USB audio cracked... finally!

[PeterSt]

51 minutes ago, jabbr said:

@PeterSt -- the above papers detail what I was saying regarding capacitive termination -- all of this would need to take into account the cable impedance (and transmission line model) itself.

 

@jabbr,

 

Prize of the day (OK, yesterday) goes to you. Somehow you must be in the wrong job. 

For others, Jonathan sent me a PM suggesting this, which I rejected. But why ?

 

I can't judge that type of solution "brain wise" because too much of the same is involved elsewhere (maybe it is better to think "everywhere"). It's a resistor here, a capacitor there and a ferrite bead such. But not in the cable (promise).

So doing it like that would be quite out of control. But also and honestly, I just would not attempt it like that because too much of real filtering would be involved and it would destroy the signal too much (maybe).

 

[PeterSt]

All I can add to this discussion is that it is not even easy to find consensus about these differential matters. Notice that this is implied by the very low speed we talk about. In my experience, when the chips are down, the EE may tell you "oh don't bother at these lousy speeds - it will work anyway".

That this is exactly the headroom we now can play with, is ... good for audio.

 

On the other hand ...

 

Get yourself a twisted pair cable which is unevenly twisted or has too few twists per distance unit, and nothing works.

 

[PeterSt]

21 hours ago, The Computer Audiophile said:

8 pages and this thread should already be well off the rails. 

 

Thanks to everyone for being respectful and helping (or not participating at all) to make the discussion enjoyable. 

 

[PeterSt]

7 hours ago, jabbr said:

 In any case the DIYINHK version is $40 but doesn't have the isolation.

 

Then you are looking at the wrong DIYINHK version. 

The one costing $109 sure has isolation. And much more than the Schiit offers. Like 1024 DSD support. Hahaha. Oops.

 

But it is still USB. Who wants that. 

[PeterSt]

1 minute ago, marce said:

All this is pretty simple stuff for something based on a USB interface (or even SPDIF) compared to some of the digital interfaces around.

 

That is what you would like (to think). Any complex interface (you implicitly refer to) is error corrected. Also, such an interface does not suffer from jitter really, assumed it carries low enough jitter not to produce errors. Think telephone conversations (ok, they s*ck to begin with).

Audio (as in audiophile terminology) is a totally different world for uncountable reasons. The most complex of all, perhaps.

[PeterSt]

16 minutes ago, marce said:

Its a basic USB interface, use some signal integrity software, play about... Me thinks some audiophiles see to many gremlins with any digital interface

 

OK, next attempt. 

 

So we have the subject of this topic - the cable. If you are certain that USB cables don't make a difference for audio, then I think it is better to stop reading.

 

Then we have my personal idea that at this moment (and for me / Phasure) the most in audio can be advanced by means of USB improvements. You may call it tweaks, but those sound only different. So improvements is key.

Of course it is (or should be) moot to you that this is my thinking. Fact is that since I think like this, the one after the other real improvement pops up (ask the customers).

 

Data integrity ? no. For many maybe yes, but not for me as the USB errors are shown in real time. They should not be there and they are not there. And *if* they are there you will hear clicks or pops or scratches, up to silence when things really fail. Btw, more USB cables fail on this than not.

Sadly, we can also say : Data integrity ? Yes. But this is more of the voodoo kind and the hope that less jitter in USB data sounds better or technical is better for the further work out of the D/A process. You may have seen me write about that I am not certain about this. Still the Phasure NOS1a/G3 too has a "femto clock" for its USB interface. Does it help ? yes, together with the remaining upgraded parts it does. On its own ? I don't think so, or it is inaudible.

 

Higher bandwidth USB ? well, this is a very indirect one, with the notice that this is about the upsampling possibility the DAC can digest (mind you, which thus the computer can spit out). Not so important ? ... only know that the filtering (which is that upsampling) "makes" the sound.

 

I'll stop for now but notice that this is all interface related only and that our interface is USB.

[PeterSt]

2 minutes ago, manisandher said:

Probably didn't get the graphs up quickly enough...

 

Nope. People might refresh the page to see them

[PeterSt]

2 minutes ago, mmerrill99 said:

The accountable reasons are that there is a digital to analog interface - staying in the digital domain is fine for all the interfaces - transitioning to the analog domain is where the issues arise & our pattern matching processing is picking up on the issues

 

Not that I disagree - not at all.

But this topic is about a worse subject : that the digital domain only already matters (and a LOT) and that again only within the USB transfer (which is digital from where it leaves the PC to its I2S output).

Btw in front of that we have again a pile of digital-only, which is merely what Jud referred to. And nothing sounds the same wherever you can change elements or settings like buffer sizes etc.

 

[PeterSt]

12 minutes ago, marce said:

if USB audio is that hard why do I see NO evidence of proper signal integrity engineering...

 

Because audio in this realm is not-error-corrected digital data.

 

That it is subject to jitter - which is audible - in the D/A process is something else.

 

You have some nice headphones or more expensive speakers, right ? Just asking.

Btw, a fine turn table also suffices. 

[PeterSt]

23 minutes ago, marce said:

Some of this is tickling me, I really do have some views on why some like to promote USB audio as terrible and have solutions for this...

 

Of course. But we all need to make some money ?!

 

All these things sound different, up to wildly different. But being an audiophile is not your hobby, so you think it sound can not change. If audio/music *is* your hobby - and especially when it is digital audio, you automatically will seek for solutions to what your ears don't like much.

 

As for the two eye diagrams you saw, yes, you are right.

The equipment needed for this is quite unaffordable for us poor guys. Maybe you disagree ?

[PeterSt]

15 minutes ago, marce said:

Again I wonder how much noise is affecting things... and with reference to mmerill99's comment, noise has increased greatly over the last decade, could this combined with cost based design (especially layout, its the most cost conscious part of any project) be one of the factors..

 

Marce, maybe it is the other way around;

Noise in good audio systems has DEcreased largely (think system noise of 150dB down and better) which may make the noise you refer to more profound. That is, if you want to think in that direction.

I don't think it is about that anyway. I do have the idea though that you might think that mentioned 150dB is crazy and not necessary at all.

 

On another note, the equipment you may have in mind for (FFT) measurement will be 12bit or worse (but fast like 100gbps+) which is useless to audio because your noise line will be at -100dB only.

I think we live in different worlds, no matter your hobby is audio too.

 

Regards,

Peter

 

[PeterSt]

3 minutes ago, marce said:

Sorry Peter I should have been more specific, I am referring to environmental noise EMI.

 

Marce, you told about that earlier on. But the point is that would this be a problem then it should hammer on the electrical voltages and audio signal. It may do that to some degree, but personally I don't see that as *the* problem. And what I tried to tell you with my last post : it seems that you are used to high speed / high frequency (GHz+++) stuff, thus high sampling analyzers BUT which will be 12 bits only (if not 8). With some optimization maybe 2 bits more.

And because good audio signal requires very low noise these days (I mentioned 150dB down) in your context (your work) you wouldn't be able to measure the signals (they will be deep in the noise of the analyzer).

Or another context : the power supply noise we use in our D/A converters is 160dB down (think +/-5V rails).

 

Moral (sort of) : I am sure we like to learn as much as we can from your high speed design work, but maybe first you need to know "our" audio environment better. Maybe. 

 

 

 

[PeterSt]

13 minutes ago, jabbr said:

Bypass caps like: https://www.johansontechnology.com/S42E and or networks of bypass caps properly placed on the ground plane should help to ameliorate such issues. (the digital gates can properly operate at higher rates)

 

Now you are too funny.

So what I design in, you design out ?

If you order a Lush I will send you a Clairixa. Saves you the trouble !

 

haha

[PeterSt]

1 hour ago, marce said:

Fast rise times are not always the best thing as they put more energy (especially high frequency) into the line. What is required is the correct rise time for the circuitry being driven, with minimal ringing... 

 

[PeterSt]

10 minutes ago, Jud said:

Does the low pass filter have its own ringing, or is that true only of digital filters?

 

Analog just the same.

 

Weren't you the one referring to loudspeakers ringing just the same ? I could be wrong. 

Anyway it would be because of the filters (IIR).

[PeterSt]

29 minutes ago, pkane2001 said:

So, as I was saying earlier: a low-pass filter. Is there a reason not to filter out most of the higher harmonics and get closer to a pure sinewave shape for each pulse? At least this will reduce any ringing around the edges of the 'square' wave possibly reducing jitter in the USB data stream and reducing higher frequency noise.

 

Not bad thinking. But a filter set up like what's implied (resistors etc., opamps - that is, if you refer to that (I did not really look, sorry)), then I would not do that because of the variance in resistance (or capacity) because of heatup of the "filter" itself, which would be dependent on frequency etc. etc. etc. (even a kind of oscillation could be your share).

The result is that now the slope to trigger on is not consistent and that implies jitter and ...

... and maybe that is not important but ...

 

[PeterSt]

OK Jud.

[PeterSt]

12 hours ago, pkane2001 said:

 

 

It's analog signal carrying digital carrier for another analog signal

 

 

I'd say it is an analog signal carrying a digital carrier for a protocol which represents a digital signal which maybe represents an analog signal.

 

The latter is, for example, not present when a digital synthesizer is the source and which never saw an analogue mixer etc.

 

 

[PeterSt]

12 hours ago, mmerrill99 said:

- when risetime is fast it may be more difficult to ensure consistency in the timing - it's this inconsistent timing that leads to jitter & this is audible

 

Why would this be audible ?

 

I'm not saying it isn't. But I also don't claim it is.

It is a complex matter and it should start with defining the domain (jitter-where and such).

[PeterSt]

On 2017-7-18 at 6:36 PM, jabbr said:

Digital switching noise and precise sequence of digital transitions are a perfect example.

 

I propose that any waveform which is used to trigger on its slope(s) does not result in "switching noise" as such. And otherwise it should be defined what is the "switching" phenomenon.

 

Switching noise is a quite formal term for a digital switch and one can spend a year in taming such devices, were it about utilizing them with the least amount of "noise". I am talking about high(er) frequency switches, like 45MHz. 

Of course this is a very low frequency compared to what exists in other worlds, but the noise coming from those switches is baaaad (for audio).

Btw, they carry jitter - but now I am trying to be funny.

 

[PeterSt]

22 minutes ago, mmerrill99 said:

Well I think one approach is to determine, by experiment, what factors do audibly affect the sound & then to try to work out the mechanism that results in it being audible.

I'm not sure that approaching it the other way "how is it audible"is the most effective way?

In other words, an empirical rather than theoretical approach.

 

Well, the emperical approach seems to be the only feasible approach. And in the end this *is* what we are doing. But it takes ages (I am talking in terms of years, really - and those are only the years behind us).

 

I must note that such a path is the most dangerous because it depends on my ears (and the ears of only a handful of others in the same "field"). So what happens is, that I judge something as sounding better (which already takes me a week to be sure) to next reason out what actually happened at the technical level. And well, as we can nicely see by this topic, is how a world of "power" is not really capable to draw firm conclusions. What also has been shown is how member marce, who for sure is right in the EE field and possibly better than all of os (for sure more experienced than myself), also would not be able to get there on his own, because he already does not recognize the problem (or maybe by now he does). So it takes rarities as John Swenson and possibly me, to be motivated to work out "something" while we don't know what the something is. We do recognize a problem "somewhere" though.

 

I still recall from the day like yesterday one of the first posts from Gordon Rankin in here, which went something like this :

 

I don't know how it works; Isochronous USB should be stopping all jitter, but whatever I do the jitter is passed on.

 

Weeks later it appeared that the (then all new) USB DAC's showed the worse measurements of anything at the time.

Today we are close to 10 years further and I could reason out how a USB cable could be made, but only based on another USB cable which was made to spec as good as possible and still only after 10 or so attempts to find the direction of "tweak". When that was found it worked in one go.

But 10 years ??

And by my ears ? ... I'd be careful if I were you. 

 

 

[PeterSt]

21 minutes ago, Dev said:

but I just received a mail from UPS that its delayed due to "Severe weather conditions"

 

WTF ?

 

So yes, all shipments to the US look exactly like this.

All I know is that this plane should go from Koeln (Germany) directly to the US, but not all with the same plane I think. Anyway this is all about CA and one about FL.

 

Severe weather ? So this plane leaves, and is in 30 minutes in Paris or so. It should be the direction, but fell out of the sky underway.

Over here in Holland there was a Thunderstorm at 4:45 am (lasted an our or so). So who knows ... it nicely fits.

 

I wouldn't be surprised if you get your cables tomorrow (Friday) after all, as the delay seems to be 3 hours only. But then of course it is easy to let logistics fail.

 

Sorry ...

Peter

 

 

[PeterSt]

16 minutes ago, BigAlMc said:

Initial impressions. It's spongy!

 

Maybe it's air only in there and we forgot something. 

 

haha

[PeterSt]

17 minutes ago, semente said:

Why hasn't anyone come up with a more effective alternative for connecting a Computer to a DAC?

 

A small bit of history, in case you don't know :

 

1.

I was working with a smart guy from Germany on world's first Isochronous USB DAC. Well, I helped him a bit with thinking about it and I helped with listening tests. The DAC was named Crazy-T and it took so long to get it working in good fashion that by the time it hit the market, it was caught by time itself and the 16/48 it could do max, was not sufficient.

 

2.

Someone named Gordon Rankin did a slightly better job, and USB started to do 24/96. Everybody was happy and I said that USB carried too much noise (this is again close to 10 years ago IIRC) and I said so on CA and nobody agreed.

 

3.

Of course I said so because I was ignorant and used a completely different interface for the first incarnation of the Phasure NOS1 DAC. It was a PCIe interface (don't tell). People maybe recall this photo (has been on CA in a larger series of other photos about the interface) :

 

It was the first interface doing 24/384.

 

4.

That Interface (some still use this first NOS1 version) gave us nightmares because Windows 7 of te time was not up to the task and it was always praying whether the system would boot. Besides that, it wasn't particularly plug and play (it required a PCIe card in the PC) and USB was the thing. So already within 6 months the NOS1a could be supplied with USB interface. I recall that it even sounded better, but not after a few months of fighting to get it right.

Because all is about numbers, it was the first (USB) interface capable of 24/768.

 

5.

When a first NOS1-USB went to the USA, it appeared that very strange DC Offset problems emerged, and from there on I knew that even the mains interferes with our precious audio interface.

 

I will stop here because it will be OT. But message is : I tried.

More in 7 months ... 

 

 

[PeterSt]

34 minutes ago, jabbr said:

How often is this done in audio equipment? 

 

 

 

At least one time.