USB audio cracked... finally!

[fas42]

1 minute ago, mmerrill99 said:

It's the ferrous impregnated dielectric that affords the reduction in high frequency noise, not what is typically called the shield in a USB cable.

 

It reduces both the intrusion & the emission of RFI.

 

Could this change the noise profile reaching the receiving end of the USB cable?

 

Does it matter how the noise arises on the cable?

 

Okay, this looks like another case of using a semiconducting layer - the ferrous impregnated dielectric - perhaps to perform some 'magic'.

 

Perfect insulators are not always the best answer - a little bit of conductivity goes a looong way ...

[fas42]

2 minutes ago, mmerrill99 said:

I wouldn't call it magic - i know you are aware that it's a well known property of ferrous material, clamps, etc. so there's no unexplained mode of action. Just saying this in case readers think you are stating that this is some unknown audiophile 'magic'

 

Gee, the magic of using '...' quote characters has worn off a bit ...

[fas42]

6 hours ago, manisandher said:

I was really enjoying the technical discussion earlier. For those of you who are technically-minded, could you offer your thoughts on the following please?

 

How can a USB cable make the system sound more musically satisfying?

 

Personally, I think it's all about timing - reflections and ringing in the USB cable that are inevitable at the RF frequencies that USB is working at.

 

Mani.

 

It's mighty simple. Musically satisfying is the outcome every time if there are no disturbing artifacts in the sound - these artifacts don't necessary have to be obvious, in the way that pops and crackles are in playing LPs; digital sound anomalies may be essentially 'invisible' to one's conscious mind, but if you remove them, and then re-insert them, then it does become obvious ...

 

If a cable, or anything else, helps to attenuate these defects in the sound then it is an improvement, for that system. All the cables that don't assist this process, by allowing the circuitry to function more 'correctly' and as intended, will be shown up as being "not as good".

 

The precise reason will usually revolve around aspects of the fact that all very high speed electrical behaviour generates noise which is much harder to isolate sensitive circuitry from - the faster the signals, the more effort required to mitigate interference.

[fas42]

20 minutes ago, greenleo said:

What makes Lush sounds better is unknown technically or I've missed the explanation.  Certainly, this magic is repeatable, hence this is engineering or science.  The engineer allows the circuit to function more correctly is to get rid of the bad things.  If it sounds worse, then probably the implementation was not good enough, the theory is not good enough, ...  By proposing a theory, testing it, modifying it if necessary is a way to proceed.

 

I've read between the lines of what's been said, and it's fairly clear to me that the circuitry receiving the USB data, and passing on the data therein to the DAC is typically sensitive to the actual nature of the electrical waveforms, aside from the data content - and that includes the power and ground links. Lush appears to slow down the data transitions, it modifies the waveforms to some degree - which experimentally has been shown to improve the SQ. If one wishes to rigorously investigate, then one could create a special test rig, with the ability to mimic USB data transmission; but with the ability to vary the precise nature of the waveforms on each line in a totally controllable way.

[fas42]

Thanks for filling in, Peter!

 

To say things another way, the same number of bytes are always transmitted, but each byte is represented by a data line switching on and off a number of times, and it's the speed with which the line switches each time that matters, how fast the voltage goes from one state to the other.

 

Make sense?

[fas42]

On 26/07/2017 at 7:55 AM, PeterSt said:

B'ASS : Bad Ass Bass.

At concerts I got intrigued by how electrical bass players are able to produce their fat bass sound, which is unfair because they have their own amplification and merely loudspeaker and how in the world to mimic this at home.

Well, add a few amperes to a DAC Gain stage and control that speaker (driver) !

 

I've looked with quite some bemusement at people who want "impressive bass", and add staggering numbers of monster drivers to get this - the result, to me, is a blubbering mess, which is miles from anything that one hears live ... capable amplifiers and stable, conventionally sized speakers will reproduce the visceral impact that the bass instruments or sounds in the track can deliver.

 

I have a ZZ Top CD from their heavy use of synthesizer years, which has very intense, subterranean bass effects in it - I have yet to hear another system do a decent job of getting this right, no matter how impressive their subwoofers appear to be.

 

[fas42]

5 hours ago, PeterSt said:

 

Maybe come look over here then ? 

 

For starters, subwoofers are out of order. They only distort (and not the least at it). So my speakers (of my own design - see sig) produce no audible distortion at 89dBSPL and 19Hz. 100Hz - 19Hz is +/- 0.5dB (and not any useless +/-3dB number - do you know how much audible even 1dB is in that regions ?).

I am pretty sure that no speaker exists that can do this, as no speaker is tuned like this. So we made this like it was a DAC (by THD numbers). And "no distortion" means that when you play this 89dBSPL at 19Hz, you hear totally nothing. And not because nothing moves so be careful that your hair remains in position.

 

 

Would be impressive to "hear" that lack of distortion at 19Hz - extremely hard to do at the lowest notes, because everything conspires against one, especially the sensitivity of the ear ...

 

I see that you're doing it by making the speakers extremely sensitive - good tactic! I use conventional speakers, but make the electronics driving them behave themselves; and don't worry about the lowest frequencies - I find the ear/brain fills in the gaps sufficiently in that region when the spectrum higher up is decent enough.

[fas42]

10 hours ago, Paul R said:

 

Hi Peter - 

 

Everything is distorted to one degree or another, in the audiophile world it is simply choosing what form of distortion pleases your the most. (Or irritates you the least. Same thing in a way...) 

 

Speaking only for me, the smooth frequency response that most sealed (i.e. acoustic suspension) speakers produce is more pleasing to me than say, the rigorous time alignment produced by coaxial drivers in speakers like the LS50. This is true even though both types of speakers are valid audiophile designs, they just appeal to different people. 

 

 

There is a specific quality level I always work towards which ticks all the boxes for me - and that is when it becomes impossible to discern any distortion from the speaker drivers - they become 'invisible' under every possible listening situation. And that includes having one's ear literally only inches from the moving diaphragm while the volume is at normal listening level - this is equivalent to standing right next to the body of a musical instrument being played.

 

There is still distortion there, but the mind can discard it - the musical message overrides everything that doesn't "fit in", at that quality level.

 

I'm curious whether Peter achieves, or understands this aspect of audio playback potential ...

[fas42]

1 hour ago, acg said:

What Peter does is off-the-charts...an entire ecosystem from playback software >> operating system >> audio computer >> cables >> dac >> amplifiers >> speakers.  That IS something truly extraordinary.  He is not a guy that concentrates on one or perhaps two parts of that chain...he sees all  parts as co-dependent and as a result is able to make some unique advances into sound reproduction.

 

Very, very few people understand the importance of this concept - it's not about adding "goodness" to the ensemble; it's about removing all the crucial bits of "badness" that may impact the overall result, in any and every area of the whole.

[fas42]

10 hours ago, PeterSt said:

That is a different story. Point is : you don't know what you are missing. OK, so all is fine. Haha.
 

 

I've never had a thing about the very low bass being important - I have yet to hear a system do, say, pipe organ as competently as I've managed myself - "gutless" is a word that I find easy to use with other attempts to play this sort of music - yet, this is without low bass in place, . The rest of the spectrum is far, far more important, IME.

[fas42]

1 hour ago, Paul R said:

 

I'm a bit curious if I this is actually a verifiable hypothesis. I expect if it is impossible to discern any distortion from the speakers, then the playback is so loud your ears and brain are - in a way - shutting down to protect themselves. This is a well known phenomena that has been experimentally verified. 

 

Tonight I am battling a bit of tinnitus and everything sounds off and different. Headphones are the only way I can enjoyably listen, but listening too loudly does more damage than anything else. As does age and far too much exposure to the the 58hz hum of Allison T56's. 

 

Yes, you're right that the ear/brain shuts down - it very specifically reduces the sensitivity of one's hearing, to handle the SPLs - in exactly the same way as listening to loud, natural sounds. When the speaker output is significantly distorted this happens to a greater degree - your mind says, "This sounds terrible - I don't want to hear it!!". But the 'magic' that happens when the distortion levels are low enough is that the AGC in your head merely adjusts to match how it would deal with those natural high level sounds - it then still sounds "real", even though it will be very loud.

 

Remember, this is only done as a checking exercise - no-one sane would stand right next to a set of bagpipes going full blast for hours on end! ...

[fas42]

Regarding, say Peter having the "best" products is not the point - rather, it's that he, and I, are sensitive to systems not working correctly - that is, not reproducing the sound quality inherent in the recording. So, you don't have to buy his products - sorry, Peter! - to get the "good stuff"; but it may be a quick shortcut.

 

My stance is that all reasonable systems can produce exceptional sound - the reason they normally don't is that not enough effort has gone into 'debugging' issues that are too audibly damaging - get everything on a level playing field, and even then some rigs will be better than others, for all the obvious reasons.

[fas42]

7 hours ago, Paul R said:

 

I am not sure any sane person would put their ear that close to a speaker playing at "full volume" even for a few seconds.  Not worth the risk of hearing damage IMNSHO.  

 

-Paul

 

 

You're still not getting the point that it is no louder, in number terms, than what every musician experiences when playing his instrument in the company of others - drop a hard object on a stone or concrete floor, and you'll experience a transient burst of in the region of 130dB sound level. Your ears don't explode as a result of this, the human system can handle bursts of extreme sound with no problems - it's only if one is dumb enough to hang around in such an environment for 10's of minutes or hours that things start going downhill.

 

One can do the maths on what the sound level really is with the ears this close - it goes up 6dB per halving of distance from the direct source; let's say the peak is 95dB 2 metres away; 1m, 101dB; 50cm 107dB; 25cm, 113dB; 12cm, 119dB; 6cm, 125dB. Check the ruler - 6cm is mighty, mighty close! And, we have still not hit 130dB - also, I've taken an extreme example; typically one is doing this with far more moderate volumes, say 85dB peak at 2m - which then translates to a 115dB peak with the ear a little over 2" away. Players in an orchestra experience this for hours on end, so ...

 

Of course, if the sound is significantly distorted then it will be quite traumatic to listen to, even for a very short bursts ... which, is the whole point of doing this exercise ...

[fas42]

1 minute ago, Doak said:

30 hours on the LUSH and first "serious" listen. 

Word that comes to mind is DENSE,, and I mean that in the best possible way. 

Which brings us back to LUSH. 

More later on after comparisons -- a few days at least. 

In no hurry to remove this cable from the system

 

I use the term "intense" - the impact of the music is such that it grabs you by the throat and doesn't let you go - in the nicest possible way ...

[fas42]

1 hour ago, Paul R said:

 

Like I said, I think we will have to disagree.  There is a *significant* difference between what "every musician" experiences when playing their instrument and what you are talking about.  I do not know any musicians who would do what you suggest. 

 

YMMV. 

 

 

Fair enough. I'm just talking about the sound levels experienced by musicians who are alongside other musicians playing their instruments, and someone, not necessarily a musician, listening that close to speaker drivers being of the same order. And for me, the impact matches, if the SQ is good enough ...

[fas42]

1 hour ago, PeterSt said:

 

Of course that does exist too. But I am pretty sure Doak is not talking about that. He is now using "technical" terms which are not related to perception.

I almost always talk in technical terms as well, as all is about elements which builds up fine sound. So I work on thos elements. I can't work on "intensity", so to speak. But I also can't work on foot tapping. This is, as it seems, where all comes together and starts to play music. It is the opposite of all technical elements working perfectly, and robots play (it is about the easiest to accomplish and the hardest to avoid ).

 

Except, I assert that "all technical elements working perfectly" and "all comes together and starts to play music" mean the same thing - they are just different ways of expressing the same status, of the playback system. If I don't get "toe tapping" then the sound is audibly flawed; some technical aspect is not good enough, and needs to be improved upon. If you mean a setup is good enough in the conventional measurements sense, but fails to deliver the music - then, yes, that happens all the time. But the only reason for that is that the wrong things are being measured - as they say, .

[fas42]

1 hour ago, Andyman said:

Errm, I think you've got that wrong. If you double per halving of distance, then as you approach the source, the sound level will approach infinity.

 

Rule of thumb for acoustics: double the distance from the source, reduce by 6dB - so, works in the other direction, also. Why the infinity thing doesn't happen is because this rule assumes a perfect point source for the sound - which is good enough for most situations of distance - and when you get extremely close to the driver or other source of sound it's no longer a point source - so, that rule can't be used any more.

[fas42]

2 hours ago, PeterSt said:

 

Hmm ... in my book that does not need to be true at all, unless you know the technical element which could be called some kind of glue, that makes the lot play music. So let's keep in mind, I can control most of the aspects we perceive from "sound through loudspeakers" as such, most of them by software alone. But there is no way that I have a dial for musicality.

 

 

The dial for musicality is to have no significant artifacts in the sound. By varying how the OS and computer processes the audio files, as you do, one will achieve different SQs, because the nature of the electrical interference alters - the ideal is to have none, but in our non-perfect world it's likely there will be some - so, play with every parameter until a reasonable optimum is achieved.

[fas42]

2 hours ago, PeterSt said:

One more for better explanation what I mean to say :

 

 

So an LP can't do that ? Music cassette can't ?

Of course they can, despite the noise, way fewer dynamics and actually lousy sound in comparison to what could be. And now watch out :

 

 

Referring to the "toe tapping" factor, - yes, it's not an if and only if situation: "audibly flawed" sound can be very catchy ...

[fas42]

Just now, PeterSt said:

 

Sure.

I see you take it lightly. 

 

Indeed I do - too many bigger problems in the world, to get hung up on things in audio having to be "my way!"

[fas42]

8 hours ago, Superdad said:

 

I think what Peter way implying is that you appeared to take too lightly the fact that he feels:

a) fine tuning of the s/w is very important to the SQ;

b) it has taken him--and the entire XXHighEnd user base--a long time and a lot of patient work to arrive at their near-ideal settings and achieve concensus about them.

 

Sorry if I misunderstood Peter, ... I come from an engineer's perspective, and to me each part of a whole system should do its job as perfectly as possible, to make sure that the combination of bits gets it right. Meaning, how a piece of software such as Windows 10 sends the music data out to the audio reproduction circuitry should have no influence on the SQ - but, the reality is that it does make a difference. I have done experiments, including using XXHighEnd, and found a particular package got the most out of the PC I was looking at - this happened to be a basic, old version of the Nero player; on my current laptop, it turned out to be Media Monkey - XXHighEnd may have improved things again, but I was happy enough with the MM results.

 

But this shouldn't be so - the hardware should do its job just as well irrespective of how the software drives it. I feel very strongly about that, and my aim would be to make the hardware solution as 'robust' as possible, so the SQ is always optimal. In that sense I take Peter's work "lightly"; but because the real world is non-ideal what Peter and his group are doing are of very high value to those are sensitive to the quality of the replay - it's simply that my approach would focus only on the hardware aspects.

[fas42]

5 hours ago, daverich4 said:

 

As hearing damage at 115dB starts at around 30 seconds I suspect you're wrong about orchestra members spending hours at a time in that environment. 

 

http://dangerousdecibels.org/education/information-center/decibel-exposure-time-guidelines/

 

The damage consideration always has to take the duration of the peak SPLs into account - if a continuous tone then things like the 30 secs become meaningful. But musical passages will only generate a millisec or less of sound at such a level - and drop back; the ears "get a rest". There are very extensive studies of sound levels experienced by members of an orchestra - it's easy to get 125dB peak, and if you're unfortunate enough to be directly in front of a brass instrument it can hit 130dB.

[fas42]

1 hour ago, Speed Racer said:

I would like Peter explain how this cable is affecting the sound output of a DAC considering the cable is carrying digital data. How, exactly, does the cable make the sound quality better? Since Peter designs cables, DACs, and speakers, this should be simple to do. 

 

A basic mistake is made over and over again in the thinking about how to improve SQ, by most people - the idea is that you add something that "makes it better". But that's not what it's about - what needs to happen is to to work out, by some means, where the sound is being "damaged" - and fix that!! What the Lush cable is doing, by the sound of it , is bypassing issues of how the analogue circuitry after the USB data side of things is being interfered with by the digital electrical shenanigans ...

[fas42]

1 hour ago, mansr said:

And how, pray tell, does a piece of wire accomplish this miracle?

 

It's already been discussed - the rise and fall times of the data lines signals have been increased - the switching is 'slower' than normal - call it bandwidth filtering, if you wish. Faster signals find it easier to interfere with other circuitry - a basic law of electronics, that engineers always have to grapple with.

[fas42]

20 minutes ago, mansr said:

Wouldn't that increase the susceptibility to jitter?

 

 

Possibly. It all depends on how the circuitry receiving the data handles the signal - if it's buffered intelligently, and the master clock is in the DAC then there should be no issues.