Misleading Measurements

[sandyk]

5 hours ago, The Computer Audiophile said:

If objectivists can agree that human adults can hear up to 20 kHz, they can surely agree on other limits when presented real data. 

 

 FWIW, Barry Diament's wife was able to hear 23kHz which is above CD's limitations .

[sandyk]

1 hour ago, pkane2001 said:

 

 

Deleted.

[sandyk]

4 hours ago, pkane2001 said:

All in one shot. If this doesn't introduce any measurable additional noise into my system compared to, say, a Lush^2 cable feeding the same DAC directly from a battery-powered PC, then I'm not worried that anything is introducing unwanted noise. 

 Why a  Lush^2 USB cable ?

 Are you admitting that some specification meeting USB cables may result in audible differences , even if not obviously measuring differently ? 😋 

[sandyk]

42 minutes ago, pkane2001 said:

 

Because I tested one.

Other than a Subjective check  , are you able to post some of your measurements, including it's measured impedance, which is possibly the most critical for a high quality USB cable , which could illustrate why so many members favour this cable over most generic USB cables ?

[sandyk]

29 minutes ago, John Dyson said:

Being able to use an objective measurement of some kind -- often the direct measurement being further processed so that the details are more clear seems to be more effective when available.  Properly presented details about the 'noise' can be helpful to pinpoint even unforseen problems.  

 During the development  of my DIY Class A Preamp I used a CRO at maximum sensitivity with an inline very low noise battery powered 10 x Preamp to highlight potential noise problems, which in this case appeared to be related mainly to nearby SMPS powered devices. Attending to these very low residual noise levels did result in an apparent Subjective improvement.

[sandyk]

1 hour ago, John Dyson said:

I don't disagree with your abilities/skill, but here is my position:

 

The problem is that just using 'scopes in the traditional ways are not very selective in providing noise information.   They are okay at general information, but it takes spectral and other presentations to study what is going on.  I am NOT claiming that *aided*  measurements are the only way to find problems, but nowdays we have so many easy-to-use information sorting aids, there is no reason not to use them.  Here is one of my long, blathering anecdotes (off topic, but an exemplar):

John

 You are overstating the problems when  using established design techniques such as Analogue Preamplifiers and Power Amplifiers based on the research of highly qualified designers such as Nelson Pass, Bob Cordell and Douglas Self .etc. 

 Neither are most E.Es across most areas of Electronics like you appear to be suggesting.

Most EEs these days appear to specialise in a selected area. Very few E.E's these days are even capable of marrying together both the Digital and Analogue areas of a DAC in order to  create an exceptional performing DAC other than one  having perhaps good measurements that sounds O.K.

 Very few complete consumer type products, other than from a small company are likely to be the result of just one person..

I doubt that Mansr could achieve that either, despite all the claims he  used to make, as well as the vast majority of the members of  that other forum that several members here keep quoting measurements from as being definitive..

 

I do however have a couple of modest  E.E. friends who do know their own limitations outside the areas they mainly specialise in,  and would .not be capable of  the  know how to make an amazingly good AM/MW/SW receiver without a great deal of research and assistance.

 

BTW, you are putting Ray Dolby on a pedestal that very few designers other can aspire to..

Alex

[sandyk]

50 minutes ago, Audiophile Neuroscience said:

 

Yellow and Red Dacs have a warmer tone, blue and white a cooler sound coloration. I think this is the main reason Toole advocated blind testing😄

 The problem here is that we are relying on Subjective reports by the suppliers and Reviewers, as well as Measurements which may or may not, be applicable , or even enough parameters measured, or for that matter, even  representative of a product due to the variance between production samples.. Perhaps even a "blue printed" or carefully selected sample supplied for reviews in some instances ? 

 I would hope that the vast majority of this forum's members are smart enough to do their comparisons "non sighted, with all components powered up as to give no hint as to which device is currently being used.

 This is certainly the way that sessions where David and myself  have been present are performed.

[sandyk]

1 hour ago, pkane2001 said:

 

Really? Mains interference is never measured? Maybe it's not something a THD or SINAD number by itself will reveal, but certainly distortion and noise floor plots will show it.

 

 

 These measurements are normally taken with the D.U.T. under isolated conditions, not as  part of a system

[sandyk]

1 hour ago, The Computer Audiophile said:

I know they fixed the zero crossing glitch just to appease people who looked at the measurements. 

 

I doubt that Mani needed measurements to tell him that something wasn't quite right.

[sandyk]

1 minute ago, The Computer Audiophile said:

I don’t doubt what Mani heard, but one guy on Earth doesn’t make it fact. 

  I doubt that there were even more than several hundred  (perhaps a thousand ?) guys on earth that had even heard that product at that point in time .

[sandyk]

10 hours ago, pkane2001 said:

 

 

Chris has been doing and saying many things about ASR and Amir specifically, but whenever I press for details, he just tells me to connect the dots myself. Not very convincing evidence there, either, I'm afraid.

 

A manufacturer certainly doesn't want bad news to be published about their product. Neither does an audiophile who invested hard-earned cash into the product. Nor the owner of a web forum which is sponsored by these same manufacturers and audiophiles. That's understandable. But that's why measurements are important. Everyone has a bias and everyone has a hidden agenda. Measurements do not.

 

I'll agree to drop it, but only if you do it, as well.

 

 

 

All this appears to show is that the iFi iDAC2 is in need of further improvements in it's own internal PSU area

You shouldn't need to go to these extremes with any external PSU, let alone the need to use expensive ultra  low noise  PSU costing as much, OR MORE , than the DAC itself !.

 However, attacking Uptone appears to have become  a hobby horse for ASR members while treading way more gently with their favoured manufacturers.

[sandyk]

2 minutes ago, pkane2001 said:

It shows that Uptone products were not doing what they were claimed to do. The reason to use the iFi DAC was specifically because it is sensitive to its USB input and any noise that it may carry.

 

 The many 100s of members who have purchased Uptone PSUs have found them to result in both a measurable and better sounding experience than virtually all off the shelf PSUs. It is up to the manufacturer of a product that needs an external Power Supply to ensure that they do not need very expensive external PSUs to make their product perform as well as their purchasers expect them to do.

 

Would I personally have used an SMPS plugpack to power this Ultracap PSU , NO , but purchasers were well aware from discussions in the Uptone Forum that it could also have been used in conjunction with a cheap linear PSU such as the cheap types on ebay that many members have used. It's not as if it was an advertised product  either, with sales mainly to A.S. members and by word of mouth from their happy customers.

[sandyk]

18 minutes ago, pkane2001 said:

Not helpful at all, Alex.

 Neither are your never ending references to Amir and ASR.

I note also, that the previous glitch measurements mentioned were supposed to be below the audibility threshold. Clearly they were not ,which brings again into question the Actual fully verified Audibility threshold with modern well designed equipment using very low noise PSUs , more recent techniques and components. 

[sandyk]

7 minutes ago, pkane2001 said:

 

But please, provide SOME objective evidence for anything you  claim. Have some respect for the rules of this forum, and share something more than a personal opinion.

 

Like you have done with actual confirmed , and generally accepted dB measurements of the levels of Audibility ,which iFi clearly believed that they had met with the crossover Glitch problem that turned out to be quite audible with a revealing system ?

[sandyk]

1 hour ago, pkane2001 said:

As I said, in most recordings soundstage is artificially generated, instruments and voices placed at desired positions using level, phase, and reverb. The soundstage you hear is defined by those parameters. Perfectly.

 

There are 100s of thousands of recordings with natural ambience and pin point location of instruments and voices, including many Classical recordings,  and quite a few famous recordings such as Jazz at the Pawnshop etc. and quite a few from Chesky, Soundkeeper and Blue Coast  Records.

[sandyk]

11 hours ago, pkane2001 said:

As discussed, it is not THD but interchannel differences that determine soundstage “quality”. These are caused by differing levels and amounts/types of distortion between channels.

Added HF detail may also result in a larger "billowly" soundstage. Many members are doing this by using daisy chained LT3045 ultra low noise voltage regulators which have a considerably lower output impedance at >100kHz.

 

This is a quotation from Bob Katz, well known recording & audio mastering engineer posted here

www.digido.com/audio-faq/j/jitter-better-sound.html

After an engineer learns to identify the sound of signal-correlated jitter, then you can move on to recognizing the more subtle forms of jitter and finally, can be more prepared to subjectively judge whether one source sounds better than another.

Here are some audible symptoms of jitter that allow us to determine that one source sounds "better" than another with a reasonable degree of scientific backing:

It is well known that jitter degrades stereo image, separation, depth, ambience, dynamic range.

Therefore, when during a listening comparison, comparing source A versus source B (and both have already been proved to be identical bitwise):

The source which exhibits greater stereo ambience and depth is the "better" one.

The source which exhibits more apparent dynamic range is the "better" one.

The source which is less edgy on the high end (most obvious sonic signature of signal correlated jitter) is the "better" one.

And a reply:
The better one, and it is better, is also easier to listen to. . . less fatiguing. I would also add to this that the low end just "feels" bigger and more solid. This is perhaps a psychoacoustic affect more than a measurable one. It may be that the combination of a less edgy high end and greater depth and width makes the bass seem better.

All of this makes sense if thought of in terms of timing (that is what we're talking about isn't it ;-]). With minimal jitter nothing is smeared, a note and all its harmonics line up, the sound is more liquid (a term probably from the "audiophile" crowd but one which accurately describes the sound none the less), and images within the soundstage are clearly defined. 

Now some extra points:
- listener fatigue is reduced or completely eliminated
- the sound can be turned up higher without any distortion being evident
- the sound can also be turned down lower & the full dynamics are still retained but at a lower volume

 

 And one from Barrows which both John Kenny and myself agreed with.

 

Quote

 Quote 

The thing is, I could imagine some people preferring the sound of higher jitter,
in some systems. In other words, a low jitter source, if one is used to hearing higher jitter levels,

may point out what I would call problems in a system. It appears to me, that some levels/spectrums of jitter

may have a euphonic result in some systems. To me, in a good system, the results of lowering jitter are increased detail retrieval

(as evidenced by image specificity, decays, more complex harmonic portrayals) accompanied by greater listening ease.

Higher levels of jitter result in a hazy sound, obscuring these same lower level details to some extent,

but sometimes higher jitter levels also result in the appearance of a larger, billowing, soundstage,

that can be somewhat impressive at first listening. I can certainly imagine a system, that is already on the bright/hard side,

where a listener might be very impressed by the sound of a higher jitter source and the little bit of haze it brings,

along with a big, billowy, soundstage.
 

 

[sandyk]

6 minutes ago, fas42 said:

 

Good post, Alex ... now, how many systems does one come across, in real life, that tick those 3 boxes ... ?

Note this bit in particular, as many seem to think that they can correct all this damage with further conversions to a much higher bit rate.

Quote

It is well known that jitter degrades stereo image, separation, depth, ambience, dynamic range.

Therefore, when during a listening comparison, comparing source A versus source B (and both have already been proved to be identical bitwise):

 

[sandyk]

26 minutes ago, Audiophile Neuroscience said:

Conceptually, and maybe ill conceived, based on previous quoted material and that quoted below, Intonation is - acoustic sum and difference tones from constructive and destructive interference patterns of sound waves in air - it happens in the real world as a natural inevitable phenomenon and would inevitably happen anew on playback. Its not a fault or distortion. Its not IMD. If correct, every speaker on the planet would faithfully create the effect just as real instruments do in a real orchestra. The speaker would be 'flawed' if it didn't produce the notes that then mingle in the air to create sum and difference tones. There's no way around it apart from turning off the sound or maybe doing some sort of fancy DSP algorithm which I am totally unaware about.

 

The 'double down' effect (my words) would occur because the mic captured the sum and difference tones of the orchestra (as well as the fundamental component notes) - then the speaker must play back everything including the captured sum and difference tones and the fundamental component tones. The latter must create a new set of sum and difference tones in your living room which add to the original captured sum and difference tones.

 

This process would be expected, normal and inevitable.

 

 

[sandyk]

15 minutes ago, fas42 said:

 

If there are intermodulation frequencies created then it's because the analogue circuitry is distorting, in the classic manner, from non-linearity in the signal path. Digital can be made arbitrarily precise, and so any distortion frequencies at levels that are meaningful can be avoided by simply using more bits to do the maths.

And of course your hearing, and Paul's hearing, is perfectly Linear without distortion ?  

[sandyk]

22 minutes ago, pkane2001 said:

What does this have to do with anything Frank said, which happens to be correct?

 

 Even if it is correct, it's a waste of time as non linear human hearing is involved, and in any event, no matter how high the initial bit rate it still ends up converted to 24 bit at best, which virtually ALL DACs are unable to fully resolve, with many DACs not fully resolving more than 21 bits properly..

 Neither have I seen any S/W signal generator that has distortion as low as you would require, not even the best of the Analogue generators such as the Wien Bridge Oscillators .

[sandyk]

12 minutes ago, pkane2001 said:

What exactly is a waste of time? Understanding what causes IMD and what doesn’t? That’s the only thing that Frank stated, nothing related to the number of bits a DAC  can resolve.

 I repeat

Neither have I seen any S/W signal generator that has distortion as low as you would require, not even the best of the Analogue generators such as the Wien Bridge Oscillators .

 Any way, what has any of this to do with the topic of the thread Misleading Measurements ?

 

[sandyk]

6 minutes ago, pkane2001 said:

I don’t require anything, Alex. Frank explained what generates IMD. That’s all. 

Everything that we do in the Analogue and Digital areas results in some degree of IMD.

 

[sandyk]

18 minutes ago, pkane2001 said:

And increasing bit size of the internal calculations in digital domain moves digital IMD to lower and lower levels. That’s what Frank said. Another way to reduce it is dithering.

And dumbing it down to playable bit rates again,  increases it again.

[sandyk]

46 minutes ago, pkane2001 said:

Now, compare that to IMD levels in an audiophile amplifier, say an SET.

An Audiophile amplifier is supposed to be more like a straight wire with gain.. 😉

[sandyk]

4 hours ago, pkane2001 said:

With very large amplitude sound waves (extremely loud) the air becomes nonlinear. Even the air temperature can change locally due to high compression at the peaks of the waveform. Localized compression and temperature change result in faster sound wave propagation at the peaks than at the troughs, where the air gets rarefied  and cooler. The result is a distorted sine wave that might look more like sawtooth than sine.

 

What has any of that got to do with normal listening to music in a typical entertainment or optimised Audio room ?

 As for temperature changes due to Output levels in any normal listening environment . . . . . . .