CPU Load and Sound Quality

[fas42]

36 minutes ago, STC said:

You may have just unraveled a mystery that has been bugging me for three years. How about the the small USB DACs. They come with single DAC. Audioquest Dragonfly?

 

There are plenty of DAC chips that have at least 2 channels - two complete, separate converter circuits in a single physical device.

 

2 hours ago, sandyk said:

 

In the meantime, quite a few members in another area of the forum are even reporting hearing differences between different types of RAM

 

 

Alex, if one goes about it carefully, it's quite easy to hear variations in SQ depending on CPU activity, hardware arrangement, and the settings - this cheapo Toshiba laptop I'm now using is fairly recent, but still the tonality varies depending upon what I do - shifting some of the circuitry outside of the plastic, like the DAC, makes some things worse, and some things better; it's all part of a flux, I'm afraid 😉.

[fas42]

4 minutes ago, PeterSt said:

 

There is no real science in this that I can discover after so many years by now (going back to 2006), but from empirical finding quite a lot can be controlled at will (like wanting a different bass in a certain direction).

 

 

The "science" is that the DAC and following analogue circuitry should operate in as benign an electrical environment as one can organise - many a slip twixt the theory, and practice of achieving such.

[fas42]

4 minutes ago, STC said:

How is that answer helpful regarding Dragonfly?

 

Say, Dragonfly Red uses

 

Quote

The ES9010K2M SABRE Premier DAC is a high-performance 32-bit, 2-channel audio D/A converter targeted for portable high-fidelity audio applications

 

[fas42]

32 minutes ago, STC said:

Even PCM2704 as in the picture is a stereo DAC. Maybe I didn’t phrase the question correctly. The data reaching the DAC chip would be carrying left and right. Is the a delay between this signal due to the fact that SPDIF carries the data in series. 
 

I ask this because theoretical value and actual value in my system differs by about 11us. 

 

The control circuitry which is part of the chip - which accepts, say, SPDIF as input - does all the necessary buffering, delaying and synchronising of the data, if it's serial - it ensures that the two channels of digital data are fed to the actual converters, to analogue, completely synchronously.

[fas42]

4 hours ago, pkane2001 said:

 

Once we confirm that CPU activity causes unusual noise at the output of a DAC, we can then talk about analysis, measurements, audibility, etc. Fair?

 

You see, Alex, that's where we have a problem here ... the technicals want the cart before the horse - unless something is measurable, it can't possibly exist 😜... the poor ol' universe, out there, struggled with self confidence for millennia, because humans hadn't worked how to 'measure' it - only now, slowly, is it starting to be feel OK about being so unusual, because people are pumping out more and more numbers about it - its sense of reality will finally fall into place when mankind gets all the i's dotted and t's crossed ... 🙂

 

Trouble is, this tidiness has failed to make it into, say, the medical world - wouldn't it be great if you were not feeling OK, and then the best experts of the day proclaimed that they can't find anything wrong ... you would be instantly cured, because "It was all in your head", 😉.

[fas42]

28 minutes ago, pkane2001 said:

 

 All else was talk about others and what they may or may not have heard and how many people here believe in it and the products that claim to reduce CPU load wouldn’t exist If it wasn’t true. This second hand evidence is not something that I can take seriously, especially since it contradicts my own experience.
 

Patiently awaiting your next car analogy to explain why this is all wrong-headed 

 

 

You want visual evidence? - our infamous Gearslutz collection of converter loop captures provides plenty of that, in that every DA/AD chain does it slightly differently - is it such a stretch that extraneous electrical activity, possibly impacting the chain, causes variations also? 😀

 

The bolded bit above is what it's all about - one has to have it happen "in front of one's ears" to start believing things; and it's so, so easy to make excuses when one doesn't want to believe ... "the room was all wrong!"; "it's a distortion enhancing the recording!", "I had too much to drink!" - that's a real goody, 😜, "the recordings were specially selected to make it sound different!", etc, etc, etc, ...

 

When one knows, in the sense of inner knowing, that something is a factor, then it becomes easy to set up an experiment which demonstrates that - to oneself. This has nothing to do with expectation bias - one can try one direction, and it makes no difference; another direction, and things get worse; then the next variation is tried, and a definite gain occurs - this is something like cooking: you play with added ingredients, to see if you can make the flavour better - or should we discard those notoriously unreliable taste buds? 🙂

 

Oh dear ... the car analogy lost out to an upstart - who would have thunk it?

[fas42]

7 minutes ago, pkane2001 said:

You forget that I’ve run these exact tests with ADC loop through multiple DACs and multiple ADCs in the process of developing and testing. If there ever was a dependency on CPU load, it was not at all obvious. I’ve also run these tests specifically to check for the effect of CPU load, as I was trying to understand all the sources of distortions introduced in such a loop. I didn’t find any effect of CPU load, I’m afraid. In fact, different capacitance of the interconnects had a greater effect than any changes to CPU load I’ve tried.

 

Wait! What? No car analogy? Now I’m disappointed

 

 

And the simple answer, in your case, may be that that the loops you were using were sufficiently well engineered for it not be be significantly affected by the types of electrical interference that can occur with changes in CPU usage. Does this translate to mean that all chains are that well engineered?

 

Personally, I have zero desire for all these factors to matter - they make life messy! I don't have a need, as some audiophiles do, to make my rig incredibly sensitive to tiny variations - this strays into hifi being a hobby where the vehicle is everything, and the road you drive on doesn't matter at all.

 

There, there ... you got your analogy, this time ... feeling better, now? 😉

[fas42]

11 minutes ago, pkane2001 said:

 

Lush produced more jitter and spurae over a generic USB cable. It didn’t improve things. This was with XDA-2. May have to do with ground loops or shield connections, since these are configurable on Lush^2.

 

Which gives a hint as to why the configuring of the Lush cable may have a positive benefit - something close to random noise being introduced, into analogue circuits that are less well isolated, could have a masking effect on unpleasant, low level distortion.

[fas42]

22 minutes ago, sandyk said:

 

 I am being consistent. I find that recordings of performances captured in a room  from a pair of speakers, and then posted on YouTube are damn near useless for most purposes !!!

 

 

They are very useful, because they make it trivially obvious when the playback has significant issues - you see, you're not trying to determine how good it is; rather, you're wearing the cap of a troubleshooter - there are very few videos of good sounding systems on YT ... because, well, they are not good sounding in the flesh, 😉.

[fas42]

As an example of what can be conveyed about replay in a YT, a solid effort that came up some time ago,

 

 

 

[fas42]

13 minutes ago, sandyk said:

 They have an extended frequency response to 40kHZ (no details provided), and have a flatter frequency response  than most headphones in the critical VF area. 

 

Massive spike in the response around 10kHz - no wonder you like what John Dyson is doing to tame the HF stuff ... 😀

[fas42]

8 minutes ago, sandyk said:

Bullshit !

 Have you checked out the Frequency response of most pairs of headphones ?  Many die in the arse just around the 9-10kHz area .

Do you think that nobody else except  yourself is capable of taking into account things like this , which are also later compared by me via the main system where there are no such anomalies, and the DAC is of an even higher quality ? This is trivially easy for me to do because all of John's samples are saved to USB memory stick initially..

 What's your excuse where you prefer the accentuated HF response of the non decoded Dolby A original recordings ?

 

Take it easy, Alex, I was taking the pi.. 😄

 

What's the magic problem with headphones, in the 10k area, anyway?

 

You know I'm taking this Dolby A thing with a huge grain of salt - there are plenty of recordings with balances which provoke the replay chain into sounding unpleasant - this current laptop of mine, with 'micro' speakers, is no different from everything else out there. Which is why I'm using my CD rip of Ring, Ring to 'troubleshoot' - in non-optimum status, very screechy; and this steadily disappears as I get closer to better settings for everything.

[fas42]

20 minutes ago, STC said:

Don’t be silly. You have no idea how this was recorded or even if this actually the sound of system in the room. If they were to use Videomic or M/S mic then you are essentially listening to the original recording with the speaker and a little of the room frontal ambiance. Stereo reproduction via two speakers could not capture the left and right exactly as how we would hear them because each speakers would be producing the same sound ( with phase and level difference). Essentially they will sound more monophonic. 
 

 

Note, I posted this because Alex stated that one could not get any idea of a what a rig sounded like from a YT clip ...

 

So, the rule is, all sound clips on YT have been faked - is this correct?

 

Left, right, stereo, matters not one iota - what is interesting is whether the tonality matches up with what one perceives as decent sound.

 

20 minutes ago, STC said:

 

Don’t post others YouTube. Record one yourself and listen to the difference. The best to do proper judgment of sound of system replace is to use vocal. 

 

You know I've done that already. And, there is a lengthy vocal segment at the end of the above clip.

[fas42]

11 hours ago, STC said:

Maybe I feel the same. I always thought I could distinguish them. Unfortunately, I don’t get a chance to prove. I have always welcomed anyone to bring their equipment and test them but after the initial euphoria no one is willing to do.

 

Some years ago I did a serious exercise, trying to see whether MP3 encoding could fool one - using LAME, I played with the settings every which way, trying to squeeze the last ounce of accuracy out of the encoding - this was listened to on the Sharp boombox, with the speakers I currently use with the NAD electronics, and using the centre box of electronics of the purchased trio, 🙂.

 

Couldn't get there ... there were always obvious differences, and each different set of encoding settings created a new version to the ear ... quite listenable to, no major flaws - but never subjectively the same ...

[fas42]

12 hours ago, STC said:

In live performance you do not hear pristine direct sound but more of the ambiance which smoothen the peaks and dips. 

 

 

 

Here's someone who obviously hasn't stood on the footpath, right next to a marching band in full cry going past ... 😜.

 

Yep, it must have been the nearby clouds that provided the ambiance which smooths the peaks and dips, 😉.

[fas42]

4 hours ago, pkane2001 said:

 

Simply because. It's a flaw that would make the device inconvenient to consume and render it unusable and unpredictable for a lengthy period of time. If it really exists in a device, it's a broken design. Any engineer aware of such a lengthy break-in period would design around it to ensure the stability of the circuit during the break-in. Amazingly, nobody does this. Not in the pro audio field, not in the sensitive electronic equipment that measures individual photons, not in LIGO in the extremely sensitive gravitational wave detectors. So why is it that audiophiles are the only group in the world that suffers from cable and component break-in? 

 

In the world of normal consumer listening, everything's good enough. In the world of people who start to realise that fiddling with parts of a system alters the sound, everything's broken ...

 

In LIGO, etc, an incredible amount of effort would go into stabilising every tiny, tiny aspect - because everyone knows, up front, that if they don't do this then the value of the experiment is almost non-existent. In audio, the people making the stuff usually don't care a stuff about the "fine details" - so the 'fussy people' then have to play catch up, after the fact.

 

I could mention an experiment in Europe where people didn't get the soldering 100% perfect, and there was a big BOOOM!!! - but I won't ... 😝

[fas42]

Straddling the fence as I do, both sides have part of the story right, and both have parts of it wrong - in the latter sense, one lot are obsessed with using magic bits in their rig, and that only ultra expensive gear can make it happen; and the other are being ritualistic, and fanatical about the concept that factors which are easy to measure are all that can ever matter.

 

It was obvious to me over 3 decades ago that these two lots of wrong thinking would make it so, so hard for good sound to be more common - from where I stand, there has been close to zero progress since ... now, where is the rolling one's eyes, squared, emoji?

 

[fas42]

The trouble is that people who care enough to participate in an audio forum, are often those who are disturbed by others not confirming their world view on "how to go about it" ... 😉.

[fas42]

2 hours ago, Blackmorec said:

 As my system has gradually improved I’ve noticed that it has become increasingly sensitive to changes,  some of which i find difficult to explain. Take running-in as an example. These days, whenever i place anything new in the signal path it usually goes through a period of sounding worse before it sounds better.  Many audiophiles have noticed the same phenomenon. I can’t explain scientifically why it happens, and many people who claim to be scientists or engineers deny that the phenomenon exists and claim that it has to do with psychological effects.  As a scientist i try to look at things logically. If indeed the phenomenon is psychological then it should happen with any change i make to the system that requires me to adapt to the system’s new sound. So is that the case?  No, far from it. For anything that doesn’t involve the signal, the change is immediate and stable. For used gear that’s new to my system but not new to carrying a signal, the change is similarly immediate. Then there’s the direction of the change. Everything new that i install sounds initially worse then better. If its my hearing acclimatising, then its always acclimatising to worse sound. That means that every upgrade I’ve ever made sounds worse than the existing system! So how come my system gets consistently better over time? So logically acclimatisation actually makes no sense.

 

Yes to increasingly sensitive to changes ... but I don't have the running-in syndrome - my very strong suspicion is that you're dealing with static behaviours to at least some degree ... I tend to automatically take those sort of things into account, plus I rarely use something "brand new".

 

There are a whole variety of parasitic behaviours, related to materials used, and construction, which one way or another generate electrical noise - juuuust enough to cause audible variations for those sensitive to such things. These all have to be got under control if one wants stable SQ - to me, any rig is always a work in progress, because there are so many aspects that can be impactful.

 

Are the factors all logical? Yep ... the bastard is human hearing, which can be sensitised to some quality in what it hears, so easily - but there is a point which I call competent playback, which does enough to keep me happy - one can do better, but how much agony do you want to go through ... ? 😛

[fas42]

18 minutes ago, Blackmorec said:

If what you are saying is that some people’s hearing is sensitive to the changes that occur, then I would agree. However I don’t believe the variable is their hearing per-se, rather its their system’s ability to reveal the changes that varies between different people. When I started on this audio road, there was no such thing as specialised, just-for-audio cables or furniture. Speakers were wired up with lamp flex, the audio version being marked for phase and TTs and amps were placed on whatever convenient furniture was at hand. Sound quality was good and there was no such thing as burn-in. But as we discovered more about how to make our systems better.....new configurations, new components, new materials, improved configurations, improved digital resolution, less noise, better vibration control, better mains etc the improved sound quality brought with it a new phenomenon....burn-in. But as is obvious from the above, burn in is only an issue with sufficiently evolved systems, that have the ability to react to and reveal small changes. 

 

I've noted that people who have not the slightest interest in the hifi malarkey game - ie, women, 😉 - have no problem sensing when the sound is in a good space - and lose interest, very quickly, when it goes off. The more of a peak the rig is capable of reaching, the more easily it can plummet down the sides into an unpleasant loss of "specialness" - the ongoing drama for the DIY enthusiast, 🙄.

 

Making a system better I see as a process of identifying the weaknesses, and resolving them, one by one. Burn-in is just one of the axes that weakness can operate on - and I don't want it to be part of the picture. My goal is for a system to be "as good as it gets" in something like 10 minutes from something changing - including switching on from cold ... I'm a long way from achieving such, but see it as a proper endpoint, 🙂.

[fas42]

2 minutes ago, Blackmorec said:

If its so well understood, how come we are still able to make significant improvements every year?   The wheel is well understood....Audio and its reproduction is altogether too complex, involves far too many variables and constantly raises questions. There are still problems to solve, it still doesn’t sound like the live event, we can still improve the sound from recordings we made 40 years ago and reveal new levels of previously unheard information. In digital we find new and better ways to remove noise, optimise processing, convert digital to analog, build better CPUs, memory storage and disc drives, improved materials. 

 

I don't see things this way - for me it's a simple lack of understanding that any audio system needs a high level of integrity in every aspect of its structure, to realise the full potential. Simple systems, easier to debug; complexity invites more degrading weaknesses, and one spends far too much effort trying to compensate for the failings of the implementation.

 

2 minutes ago, Blackmorec said:

So in summary, we don’t know where next year’s, the next decade’s or the next centuries’ improvements lie. We don’t know what new materials will be developed, what new measurements will become available, what new physical laws will be discovered,  what new phenomena we’ll come to understand, but what we do know is that our understanding will continue to evolve and sound quality will continue to improve. 

But if we ignore problems or claim they don’t exist, then the driving force of progress stops. The more we recognise problems, the faster we solve them and improve the state of the art. 

 

I see the biggest problem is that far too few people realise how impressive playback of very ordinary recordings can be - if you don't know what you're missing, why would you bother trying to get it happening? 😉

[fas42]

11 hours ago, Racerxnet said:

Frank, can you tell me exactly what you are changing on the laptop to get the magic. IE, are you moving cables, changing the speakers, sitting it on toilet paper. What exactly are you doing. Modded bios? Please answer with some specific details. 

 

Mak

 

Well, I'm telling some of the story in the Step By Step Surgery thread - aim is to not touch anything of the hardware, apart from disconnecting everything that doesn't matter. This is just another version of what people like PeterSt do - change how the software drives the m/c ... I still haven't finished, being sidetracked by forum activity, and that the power adapter failed - did a rough fix to get it going again.

 

No point in giving a precise list yet, because I might go backwards on one of the steps - when I've gone as far as I feel is reasonable, I'll try and give a full rundown ... OK?

[fas42]

6 hours ago, Blackmorec said:

Hi Racerxnet, 

This isn’t about golden ears....educated ears maybe be certainly not golden. What it is about is taking great care in setting up all elements of a system properly, starting with the room and its acoustics, the power supply to the room, the speaker positioning within the room, vibration control of all system elements,  any network supply and EMI and last but not least, excellent synergistic components and all its interconnecting cables. 

When all that is correct, you’ll have an absolutely stunning system that is a joy to listen to and most recordings will sound very good to exceptional, although there’re always a few baduns. 

And when you then listen to other systems that have not been optimised, what you’ll often hear in comparison is a slightly homogenised sound, lacking in detail, with tonal anomalies, lack of spacial focus and definition, that has poor timing, sounds rather boring and is generally not a lot of fun to listen to, as though you are listening to a rather poorly executed recording. You’ll also likely hear both loudspeakers as easily identified sources that is a sure giveaway that something is lacking with the set-up

 

 

 

 

This could almost be me talking - except I don't worry about the room and its acoustics, or speaker positioning within the room.

 

With regard to listening to non-optimised systems I could use the same language, except that I then deliberately use a 'difficult' recording, which provokes that setup to produce every obvious distortion - I now have a marker which guides me to improve things.

 

A simple example with the Ring, Ring track I'm using with the current laptop: the chorus line in that song, the actual phrase "Ring, Ring" as sung by the girls has significant distortion, a screechiness, when non-optimised - this steadily is got under control as changes are made in the right direction, and translates to the direct and added reverb sound, makes complete sense to my hearing.

[fas42]

3 minutes ago, Racerxnet said:

You optimized what exactly?

 

MAK

 

CPU Load ... 🤩

 

Nothing to do with bitperfect; all about altering the electrical activity within the plastic, while the track is playing. Since all the circuitry is next to each other, with only the most basic power supplies in the thing, it should be pretty obvious that one can alter the interference activity going on ...

 

BTW, nice that you are playing with Nudell's offspring - one of the best LP experiences I had decades ago was in a dealer's home, with such, and Goldmund Reference, Audio Research biggies - the classic combo.

[fas42]

7 minutes ago, Racerxnet said:

 

There is not much you are going to do within the confines of a laptop. Putting it on the battery?? Then what! Dead end. If you are telling me otherwise, show some pictures..

 

MAK

 

P.S., It is Nudell, and McGowan's work from the genesis line.

 

I note that you responded on the Step thread; so you know what I'm exploring - I just take it as far as I can, and see what that gives me ... patience, grasshopper ☺️.

 

Have you looked at https://www.genesisloudspeakers.com/products/ ?