is preceeded with 800ns to be recoeverd by a PLL ... I don't know, but maybe it can be brought down to 10ns or so, but never a 1000 times less (up to that few ps). And on a side note, where I myself just implemented the 4ps net (!) in my own DAC, I'm ready this weekend to heat up the iron to make 17fs of that (which is again a 1000 times less !).

So ... things go fast, but the main message of the above is that a Squeezebox has the "opportunity" to work at jitter levels the clock (crystal) implies. Now, the bad thing is (for understandings), that you hear that. Less jitter is just audible, so a Squeezebox sound better. But is it good ? hahaha, it is a most poor noisy device with a most poor DAC design and ... it may sound better at aspects (jitter related).

Very similar is asynchronous USB. That too eliminates incoming jitter (although not all of it), but it doesn't tell a thing about the DAC itself. And again, the designer just has a better start, because he doesn't have to deal with the incoming jitter. The remainder ? well, no difference.

I can tell you that jitter does matter, but all of the other aspects within a DAC are far more important. Those who auditioned my DAC know how good it sounds, but it was all regardless of jitter elimination, because I just left that aside. So, it is only now, everything being actually ready, that *afterwards* I apply the best clock possible, and what comes from it is that it only gets better.
So you see, this approach is the exact other way around, and this is not to fool myself with good sound because of less jitter. A DAC should produce good sound because it is so good for its (ESR etc.) design, and when that happened, less jitter will make it all better.

Lastly, but this may be confusing, what XXHighEnd does is the first steps of letting a DAC itself sound better. Thus, it influences the DAC. This is where software is priceless (well, to me at least). And the fun is, this works so explicitly and good, that it works for everyone. Not if you have a Squeezebox (etc.), because then there's no software involved that *can* influence. So, you could say that software influence is a bad thing, but in the end it makes your DAC perform better. If that were not so, it would be the other way around, and XXHighEnd would sound the worst of them all. But I never heard someone say that ...


I hope this gave you some insight, or a good base to start thinking and workout things yourself further. It won't come easy and it won't come fast, but you - and we all will get there. I too keep on learning every day.

Peter

[PietPara]

Fantastic Per and Peter!
Thanks for your insights. It is still a lot to digest and for Peter's story I amnot sure I can follow it all, I think due to my lack of the technical knowledge about DACs, jitter and so on.

In any cae, I am happy to hear there is someone else with Duson!!! 
I recently switched to a pair of new Dynaudio Audience 122 speakers after 23 years (!) of listening to my KEF Reference 104/2, which I've always liked but also always lacked some stage and the crystal clarity of Dynaudio's. And now I am enjoying my own renaissance in musical fidelity after it had been dormant for a while. Whence my current enthusiasm to embark on a comparison and knowledge enhancement quest.

I'd like to come back to the topic of DAC versus all-in-one CDP. I know a few people with a lot of high-end experience and they all say that they "hate" pc-audio, because in their listening experiences they have always found it to be inferior and sounding more digital or cold and less intimate as, what they'd call, the real thing, a CDP. An example is that when listening to an Ayre setup where a comparison was made with the CDP and external DAC being the exact same as in the CDP, and playing the exact same CD (obviously one ripped to the PC) the sound was clearly still inferior. Just like a lot of people have a forced hesitation to believe in digital (some die hard phono lovers) there is also an army of people that just despise PC-audio. They would even say that it is even much less convenient (due to all the setup issues and what not). I personally believe that a lot of this is caused by suggestion, where the brain influences your perceptions and emotions due to your knowledge. Suggestion plays a strong and important role in experiencing high-end audio , which is why the term "sounds better" will never be an absolute statement.

Anyway, I'd still like to learn how to more or less discuss and maybe convince such people that are supposedly so against it. Knowing more about the techncial facts helps, although in the end it will indeed have to "sound ebtter" for that person.

I have experimented 1 demo time yesterday with the XXHE player. It was quite annoying that the player just suddenly stops, but anyway, I think I did hear a nicer, clearer sound. I want to test much more, but with the player quitting like that I cannot really get into a concentrated mode, that I need to make a good comparison. That's a real pity. Furthermore, I see a lot of parameters to play with (e.g. Q) and hope to find some explanations here.

I am lovin the talks.

all the best wishes for easter!
PietPara




 

[PeterSt]

I want to test much more, but with the player quitting like that I cannot really get into a concentrated mode, that I need to make a good comparison. That's a real pity.

Well, I hope it is not a stupid remark that this is because of using the demo version ?

I know a few people with a lot of high-end experience and they all say that they "hate" pc-audio, because in their listening experiences they have always found it to be inferior and sounding more digital or cold and less intimate as, what they'd call, the real thing, a CDP.

Very recognizeable, and in the end the exact reason why I started XXHighEnd. And I must say honestly, having a good player (for SQ) is not all there is to it. As a matter of fact there are many things which can be done wrong, and right now I don't even know them anymore. What I do know though, is that when you start this all, you don't have a reference other than your CDP, and however it is when the first sounds come from the PC, you really won't know what to tweak to go to what direction - and worse - you won't even know that there *are* things to tweak.

I won't be able to explain it (not by means of "just writing a short story"), but chances are very fair that if you remove your preamp it will sound horrible. What does that mean ? well, that something (or near all) in your system just *is* horrible. I mean, at removing the preamp (but take care of proper impedance matching in the first place !) you will be removing a huge filter. Thus, if it now sounds worse, something just *is* bad. This is a pretty simple test, although you won't be knowing what to change next. On the other hand, if it sounds better, you're on the good track, and have learned what something like a preamp does to you. And keep in mind : no matter how good or expensive the preamp is, it destroys. Always. But, this is with the exception of the preamp arranging that proper impedance match or acting as a good buffer.

So, let's stop here (already), because it is the best example or test you can start with. It will give you insight either way, and you will be on the track of learning what PC playback can bring. One thing is important of course : your virtual reference of some (like me) guaranteeing you it can be done. This starts with the playback software, and a decent volume control (and as far as I can tell XXHighEnd is still the only player doing that the 100% right way). So, don't have a decent digital attenuation, and you're lost forever. Why ? because you will need the preamp again.

The main motto is that PC playback teaches you what is good for your system for all elements. Thus, now *knowing* that the player can't be doing it to you (for now this may be just trust you must have), if you remove the preamp and things sound very thin or hars, well, things *are* very thin or harsh. Now it is a matter of finding out why;
You are going to think in directions of "what can't follow". Oh yes, it will be about this for a great deal. Thus, at leaving out this huge filter (the preamp) you will be pushing through more "digital". Higher transients of you want. If you now further think like me, it is fairly easy to think about measureable distortion. Or IOW, the less distortion an element shows, the better it will become. This is not stupid theory, but derived from my own work at improving, which includes measuring. And indeed it appears to be simple : if something measures bad, it will sound bad. Sadly it doesn't work the other way around, because todays OpAmps measure very good (like 0.0001 % THD+N), but still sound lousy. Oh, does the OpAmp sound good to you ? then you have another problem, which is the lacking reference. In that case you could say there is no problem because it sounds good to you, but since it just *is not* good (this is another guarentee of mine), you will be masking your sound forever.

Well, skipping most of the process, with some fairly good equipment elements at some stage you will be over a hurdle, and you will be able to judge instruments for their reality. Notice the huge difference with your ever attempts to let things sound less disturbing. It won't be all *that* easy to arrive there, noting that most probably 99.99% of people are still in this last leage. For me and a few others I personally know this is easy : go to the best high end shop you know, and listen to the horrible sound which the salesman shows you as their reference sound. When I say horrible, I mean that.
So, the real advise would be to get over the hurdle of judging instruments, and leave the "less disturbing" behind you. Easily said than done ? maybe, but the most important message is it *can* be done, just because I have that here. Once you're in the good leage it will go fast. Why ? because now you judge reality and you will be able to tweak towards something which is logic (meaning : letting a violin sound more real is a logic thing (and not subjective btw), while letting something sound less harsh is not logic at all, because a filter already would, but is a wrong solution).

(edit : scratched an unnecessary sentence here)

Peter

[PietPara]

Hi Peter,

thanks again for your sharing!

(The point was that I want to test XXHE before paying 72 euros for something that I don't know whether it is better than foobar for me, even though all of you say it is. Just as I wouldn't ever buy Bose even though there are millions saying it's great! ;-) )

But I'm not sure I understand your recommendations. You say "remove your preamp". Do you mean I should try connecting my sound card's (Xonar Essence ST) output directly to the end stage amp (Duson A10 in my case) and see what the sound is like (using XXHE)? I have done this before (with foobar) and thought it sounded purer but less warm (which matches your statements BTW), but it didn't sound horrible at all.

On a side note: I want to actually do this preamp bypass thing for my 5.1 HT that I am going to build (probably with a Rotel RMB 1565 5 channel Class-D poweramp), just to save me the money of buying a AV receiver.

But then, after removing the preamp and listening to the "purer" or more direct sound from the XXHE player, what am I supposed to do with what I learn from that exercise? Sorry, I don't think I got the point you were trying to make.
 

cheers
PietPara


 

[PeterSt]

Oh, I am sorry that I confused you. Ok, you may be reading this with the objective of deciding whether XXHE is "it" or not, while I wrote it in the context of all those audiophiles being stuck in their CDPs and turntables. So, the "you" in my writing was addressed to all those who don't believe in it yet, but who *will* have lousy sound in the mean time. So, my little story was merely meant to get them on a track towards computer playback, with the undertone of "and it will always be worse at first". This, because so many things will be wrong in the chain.

So PP, it was not meant to be performed by you, but merely meant to pass the word on how to go about with this. It is very general and in the end is not much related to computer playback, but *if* one takes this route, he will *have* to investigate what's going on in his chain, instead of taking it for granted (as good / the best, etc.).

Allow me to give a speaking example, certainly not meant to offend. Not anyone, and certainly not you (which will be logic if you read it ).

Suppose you were able to connect a most "fast" speaker to your set. Very generally, this most fast speaker (think in high resolution coming from it) will be high sensitive in the mean time (it just goes along with it). Now, the first thing you will notice is that noise blasts all over from your C100. Your high sensitive speaker made it worse ...
In the end, of course, it is the C100 doing wrong, so there you go, up to another one. Or hey, leave it out ...
Next thing you will notice is that the high resolution expected from the speaker, isn't all that high. Not that you will know it (without the reference), but try what happens when you replace the A10 with a fast amp ...

The fun is, the changes needed only make it cheaper. Generally this is so because less is more. Resistors make noice, supplying caps make noise, connections make noise. Leave everything out and besides there can't be coming noise from the stuff you leave out, it becomes way more fast of it. And this is key. Now, the more you allow "digital" to pass on to your speakers, the WARMER it will get. But not the warmth you are used to from a sloppy bass, but the warmth of harmonics and everything just fitting.

Those who are here from the start know that I strive for 1:1 as good as possible, BUT with digital as a base. Isn't that a strange thing to do. Still I do, and when I find something which again will be a better representative of the 1:1 thing, it sounds better again. Of course, in analogue this is logic all over, but in digital this seems not.

The foremost important is the source (and not the speakers, haha), because that is where digital is converted to analogue, and when that isn't done properly the whole mess is amplified. Of course, we all knew that, but did we know how bad the CDP is at that ? and did we know how hugely important the current draw is for this (D/A) operation ? No we did not. But if you see how stupid software can incur for totally different sound qualities (compare Engine#3 with Engine#4 to have a drastic one), you will know how fragile all is, and how important that is to good sound.

Ok, I'll quit the blahblah. It's not related to PCI soundcards anyway.
Peter

[Per]

Hi Peter,

"The best amp and speaker setup for digital playback" - that could be the name of a very interesting thread based on your latest post here that really started me thinking.... Hope we can discuss all the aspects in a separate thread as I have som questions for you - and they are not related to PCI Souncards 

Per

[PietPara]

Great stuff Peter!
I think I am getting your message. Actually, I know exactly what you mean now. In fact, the last few weeks I was having a very long email discussion about this topic (digital, pc audio, CDPs versus "the real sound") with two of my dearest "high-end friends". The discussion started because of my statements about PC-audio and in particular the versatility a PCI soundcard can bring to the equation.

[ back to the PCI soundcard topic ]

And believe me these friends are not faint hearted connaiseurs of what good sound should sound like. They will agree with what you say. One of them has a great Pass labs XA.5 amp, which is really a fantastic piece of equipment. The first experiences with that amplifier are like you described. It was like a background curtain falling away. The curtain normally would hold all the musical instruments and whatever sounds flying stereo through the room and give the listener the imprssion that there is a huge soundsstage with depth and all you think you like. But this Pass Labs brings one to reality and shows that the sound does not need to be held up by a curtain for it to be sounding real. With this amp you really hear the band playing in the middle of your room and you can walk in between the band's players. (As a matter of expression of course). For me it has been the first time I really hear an audio set sound completely and utterly natural. I must add, though, that this sound may not appeal to everyone because, as you say, the quality of the source may be worse than you thought and was always masked by the other components in the chain. Not sure if I could bring across what I mean. The next step this friend will have (DIY built) is a pair of Soup Ceramique speakers (also Dutch made!). With every step he is reaching what you are explaining. The only step still missing, due to his digital reluctance, is PC-audio. But he told me that I am there to convince him of it.  
I have to be quick because he would like a Wadia or a dCS Puccini.

BTW, when you speak of sensitive speakers, do you mean these Magico's at >110dB?  

OK, then, I am getting closer to the urge of trying out XXHE, even though it's just with my measily Soundcard and not some brilliant DAC. But I am convinced though that one does not need money to throw around at equipment to get high-end sound.
There are still other ways.

 

PS maybe we should move this thread.

[PeterSt]

Yes, everybody who thinks this is going way off topic right from the beginning, is absolutely right. So, I created a new topic for it with the suggested title from Per.

BTW, when you speak of sensitive speakers, do you mean these Magico's at >110dB?

Haha, almost. Mine are 115dB. You may choose something from here : http://www.bd-design.nl/index1.html

Peter

[Calibrator]

You may choose something from here : http://www.bd-design.nl/index1.html

.... or even from here :...... http://sgraudio.com.au/index.html

if you live on the other side of the world.

Couldn't resist getting a little plug in for aussie know-how and leading edge ingenuity 

Cheers,

Russ

[PietPara]

Looks like good stuff those horns. I still do prefer those Magico's, just for shear majesticness of them. What a construction. Anyway, just dreaming... probably will cost around half a million a pair or so.

Those BD-Designs, though, I see that they don't include any bass drivers, and go only down to around 200Hz. That means you need subwoofers. So far all my experiences with subwoofers is that it just doesn't make it. Especially people with just 1 subwoofer, that's got nothing to do with the origins of the recording. There are people argumenting that with such long wavelengths you won't hear stereo anyway, but that is just a bad excuse, since the uniformity of bass in your room increases with more bass units (standing waves and better room correction). Anyway, I'm really a sucker for good bass representation, and have not yet heard a qualifying setup with subwoofers. It always leaves a big gap of sound between what your speaker delivers and the sub. It's too difficult to match to each other, I think.

cheers
PietPara



 

[Jack]

Maybe look a little closer at the BD site.
He makes his own 15in bass driver which goes as low as I need!
No sub needed, but available for HT etc.
Jack

[Per]

Time to get some momemtum in this thread. Peters new dac is coming (soon) and we all need to know what to plan for dream about to keep up with the level of soundquality hopefully coming from the NOS1 DAC.

So, as I was the one asking for a new thread to discuss the ideal amp and speaker setup to release the full potential of digital playback. I still remember the excitement when I was going to listen to the first CD-player back in 1983 - just after the Danish release of the medium. I was invited to hear 2 (two!) players, the Philips CD100 and the Hitachi CD player, I don't remember the model. We were all kind of disappointed, but the CD100 did best and was the most natural of the two.

Well, is HAS improved since then, but I have a strange feeling that somehow we have not hit the potential on the digital side - that is why Peters talk about his dac and what (he thinks) matters in digital sound get me more and more convinced that he is right (the future might prove that I am wrong, but then Peter will be more wrong than me  What really made / makes me believe that we are NOT getting the full potential of digital playback is when a digital recording (maybe 10-15 years old) can be released on vinyl and then beat cd playback. The only reason for something like this happening is that we / the industry has truly missed something. I know, that some explains it by "the conversion to analog / vinyl adds a pleasant kind of distortion and ..." but  still we have to feel disappointed.

Maybe we - like Peter says - HAS to focus on parameters in the amp / speaker setup that is MORE critical for the release of the digital playback potential than it is for analog playback. In Peters book this is SPEED, SPEED, and SPEED. First I did not quite follow him, but maybe he is right. If SPEED has to work right we need PLENTY of power. CLEAN power that is, if I am not totally off track.

And now to my question to Peter and you guys / girls (?)

Would very fast amplifiers that can swing great voltage with ease (like huge Class A power amps) and planar speakers with super light mylar speaker membrane be the ideal setup? Or what about tubes and electrostatic speakers?

From your experiences: What amp and speaker setup favorizes digital playback? If we can answer this - we might be on track to find the best combination / setup for digital.

Just my babling away. Now I have pain all over  (due to my muscle disease) so I better stop and lie down. Best to you all. Hope here is fuel for a good discussion. I will join in when I feel the strength to read / write again...

Per

[AUDIODIDAKT]

Just reading this topic again,

I think we are on the brink of a new age into digital playback all together.
The word will spread like fire, I dare to claim publicly,
There is no better CDP on this globe that outperforms a the decent pc driven by xx-HighEnd.

I can only imagine that when NOS1 releases a new standard into getting digital into analog emerges
This will redefine standards (and possiblities) for amplifiers, speakers and the whole chain behind/between it

So I dont hope it will just end here.

If I am correct we a working a kind chain here:

1. The whole subject about how to get the music ON your PC (EAC, DBpoweramp, Cuetools, accurateRip, audiochecker etc etc.)
   Maintaing a music database - pretty well under control

2. How to get music OF your PC, using a well build and tweaked PC and ofcourse xxhighend software
   And get it into your DAC - pretty well under control

3. next point, DAC --->> Peter's NOS1, will get it into proper analog, all explained pretty well by Peter his nice posts.

yea, IMO lets go on to nr 4, please


What I dream about and foresee in the near future, is when the whole sytem works as one chain.
And can be controlled by PC and software (xxhighend)

See it like this
for eg. a cd-rom drive has an certain drive offset--->> you know this offset (by reference)-->> you correct it before procedure-->> output will be zero offset (thus original)

Now see *this* in your audio-sytem,
with PC-playback you will be able to correct all certain problems like input/output differences in amps/speakers/crossovers/room-correction etc.
By knowing (measuring) the difference between input and output you can theoraticly correct this by software means (thus PC) before procedure (or coverting)
And get "zero offset" (thus original) out of your speakers.
This is tweakers heaven, my man !
No CDP can do that, rest my case.

Peter,

Can you tell "us" maybe a little more about the future after NOS1.
What further plans do you have or not have, hope you keep the community alive.
I see this all just as a decent starting-point.

Do you think you came to the end of your project or will this open up more projects.
I can say you made your point against the "standarized thinking" of how things should be approached.

Will Phasure make more products and/or projects available ?

Or do you take a long vacation ?


I dont believe you just can stop being an AUDIOPHILE 

Roy

[PeterSt]

Reading this from of Per's post, it all seems so easy. But I am afraid it is not;

Speed, speed, speed, yes, but still the best measuring OpAmp doesn't sound (please remember, - to me).
An electrostat may have great speed, but maybe not enough power.
If a high frequency unit works at the proper level, that doesn't tell the bass driver does as well. Let alone what happens at the xover point(s).
When a DAC measures with good THD+N figures maybe it can't drive a main amp (so there go your figures).
When your speakers measure good they may be of less efficiency, and the good measuring amp to drive them may be unaffordable.

And don't think I know it all. But, I try to listen carefully and I always try to reason out why something sounds as it sounds.


If we look at the 1:1 concept of things - meaning that once all is passed with the least distortion possible, currently there is one element in this 1:1 chain that doesn't fit the bill : the analogue stage in the DAC with the OpAmp. I just can't think of why this so good measuring "thing" (disputed by many) won't sound. So, in order to make you crazy once again, this is my experience with it :

In our systems, we often judge the merits of it by cymbals. This may not be for all, because cymbals must first sound good enough to "do" with them what I am about to tell. Otherwise they are merely distorting, and they had rather not be there.
And in between the lines, when you are working on "good sound" at the level I do (meaning with the possible influence, software, hardware), it is a big big challenge to get cymbals right. They can sound too plastic, too light, all the same (no size audible), to rough (all sounding like Chine cymbals) and a few more things. One of these things is quite unknown I think, but it really exsists to my own experience :

The length in time a cymbal sounds.

And this is where my stupid OpAmp solution excelled.
Now try to imagine ... on a random track there's a random hit on a random cymbal, and it takes 5 seconds to die out. Your system sounds wonderful to begin with.
Now I use that good measuring OpAmp in the analogue stage of the DAC, and suddenly the cymbal is audible for 20 seconds. Yes, a completely crazy difference.

In the mean time, this OpAmp doesn't "sound". But why ? this goes against my own theories just following (my) practice; if it measures good it must sound good.

Now, the remainder for now is FYI only, but indicates a bit how I think :


As I said in the other topic the other day : at feeding the DAC with 352.8KHz things suddenly didn't work anymore (while 176.4 did). I seemed to overdrive something. I thought I could hear it was caused by capacitors not performing anymore (mind you, in the digital section of the DAC, and nowhere near of being in the signal path). And I turned out to be right. Fine.

Now, one year after my attempts with the OpAmp, this weekend I will try that again. Hopefully by now I will be able to hear what is wrong with it, or maybe it isn't wrong anymore in the first place. I don't know yet. One thing I do know : they *should* sound right because they measure right. So here's the 1:1 theory and its impact :

If I stuff in the chain something which measures right (think of 0.00001% THD+N) and *that* causes things to sound wrongly, it will have emphasized something which wasn't emphasized otherwise (notice the OpAmp in this case is about gain). This can - IMO - be fed back to something being in lack of drive otherwise (without the OpA). The latter sounds logic to me, because of the crazy long sounding cymbals, which don't sound crazy long without this OpA solution. And the worst of all is : cymbals DO sound long. Try it !

If this isn't about something that gets emphasized, it should be about a feedback of some kind of the device itself. Back into the transformer, while that transformer has other SQ related stuff connected.

But there is a way more creapy possible reason :
Back at that time (a year ago) I was using or no filter at all (this smears to name one nasty habbit of that), or I was using an SRC (just plain wrong), or I was using the common filtering (if AA (now called AI) filtering was introduced around a year ago, it will probably have been that). In either case, all of these distortion types smear and also react to eachother (so the smear smears). And, besides the long sounding cymbals, if I recall one property of that OpA experiment (against all odds as I thought at the time) it was sibilance in woman voices.
And did I mention the higher jitter clock I was using at the time ? another source for smear.

I can add to this, that all what can be wrong with the above mentioned subjects, will be about high frequency products ...

And NOW I am going to add that device which can drive and has sufficient power and SPEED to ... emphasize just that.

By now I seem to have already convinced myself that the OpAmp today will be working. But many things have changed, with as the most important (for this subject) the Arc Prediction Upsampling and probably a low jitter clock.


Well, you see ? this is how to build "a chain". This is quite contrary to removing a device which doesn't sound good (and mind you, devices go up to resistors !). You'll have to *understand* why it doesn't sound good, or otherwise you may be left behind with a not optimal system.

It all takes quite some learning and experience, plus the not-wrong attitude. I mean, it is maybe two months back only that I said that OpAmps unconditionally don't sound good, no matter how good they measure. But it is completely against my own thinking and experience (except for this one then), and it is difficult to sleep on it.
I am not sure how this one will turn out, but supposed I get it to work now (good sounding), there is a definite rule to remember for the future for everybody :

When an element is added to the chain, and which element measures at least better than the remainder of the chain while the sound gets worse from it, don't remove that element, but keep on looking for something else which cannot cope.
"Cope" is in the line of speed, but with digital all just *is* about speed (think possible transients, which no vinyl can follow).

This should be sufficient for the weekend !
Peter

[Telstar]

And now to my question to Peter and you guys / girls (?)

Would very fast amplifiers that can swing great voltage with ease (like huge Class A power amps) and planar speakers with super light mylar speaker membrane be the ideal setup?

This is my recipe. Horns and chipamps can be another one. Etc.

[manisandher]

I'm currently driving a pair of Quad 2805s (planars) with a Pass Labs Aleph 4 (single-ended, pure class A, 100wpc... and lots of heat!) and getting a really nice sound. In fact, the most 'coherent' sound I've ever achieved from speakers.

I think this is one (perhaps of many) recipes that actually works... thought not if you're into raves.

If only the Quads were more efficient, I'd be driving them with my Berning Siegfried.

Mani.

[manisandher]

Back at that time (a year ago)...

Wow, it's amazing how much your thinking has evolved in a year!

When an element is added to the chain, and which element measures at least better than the remainder of the chain while the sound gets worse from it, don't remove that element, but keep on looking for something else which cannot cope.

When you refer to 'measurement', what else are you measuring beyond just THD?

E.g., how important is slew rate in amps? Here's what Nelson Pass thinks (taken from the Aleph 4 manual):

"The slew rate of the amplifier is about 30 Volts/uS under load , which is about 30 times faster than the fastest signal you will ever see, and about 100 times faster than what you will be listening to."

And yet, if you look at the specs for something like the Spectral DMA-360, the slew rate is quoted as "600 volts/microsecond"!

Thoughts?

Mani.

[Telstar]

E.g., how important is slew rate in amps? Here's what Nelson Pass thinks (taken from the Aleph 4 manual):

"The slew rate of the amplifier is about 30 Volts/uS under load , which is about 30 times faster than the fastest signal you will ever see, and about 100 times faster than what you will be listening to."
And yet, if you look at the specs for something like the Spectral DMA-360, the slew rate is quoted as "600 volts/microsecond"!

Thoughts?

It depends first of all on the fastest signal that will pass through the amplifier. with the NOS1 that's 352khz.
Then on the maximum power at which that signal will be amplified.

There are some simple formulas. in short the 30V/us are good only for low powered amps (<40W).
The classic formula considers 20khz as the maximum frequency containing music signal, which is not the case with upsampled and unfiltered source.

Unfortunately the slew rater measure only vertical speed. I haven't seen any (way to) measure the horizontal speed. But it's the only bit of information that we have that gets close to the definition of speed.
The open loop bandwith is another one, but almost impossible to know unless you can simulate or measure the circuit.

PS: high-efficiency planar(or ribbon) extended midrange is the goal for my next speakers.

[PeterSt]

I haven't seen any (way to) measure the horizontal speed.

This isn't necessary either, because it is a state (like continues 6V). BUT :

There's always ringing, and here is a key which influences the projected vertical speed again, *as* it does with that "6V state". Thus :

Before any transient can start (but think of a real steep one), preringing is needed. This is physics. But, it depends on the power how long that will be (the more power the less). The same happens when the 6V is reached. Now we have post ringing;
What I have seen from my measurement post ringing can be virtually endless. Maybe in electronics some kind of feedback exists to get the state as quickly as possible stable, but the feedback by itself will have to be very smart (and super fast) or otherwise it creates the "ringing" by itself.

We see the above when looking at squares, but of course these kind of things happen in real life music all the time; You may recall my measurents actually showing these things. This one is always the best example : Re: Measuring XXHighEnd ... which as far as I am concerned is one pulse with an after swing. Notice this is just music data and no test signal.
I happens all the time, and is not good for the best sound.

Of course, when we'd have these things under perfect control, sound may start to sound very analytical. But on the other hand it should sound as intended.

The foremost thiong for me remains : when this would be output from an OS DAC (or with normal filter anyway) such pictures would be a complete mess with no reference at all (I tried it). This doesn't imply you won't here the gueste of what the artist intended. But it can't be more than that. You can just see it ...

Peter

[bhobba]

Well I don't know if its the best but after a lot of careful consideration I have chosen my amp, louspeakers, and DAC

Loudspeakers - Aether Audio Timepiece Mini:
http://www.aetheraudio.com/
http://aetheraudio.com/timepiece_mini.html

And Bob (that the hippy looking guy in the picture - great bloke IMHO) has a new development that allows a crossover at 527hz.  Both the frequency and phase response are virtually ruler flat and talk about power handling and speed - well their speed rivals electrostatics while their power handling is simply unbeatable - they handle 1K monster amplifiers with ease - no distortion whatsoever.  Now I won't be using them with anything like that but it is good to know they can handle it.  Don't take my word for how good they are.  See what the great audio engineer Jim Merod, who has heard everything and anything in his time has to say:
http://www.positive-feedback.com/Issue30/sptech_2.htm


Basically as one person who heard them said - they are shockingly naked.

For amp duties I have chosen the latest design by Hugh Dean the Maya.  Its very new and he hasn't got it up on his site yet but here is what he does have up:
http://www.aksaonline.com/products/products.html

The Maya is supposed to be a touch better than the Soraya and that is some feat IMHO.  Check out: 
http://www.diyaudio.com/forums/solid-state/108756-soraya-cb105.html

As destroyerx wote (what a name): This Soraya is not more, not less, than the best.  That must be some statement because he has had '49 years buiding audio amplifiers, more than 4 thousand units made and tested'.

For the DAC I have chosen the new DAC2 from Wyred For Sound:
http://www.wyred4sound.com/webapps/site/74030/117839/shopping/shopping-view.html?pid=457975

I suspect some dacs like the overdrive and new tranquility dac may have a slight edge on it in sonic terms (as judged by some) however I chose it because it uses the Saber 32 bit dac which headphone guys are really impressed with.  They use words like - removes a layer of grit from CD's.  They have heard nothing to beat it and that is good enough for me.  Both the overdrive and tranquility are nos dacs.  I have not had good experiences with nos dacs.  All the dac's I have heard that I like were oversampling.  Others however have commented they were a bit too clinical and detailed - they liked the more laid back sound of nos DAC's.  I am the exact opposite - to me clinical and detailed just sounds REAL.  I may however upgrade later.  Oh and the tranquility DAC is only 16 bits so a digital volume control which I also like is out.

Thanks
Bill

[PeterSt]

Hi Bill, nice story and thanks for sharing.

Maybe it's allowed to tell that this topic is supposed to be about theoretical best stuff and less about our subjectiveness of hearing ? Thus, this isn't about a bunch of head-fi people noticing layers disappear after using OS DACs all their life with layers to begin with (in the middle of an environment where you're shot when you dare to talk "NOS" ). It also isn't related to what I find sounding best. Nor anywone else.

It is about what *IS* the best. This is an absolute thing and not subject to what we (want to) perceive from it. But :

Even when listening would be involved, it is my firm opinion that this is not subjective IF you have the reference first. And the fun is, today's reference allows us easily to listen to the instruments how they are, or at least I can do that here (with my Phasure NOS1). The world of audio really changes from that. What I mean is this (and it has been told before) :
Without the reference, we will be used to let our brains speak for whether we find something good or not. "Good" means acceptable. Telstar might jump in to confirm this, at a comparison with an ESS implementation (24 bit) we did a year back or so. Thus, without the reference the owner of the ESS (which was not Telstar) was completely satisfied. He had no other reference. Telstar was not satisfied because one "hears" better than the other. Also, at that time Telstar didn't care much about OS vs NOS. I could hear it was wrong all over, because I did have a reference (for the better, which was my own DAC). Now :
As soon as we hopped over to my DAC, the ESS seem to fail all over. It really needed 2 seconds of listening (and I mean this) to judge the ESS as broken.

I can tell you, at that time my DAC was  just "NOS", although technically very good. But nothing about the needed filtering which is in there today (and which in the end is Arc Prediction from XXHighEnd). So, my reference already changed again, and something like the ESS would be the worst (I am just using superlatives because it is so).

Keep in mind, the last story was about listening not being subjective at all, because an instrument is not subjective (if you only heard it live once) while todays capabilities of audio allow us to listen to that instrument. So, it is the instrument (good playback) or it isn't the instrument at all (bad playback).

Once we are in the stage of listening to real instruments, we are still far from being there, because there's also the foot-tapping or tear drawing mechanisms (analytical vs emotion). They are hard to control, but *are* a phenomenon which can be explicitly tested. And the worse is, emotion can work just the same with a -30dB noise audio cassette tape. Things like noise even seems to be a prerequisite sometimes (vinyl). So, it is about bringing this all together, and luckily the Engine#4 version of XXHighEnd contributed a lot to the emotion part.

Back to the real thing : To my findings - and I am really explicitly working on that for quite some years now - all works out the best if digital is treated how digital should be treated. This is not massive oversampling and (today) it is very easy for me to proove where it destroys. Mind you, I could always do that, but before this was without alternative. So, before this was about NOS being wrong as well, but in such a completely different way that it came down to a subjective choice what was better. NOS with heavy harmonic distortion (and I mean HEAVY) or OS with EVERYTHING wrong, but unmeasureable (because the used filtering is theory only and doesn't work in practice). But this changed since decent filtering can be applied to NOS, and now it only has benefits.

All 'n all I am trying to pass through as 1:1 as possible that stupid digital redbook data, which can't be 1:1 to begin with because of a too low sample rate in redbook. Still the 1:1 principle counts, because in every aspect where I achieve an improvement regarding this, it sounds better (not subjective listening !). This is important to know, because we'd expect it the other way around. If something sounds "digital" it must be rounded to more analogue. WRONG. If something sound digital, something can't cope and you won't be hearing the bits or anything, but just wild distortion.

Two or three days ago I talked about the OpAmp thing. I guess that was a less blabbering story than this one. But I can tell you now : I was right, and it now works. Remember, the OpAmp "gain" wasn't change a bit. The rest did though, and is now ready to receive the high frequency data which it couldn't cope with before. It's that easy, but you have to know where to be to get there.
I think it is very important to realize that when e.g. your speakers can't cope with high transient data (because that is what we're striving for) it is not that you won't hear the transients as should, but you'd merely have distortion;
If you could look at a wave which can't follow what's intended (which is what I could measure as well) it is easy to see what happens with it. It is not only the first cycle which can't reach its peaks, it is merely the second and further which become a great mess. Thus, before the first cycle is at its peak, the now negative voltage draw will try to pull the wave down but that works out too late and in the end a sine has become, well, a nothing. Noise. I am just talking electrical here, let alone mechanical loudspeaker diaphragms.

In the very end it is just about measuring, but the trick is that it is about the whole chain. Thus again : when something in the front-end measures better, it may sound worse and we say "see, measuring tells nothing". WRONG. We may try to feed our speakers with transients which weren't there before, and because of the speakers can't cope it sounds worse. And keep in mind, this is just a rough example, because it really happens from the one electronics part to the other as well.

Blahblah Peter.

[Telstar]

Just two quick comments, Bill.
1) Try to listen to something before you buy it, unless you are totally sure that your tastes and judgement closely matches the one of the reviewer.
2) The speakers are rather unefficient, that requires lots of power to sonorize a medium room (the Soraya is only 105W).

I dont think you are up to a big disappointment, but a not ideal match is very likely.

[bgjohan]

Peter

You have stated that sufficient speed in the amplifier is essential for excellent sound.

I am just wondering if running a stero amp in balanced (mono) mode is a way to increase speed, or this just adds power to the amplifier whithout any impact on the speed.

Bjorn

[Telstar]

Peter

You have stated that sufficient speed in the amplifier is essential for excellent sound.

I am just wondering if running a stero amp in balanced (mono) mode is a way to increase speed, or this just adds power to the amplifier whithout any impact on the speed.

Bjorn

Actually is none of this, although using a mono (or dual mono) amp has several advantages.
It's how the circuit has been designed and build.

[PeterSt]

What Telstar says. I GainClone - and how it should be built - is a good example of it. All the signal wires involved can take a few mm only really. And even short wires me contribute to speed (think of "elasticity" at a longer wire, or better : reaction (stiffness) at the other side). Of course, the longer wires (like for interlinks) can't be avoided, but it is here where "drive" (power, or actually current) comes into play.

Don't misunderstand me with the example of the GainClone (or the real Gaincard for that matter). You may gain on the short wires, but you may (or will) loose it on the high distortion such a chipamp produces in the first place (think near 1% at full gain).

Bringing all the virtues together is the big trick, and this starts with the "design" and (knowledge of !) available parts.

Peter

[bhobba]

Hi Peter and All

Back to the real thing : To my findings - and I am really explicitly working on that for quite some years now - all works out the best if digital is treated how digital should be treated. This is not massive oversampling and (today) it is very easy for me to proove where it destroys. Mind you, I could always do that, but before this was without alternative. So, before this was about NOS being wrong as well, but in such a completely different way that it came down to a subjective choice what was better. NOS with heavy harmonic distortion (and I mean HEAVY) or OS with EVERYTHING wrong, but unmeasureable (because the used filtering is theory only and doesn't work in practice). But this changed since decent filtering can be applied to NOS, and now it only has benefits.

Thanks very much for posting that.

Although I perceived the sound of oversampling DAC's to be better your point is well taken.  My preference may be for a form of distortion and I have had bad experiences with that in the past.  My first speakers were the legendary Gale 402's which had sealed bass.  When it came time to retire them I bought a pair of LS88's which had ported bass.  Initially I thought the ported bass more full sounding and richer - it had the  more immediate appeal.  But over time I came to realize this was false - it grew tiring.  The sealed bass had a rightness about it which a ported design simply could not match.  The 88's had other qualities I liked a lot that more than compensated, but the bass was definitively a minus.  That's one thing I like about the speaker I am interested in.  It is not ported bass - it is a hybrid transmission line ported  design that its designer claims (and I tend to believe him) gets rid of most of the problems with ported designs.

Anyway another choice of DAC would be Steve Nugents overdrive.  It is more expensive, does not have a remote volume control (which I really like as I have bad arthritis), and does not have a home theater bypass mode.  To me all these are also important.  My gut feeling is I may in the end settle for the DAC2 - at least initially - with the idea of upgrading later.  But I may also just decide on a 'better' dac such as the overdrive or Peters Phasure NOS1 (which right now I don't know too much about but will do a bit more investigation).

Thanks
Bill   

[bhobba]

High Telstar and All

Just two quick comments, Bill.
1) Try to listen to something before you buy it, unless you are totally sure that your tastes and judgement closely matches the one of the reviewer.
2) The speakers are rather unefficient, that requires lots of power to sonorize a medium room (the Soraya is only 105W).

I dont think you are up to a big disappointment, but a not ideal match is very likely.

Thanks for posting that.  This is something that has been subject to a lot of debate over on Aether Audio's forum with a lot of very interesting discussion.  A couple of points first.  The Timepiece Mini I am getting, while audibly virtually identical to the Timepiece (it has 85 db sensitivity which is low) except for a bit less bass extension is actually not too bad in the sensitivity department at 87 db (they actually do one thing better than the Timepieces - since they are smaller there is less refections so they disappear - as their designer says 'thought we were listening to a standard Timepiece that had been tweaked...except that this pair of TPs completely disappeared - poof...gone.   You close your eyes and float away to this enormous soundstage and 3-D image.  Then you open your eyes and the brain just dis-connects.  There is no way on earth such a soundstage and dynamics can come from a speaker that small').  The designer Bob thinks 100W is more than good enough to 'send the willies up you' (his words not mine).  The other thing is the Maya that is Hugh Deans latest amp is 150w - not 100w.  Combine this with the fact I listen at lowish volume levels I think it should be fine.  But to be on the safe side I have corresponded with people who have this speaker and they have checked them out with the Soraya and it is fine with that amp - even at the levels they listen to which is well above what I do.  They like them with 400W Cherry amp monsters.  Thats the thing about Bobs speakers they can take that type of power with zero distortion no problemo - it has to do with the waveguide design he uses (he has to pad down the output of the upper end of the tweeter with the crossover - because of that they can handle god awful amounts of power without blowing them or suffering compression of any kind - I won't use them at that sort of power - but its good to know they can handle it).  Anyway despite the fact the Cherry is a bit cheaper I still prefer Hugh's amps - people whose ears I trust say they work very well together.  Also remember the ear is logarithmic in its response - 100W amp only sounds half as loud as a 1000W monster.

I could write a bit about 'listening' to equipment and I will probably do a much longer post on it one day.  Suffice to say I am not that big a fan of it despite the fact everyone says its what you should do.  One reason is you can be easily fooled - and I have been.  I tend to trust guys who listen to a lot more stuff than I do and who technical skill is beyond reproach.  These are guys like Hugh Dean at Aspen and Bob Smith at Aether Audio.

Thanks
Bill

[xp9433]

Bill

Speaker sensitivity and amplifier power rating are never as simple as people expect. For example a 3-way speaker makes greater demand on an amplifier (both current and voltage) than a single cone of the same sensitivity. It is very easy to overstretch an amplifier and greatly increase distortion/clipping on musical transients. The arithmetic doesn't always work and your ears will tell you easier than the maths will, providing the listener recognises the symptoms.

I can remember at an CES where a new open baffle speaker design (94dB senstivity 8 ohms) was being demonstrated using a respected 35watt tube amp driving the satellites. The music was hardening on peaks, for example when a soprano was singing. I offered the use of a Plinius SA250, which I was taking for review, and the problem disappeared. Linkwitz (one of the speaker designers) measured the output from the Plinius on "soprano peaks" at in excess of 180 watts, and this was in a smallish room. Go figure.

Needless to say I am a big fan of powerful amps rather than the other way around. I believe in amplifier headroom. I have seen too many speaker drivers destroyed with underpowered amps, and many owners suddenly relax with the new "natural unstressed reproduction" on dynamic musical passages when a powerful amp was inserted into their system. Most didn't know they had an underpowered amplifier problem until the stress and distortion problems were taken away.

Frank

[Calibrator]

I could write a bit about 'listening' to equipment and I will probably do a much longer post on it one day.  Suffice to say I am not that big a fan of it despite the fact everyone says its what you should do.  One reason is you can be easily fooled - and I have been.  I tend to trust guys who listen to a lot more stuff than I do and who technical skill is beyond reproach. 

G'day Bill,

that's an interesting approach to choosing equipment, however, it's not one I adhere to.

I would rather suffer the consequences ( if they occur ) of making a fool out of myself, than have someone else make a fool of me. If I am going to buy a wheelbarrow, I like to push it first !

Music enjoyment is a very subjective and emotional game, and while there may be others that share ( to a degree ) your philosophy on sound reproduction and techniques, and can offer some insight as to how certain combinations work together, I think it is limiting to one's enjoyment if you rely exclusively on others to make the choice for you.

It takes a little effort to train yourself to recognise specific aspects of sound reproduction, but it's not beyond the capability of the average audiophile given a little guidance.

Just my 2 cents worth for the moment,

Cheers,

Russ

BTW ... a friend recently suggested "Rapigel" as being very helpful in reducing the effects of arthritis and muscle pain. Widely used in the sporting world apparently. Do some googling about it.

http://www.horsesrus.com.au/products/rapigel.html

 

[PeterSt]

Another 2c attempt from my side ...

My woofers as well as the sub woofers are ported. The ports are there by design obviously, but like Bill said, I think it is fake (low frequencies). I am not sure whether this is in the "there's no free lunch" department, but at a certain stage I closed all the ports, and now my bas is ... more warm.

I won't repeat my ever lasting story about standing waves and more I seem to be the only one keen on (how to avoid them), but if you think carefully about this, it is again the same principle (the 1:1 really is everywhere). So, my subs, officially rated at 12Hz won't go that low anymore. Fine. My woofer, rated at 27Hz won't reach that anymore (I didn't even care to measure it after closing that port), but the bass response gets more outlined from it, and the interaction of the PROPER waves (frequencies) let excel the harmonics - otherwise vanished in the wobbling so called deep bass.

This too is an experience one can have a reference on. Or at least everybody should have heard the difference in order to recognize its importance;
Bass can be low but without a real frequency or "pattern" (harmonics !). Lay your hand on the (sub) woofer's diaphragm and feel a wobbling cone, or feel the repeated pattern of the instrument, including the wood it is built around;
Because not all music and instruments allow the judging of this by hearing, you can always feel it (the relative difference between good and bad).

No, I don't see how a ported speaker design can workout for the better.
All of you who have ports in there, for fun close them for a week or so. IMO (and as far as my knowledge goes on this) this doesn't work destructive for the sound. The port is just there to output longer waves, which hopefully are a kind of in phase with the normal driver output.
If you have a port only (and no normal driver output), leave it open ! hahaha.
But I'm serious. Try it, and next focus on low frequency sounds that maybe have become strings, or a clear synth, or even a tuba.

Peter

[manisandher]

All of you who have ports in there, for fun close them for a week or so.

I tried this with my Wilson Benesch Chimeras for a few months, and indeed, much preferred them with their ports closed.

HOWEVER...

I was using them with active x-overs (internal passive x-overs removed) and could adjust a whole host of parameters (output, freq., order, Q, etc). If you close a port, you really need to be able to readjust a bunch of stuff. Worth trying anyway though.

I've never really understood ported designs - the ones I've heard and owned have generally sounded terrible. I guess it comes down to the manufacturers being able to quote low frequencies in the specs... This is such a shame.

Mani.

[Telstar]

The Timepiece Mini ... 87 db ... the Maya is 150w

87dB speakers in a room of about 90 cubic meters requires 75W for 110dB peaks (peaks are 20-30db above the average).
I think that you have a good headroom with your 150W.

I could write a bit about 'listening' to equipment and I will probably do a much longer post on it one day.  Suffice to say I am not that big a fan of it despite the fact everyone says its what you should do.  One reason is you can be easily fooled - and I have been.  I tend to trust guys who listen to a lot more stuff than I do and who technical skill is beyond reproach.  These are guys like Hugh Dean at Aspen and Bob Smith at Aether Audio.

For me the other way is true.
Got burned once. Got luckier with several peoples opinion rather than a single reviewer. But I'm 100% satisfied only with my ears.

[Telstar]

Because not all music and instruments allow the judging of this by hearing, you can always feel it (the relative difference between good and bad).

No, I don't see how a ported speaker design can workout for the better.

It can be done. it's called overdamping. Very few BR speakers do that (Avalon, Tidal among the ones that sound good) because people likes boombastic bass and also because there is a natural dampening in American houses wall construction (thanks to a friend i finally understood this mystery just a month ago).

Rather than going to closed box (penumatic suspension), a BR can be designed not to boom. I have read some paper recently very enlightening, but i cant find it right now. I'll try to if you mail me for a reminder, Peter.

I prefer even more damped loading systems than the closed box such as infinite baffle (if i had a free room behind my listening room, thats THE way to go) and open baffle (which requires LOTS of space IN room).

Edit: Oh, i remembered one thing about that paper, and it is that low frequency reach is much more important than the SPL in low-frequencies. A very low Q design just sound more natural to the hear even if it is -10dB at 20hz rather than -3, while the first is at -15db at 10hz and the second at -30

[bhobba]

87dB speakers in a room of about 90 cubic meters requires 75W for 110dB peaks (peaks are 20-30db above the average).
I think that you have a good headroom with your 150W.

Exactly - thats what my math tells me as well.  Indeed there is really no need to go above 90db as well since it is generally recognized listening to even that level for extended periods will damage your ears - let alone about the neighborers, quite rightly IMHO, complaining.  I know I got hoping mad when the guy next door at 2.00 in the morning new years day decided to blast his boom box (I won't call it a stereo) full blast.  I couldn't even shout over it to tell him to turn it down to a sane level.  One person eventually heard me and turned it down but they didn't care and turned it up again.  I rang the police who put a stop to it quick smart.

I generally listen at about 80db if that.  This has another advantage as well.  It is generally recognized it is cheaper to get a good low powered amplifier than a high powered one.  Don't listen over 85 db and a 40 or 50w amp will suffice saving you real money to spend elsewhere. 

For me the other way is true.
Got burned once. Got luckier with several peoples opinion rather than a single reviewer. But I'm 100% satisfied only with my ears.

Yes - no one agrees with me on that.  But it has served me well for quite a while now.  Although I have had a few nibbles from guys burned with the conventional approach asking exactly how to go about it.  The main thing is to form a direct relationship with the designer, follow exactly what they are doing, and see what others say about it.  I judge it not just on subjective opinions but on technical details as well.

Thanks
Bill

[bhobba]

Needless to say I am a big fan of powerful amps rather than the other way around. I believe in amplifier headroom. I have seen too many speaker drivers destroyed with underpowered amps, and many owners suddenly relax with the new "natural unstressed reproduction" on dynamic musical passages when a powerful amp was inserted into their system. Most didn't know they had an underpowered amplifier problem until the stress and distortion problems were taken away.

Thats the view of the guys that like those 400W Cherries:
http://www.digitalamp.com/cherry.htm

But even those guys have tired the minis with the amp I am interested in and its just fine - even at the levels they like to listen at.

Thanks
Bill

[bhobba]

Rather than going to closed box (penumatic suspension), a BR can be designed not to boom. I have read some paper recently very enlightening, but i cant find it right now. I'll try to if you mail me for a reminder, Peter.

Although I much prefer a closed box to a ported design even better is transmission line bass of which I have only ever heard a few.  However the guy I plan to get my speakers from is a master at them.  His designs are always hybrid ported/transmission line (that evidentially gets rid of most of the problems with ported designs - I will soon finedout if thats true - but from what other guys tell me it is to a large extent true) or straight TL.  However his TL speakers are supposed to something extra special in the bass department - one guy reports they literally make people swoon when they hear it:
http://www.audiocircle.com/index.php?topic=54021.0

If you are keen Bob can do you a sub-woofer:
http://www.audiocircle.com/index.php?topic=38864.0

Personally though I am not going to bother - at least initially anyway.  I had an Evocatour sub-woofer for many years and ended up leaving it at a friends place when I moved.  He was staying with me at the time and we used to call it our dunny speaker.  When you engaged it you actually didn't hear too much extra stuff but you sure felt it when you went to the dunny.  I may be cajoled though once I get my system fully set up and looking for a little tweak to add that something extra.  But for now I am not too worried about extra low bass.

Thanks
Bill

[PeterSt]

But for now I am not too worried about extra low bass

I estimate that this is in 30%-50% of random albums (I mean, in at least one track of them).
Wouldn't want to live without it.

Here too, this is about "control". Good sub low is never just "low deep sound". It's individual vibrations (say, under 40Hz) up to "shaking" (under 22 or so Hz) because you can't hear it anymore. If you don't hear vibrations from a sub, it is no(t) good. And certainly here, feel the diaphragm. If you don't feel what you'd perceive from the instrument (or sound) involved, it's wrong (for music anyway).

Don't use a sub higher than the normal woofer, and try to cross it as usual (which may mean adjustmend to the woofer's filter as well). Without this, forget it, because you will be destroying your woofer('s sound).

Don't connect the sub in a 2.1 etc. configuration, but just satellite it (from the interlinks when active or from the LS cables when passive) (which goes along with my above crossing etc.).

Never listen to those who tell you that subs can't work for music. But it may be a tough job to get something which is good. Next, it still is difficult, because you'll need to calibrate the sub with the woofer/normal speaker. This depends on the distance from the normal speaker and the listening position, and is about the phase which has to be aligned.
If you think you'd better have two subs for stereo (which is certainly what I would do and have), find yourself calibrating the whole lot for ages. This is very difficult (both will cancel out eachother, depending on where you measure) and you really have to know what you're doing and why.

Never think low frequencies are not directional. They are, when they are "good".

Peter

[bgjohan]

Peter

When you suggested covering the port in a vented speaker, do you mean completly sealing (as with a piece of solid wood) or only partially sealing (as stuffing the port with some foam, as I have seen suggested on a a speaker construction site).

Bjorn

[Telstar]

But it has served me well for quite a while now. 

Whatever works for you.

[PeterSt]

Hi Bjorn,

Foam is what I mean. Old socks usually do. But is must really fit tight, or otherwise is't blown out again (the forces are rather high).

Peter

[stoneman]

I'm currently driving a pair of Quad 2805s (planars) with a Pass Labs Aleph 4 (single-ended, pure class A, 100wpc... and lots of heat!) and getting a really nice sound. In fact, the most 'coherent' sound I've ever achieved from speakers.

I think this is one (perhaps of many) recipes that actually works... thought not if you're into raves.

If only the Quads were more efficient, I'd be driving them with my Berning Siegfried.

Mani.

Hi manisandher

I use a pair of Quad 2905s and I supplement the bass with some (very old) Celestion 6000 subs.  Since the workload of the low frequencies is managed by the subs, the system can go pretty loud; despite the Quads being relatively inefficient.  It also means it's quite difficult to short the panels out.

regards
Andy

[manisandher]

Ooh, I'd love a pair of Celestion 6000 subs.

But not for my 2805s - I'm quite happy with the balance I'm getting, and also just love the speed of the bass and the general coherence of the sound... although I must admit it'd be nice to have a little more punch low down.

No, I'd love to mate the 6000s with the SL600s in my office

But you've got me thinking - high pass the 2805s around 100-150Hz, use a pair of 15inch drivers in sealed boxes down to ~50Hz and a pair of subs <50Hz. The only thing I'd be worried about would be speed. Surely no conventional driver could keep up with a 2805/2905s. What's been your experience with the 6000s?

Mani.

[PietPara]

Hi,
if we speak of reproducing pure 1:1 sound, does anyone here know anything more about Bybee Technologies and their Quantum Purification technology used to purify the sound traveling through cables?

See here: www.bybeetech.com

Sound interesting to me, but somehow it also smells like someone selling a Heisenberg compensator.

Maybe some of you have heard about it and know the truth.

cheers
PietPara
 

[Calibrator]

Smells like an oily substance that eminates from snakes.

I suspect the only purification that will happen, is to your wallet, as your hard-earned money is removed.

I had a good laugh while reading thru that site so thanks for the link.

Cheers,

Russ (living in a real world)

[stoneman]

Hi manisandher

(Re experience with the 6000's)  To be honest, I dont really know! Sounds a bit daft, but I've had them for so long (18+ years?  Not sure now) that I dont have a separate reference point. 

I cant say I notice them being slow vs the Quads - its all quite lifelike (well, as lifelike as audio systems can be).  I suppose since I run them below 40hz it doesnt really make that much difference, they just 'sit' sonically under the Quads filling in the last bit of the spectrum.

BTW, I know this is criminal but the SL600i's have been relegated to satellite duties from the Home Theatre system when I'm playing music in the lounge off that PC - one on the balcony and one in the dining room (on the original Foundation stands too!). I couldn't bring myself to part with them.

In my office I use a NuForce Icon amp into a pair of little KEF 3001's (I think) that sit on the desk on either side of my monitors (3 x 30" Dell) and one of their matching do-nut style subs near my feet, works quite well!   

regards
Andy
     

[han]

Hi everyone!

It's quite a thing, getting this ridiculously expensive Pre-amp out of my system.... But since a few of you guys out there, claim that it is the only way to purity (and yes, getting a part out of the chain should be purer) i gave it a try.

My NAT Utopia pre is very very natural because of the very few elements in the signalpath but there is a difference indeed. My sound is even more open, bit more stage and so on. Seems to be a good step. But this experiment leaves me with one question and i hope that one of you can give me some answers. My listeningroom is quite small right now (30 square meters) but it will shortly be much much bigger (over 100 sq. m. and some 4m high). The outputlevel  (volumewise) from my 92 Db Kaiser Kawero's  is just ok for now (i mean it plays just loud enough), with the pre-amp i could blow my windows away.

I wonder if this setup will be able to play loud enough for the much bigger room

And second is there a way to digitally manage the volume or would for example the Weiss 202 DAC solve this "problem" because of it's volume knob and the 4 coarse settings?

Hope that someone can help me out. If so, i have a qwonderful pre-amp for sale.....

Thank you in advance!!

Han

[Nick]

Han hi,

Nice looking amplication you have. I am using 103db horns driven by 30 watt amps. The speaker have active bass drivers with 150 watt per channel. At the moment I listen in a galleried hall landing space (not ideal) of 80 square meters by 5m high, so quite similar in volume to your new room.

In my case I have to drive the speakers quite hard. With 92db speakers your system will need to play at quite high amplifier outputs fill the space. I listen from about 6 meters and have tuned the bass crossover response up by about 5db. I still do not get the balance of smaller rooms. My guess is that your near field listening will be ok but listening at greater distance may change the balance. Best to give it a go and see.

Regards,
Nick.

[PeterSt]

Ok, let's see ..

My listening room is 12 x 8 x 3 = near 300m3. Yours (Han) is 400m3. That's 30% more.

My amps are 27 Watts. But, 27 for mid-high and 27 for the bass (the mid-high acts as a pre-amp for the bass).
With my DAC -which outputs 1.5VRMS- I play at -15dB for the softests albums (normal would be -27dB). Btw, of course I use the digital volume from XX - which I thought was an obvious solution, or I don't understand the question about that).
My speakers are 115dB.

The above really puts the room under pressure, and listening level is easily 90dB everywhere in the room.

While my problem is that I don't know the relation between needed dB and the volume of the room (hopely others or you can take the time to google it out), it would theoretically come to this when we only use distance :

Your room is 6M long, mine is 12. That needs 36dB (6dB per meter) more.
Your speaker is 92dB, mine is 115. That needs 23dB more.
Your DAC will be 2VRMS, while mine is 15, that needs 6dB less (not 100% sure about this one).
I play at -15dB at the most, so that's 15dB less. But, a that soft album can be boosted by 3dB (XX will do that). So 18dB less.

So we have 36 + 23 - 6 - 18 = 35dB you need additionally to what you have now, but which will be at the low side, because of the higher ceiling. Maybe the 30% because we already took into account the sqm roughly.

Even if with pre-amp your windows go out, I don't think a preamp will have that gain. You'll need a bigger amp ...
Maybe it is not fair to use all that volume, because at a normal listening distance things change drastically. But it is my opinion that the room needs to be under pressure if you really want to do it well, so YMMV. And like Nick suggested, you won't be able to listen nicely back in the room (might be the kitchen) which in my case really doesn't matter. SPL doesn't drop much, but first you have to have it loud enough.

Maybe first look for an amp, and next for a DAC ... Or keep on using the preamp and see how far you come with it ()
Peter

[CoenP]

I don't want to sound like a tutor, but in a free field a point source looses power-to-three acoustic power with distance. So for a DOUBLING of the distance there is a 8X or 9db decrease in acoustic power.
This will correspond with the nature of low frequencies that diffract and radiate in all directions. High frequencies bundle (as result of conus or horn geometry) and will loose half space power of 4X or 6dB per DOUBLING of the distance in a free field. This is worst case since bundling is expected to lead to less than halfspace radiation. Consequently the difference betwee listening at 4 or 5 meters is smaller that between 4m and 8m!

Note that the room also reflects the soundwaves, wich is especially important for low frequencies which refections add to the LF power in the room. Since the ceiling is high, power is lost there. Hard walls also refelect the mid and high freqeuncies, so dependent on the radiation pattern of the speaker these refections play a role or not...
All in all it isquite little hard to predict the actual acoustic powerresponse of a given loudspeaker in a room, I believe it to be less than 6dB per doubling of the listening distance and compensation for LF is likely.

I would expect the bigger room and further listening distance needs at most 6dB more loudspeaker output than the room of Peter, however the difference in loudspeaker efficiency is vast and more important. This is 20db or 100 times more amplification power for the smame acoustic output. You need a very big amp and speakers that can do away with the heat and distortion at high excursions.
You need in the end either a smaller room, lower listening levels, closer listening or more effcient speakers (arrays or horns) to play loud.

regards, Coen

[Flecko]

I have an eye on this one:
http://www.hs-devices.com/english/PDFs/Datenblatt%202510c.pdf
It has even much lower distortion than the BMS 4590!
my2c
greetings Adrian

[han]

Thank you so much for all your kind advise!

Gonna plough trough these advises over and over but I guess that "the proof of the pudding is in the eating" so to speek. I'll keep you informed about my findings in the new situation.

The main issue and answer is that with a bigger amp problems can be solves if they occur! Hmm, did one of you ever take a look at the NAT Transmitter monoblocks? They deliver 120 W pure class A Single ended.... Nice pair!

I'll try to get my mortgage a bit higher!

Thanks again, regards,

Han

[Telstar]

The main issue and answer is that with a bigger amp problems can be solves if they occur! Hmm, did one of you ever take a look at the NAT Transmitter monoblocks? They deliver 120 W pure class A Single ended.... Nice pair!

How many paralleled devices? silicon or vacuum?
120W SE class A is really an achievement.