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Blue or red pill?


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34 minutes ago, semente said:

 

I guess I should have written binaural playback.

 

I don't agree that DSP processing (ambio, etc.) is able to realistically reproduce the original soundfield. This is why I wrote that because both DSP and stereo have different qualities and defects opting for one or the other is a matter of taste.

 

Yes, the original soundfield can never be realised. Or it can only be done by using completely OTT processing. Which is why the smart way is to switch on the most powerful sound processing engine of all, the brain. All the standard DSP processes don't address the underlying problem, which is that playback quality is too degraded by the time it emerges from the speakers - so, it always sounds 'fake'.

 

It takes quite a bit of effort to eliminate all the disturbing anomalies that inhibit the mind's 'DSP' capability. But it's worth going there, because it's so extremely capable - the ear/brain handles everything thrown at it, and keeps the illusion rock solid under all circumstances.

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1 hour ago, mansr said:

Sorry, not hearing it. Left/right positioning is great, but I'm not getting a sense of depth or height. Maybe if I had a pre-conceived idea of where things should be, I could will myself into hearing them that way.

 

Well, the website that George pointed to is a disaster - for me, at least ... trying to hear some of the samples only achieved something like an old time short wave radio quality - so I tried YouTube :D. Came up with this clip, which may or may not be done by the man,

 

 

In any case, the quality shows up all the good things - would do very nicely on a full blown rig ...

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45 minutes ago, mansr said:

I don't know which website you're referring to, but I bought the CD quality download from Qobuz. It's a nice recording. I'm just not hearing any depth or height.

 

https://www.qobuz.com/gb-en/album/united-states-air-force-band-evolution-gustav-holst-peter-mennin-howard-hanson/0019688116125 - I tried listening to the samples available on that site to get an idea, but wasn't successful.

 

I'm sure there are many people whose brains are hardwired so that they can never perceive the illusion I speak of - but I can certainly pick up the clues from that YouTube clip; on very good playback there would be an enormous sense of depth and space, for people who hear in a similar way to how I do.

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1 hour ago, sandyk said:

 There is a small amount of depth information from this example in a couple of places, but your brain would need to work very hard filling in the blanks to get much more from it.

 Very few YouTube clips have much depth anyway, let alone height,  due to the max. 187Kilobits .aac used.

 

It's always about the playback quality - I've 'learned' to pick the clues from poorer quality renditions of a piece, to be able to extrapolate how it would sound in prime playback. Sometimes, a YouTube clip will come across better than I've heard the CD on ambitious systems, because key parts of the content are not blurred as much as they are on the "big rig".

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2 minutes ago, sandyk said:

 

 A YouTube clip should NEVER come across better than a well recorded CD on a decent system !

 

Amusingly, the audio friend I have down the road went to some effort to optimise the direct replay of YouTube clips over his "big rig" - I was impressed! Remarkable was the impact of some pieces - way better than a lot of audiophile setups I've heard over the years ... :P

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1 hour ago, STC said:

 

???. Now FAS24 is going to get upset. He is using some Philips HT box setup. 

 

>:(:mad:>:( ... ^_^

 

Was using Philips HT box ... it died, and then I tried other things. Latest toys are NAD CDP and amp, Sharp speakers - not yet competent, but usable.

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A favourite way I used at one stage to pick whether there was a true audible variation between, say, two file versions of some track was to load both into Audacity, align them accurately, then select a short but 'interesting' area of the waveform, and set off play in repeat mode for that tidbit. First run one version, over and over again - it becomes an almost hypnotic rhythm of pure sound, no longer music - at some point, flick over to the other version, without missing a beat. Any difference stands out immediately; the original, hypnotic pattern has been broken, and a new pattern is now clearly evident.

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14 minutes ago, STC said:

 

There was someone like him who used a JVC player, a mysterious clock, stacked speakers and 8 subwoofers. During his lifetime he probably visited more setups than anyone else and known to almost all audiophiles in my country. The first time I met him was when he tagged along another audiophile who came over for a visit. After he made several visits, he invited me over to his place but I procrastinated for along time considering his ridiculous setup. Finally, out of common courtesy I decided to visit him and invited my other audiophile friends to join me. None of them wanted to join me. Among them one guy who previously visited him criticized his bed room cum listening room (  10ft x 8ft) and his cheap system.

 

Having no choice, I reluctantly went there alone. 

 

That's when I learned not to look down at someone's setup based on the price or approach. I heard a good setup, much better than some tens of times more expensive system.

 

@fas42 could be doing something like that and until then I will give him the benefit of doubt. :) 

 

 

 

 

 

 

I've come across a few fellow travellers over the years - all have used very different methods and ideas; but the common element has been that they've understood that one has to be pedantically fussy about 'strange' areas of the setup. Of course, those areas are not really strange; they just happen to be focus points of weakness in the system, and by manipulating or addressing these the key audible artifacts, that really matter, can be sufficiently attenuated.

 

What these setups had in common were invisibility of the speakers, being able to project "big sound", a sense of deep immersion in the musical event captured in the recording - this is beautiful stuff to experience, and makes all the tiresome efforts to "get it right" worthwhile.

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8 hours ago, manisandher said:

 

That's a specific case of the more general hypothesis: a file played* back bit-identically can sound different.

 

Specific cases then include:

- different storage media

- different digital cables (spdif, USB, etc)

- different software player configurations (buffers, etc)

 

I'm considering which of these would be best for the ABX, and am leaning towards different software player configs. Once you're satisfied that there really is an audible difference between a file played back bit-identically (whether you hear it too, or whether I manage to prove it in the ABX), then we can certainly explore the other specific cases, if you have the time and inclination.

 

Mani.

 

I find it interesting that some people seem oblivious to the major changes in perceived quality when altering the software used for playback, and the settings used - on 2 quite ordinary PCs, one a desktop, the other the laptop, I went through quite a long process for each, on separate occasions, finding the player, and settings which gave the best sound over the inbuilt circuitry, and auxiliary, basic monitor speakers. Different players gave the best result for each one - and reverting to the much lower standard of the usual suspects is then quite irritating to the ear.

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5 hours ago, manisandher said:

 

Paul McGowan shares his views as to what might be happening here:

 

The relevant discussion starts at around 12 mins.

 

Mani.

 

Hmmm ... same ol', same ol' - isn't it amazing, that the precision of the analogue areas may be affected by noise and interference from digital circuitry behaviours, etc - gosh, you would think those stupid electrons on the two sides of the fence would know how to keep out of each other's way, rather than relying on human engineers to skillfully keep them apart   ...

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13 minutes ago, sandyk said:

Perhaps I didn't word my reply carefully enough, but many players such as .VLC Media player, Power DVD etc. etc. do NOT load the contents into system memory before playing them. This is especially evident when you play from a USB memory stick for example, where there may be several seconds of delay before play starts, where you can see it's LED flashing before play starts , then cease when  the file is being played. 

 

Sometimes it can be easy to pick the obvious signs of non-optimised programming - on the laptop, the "poor" performers were constantly driving the physical media throughout the track, and chewing up significant CPU cycles - Media Monkey, which turned out best, buffered like crazy initially - and then ignored the drive; and the CPU effectively went to sleep during the replay ... the effort had been made to tune the software to just do the job it had to do, well - by not disturbing the rest of the system any more than necessary.

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15 minutes ago, sandyk said:

 

 Dennis

 Yes, ideally you need some kind of reference. After quite a few repetitions, everything starts to sound the same.

 

Years ago, a friend had me perform a test with 3 x 20 pairs of  comparison files with the order changed randomly with a computer program he wrote.  He had taken my files and made up 3 new folders on a USB memory stick after shuffling them around between 2  HDDs on his PC, but none of his copies sounded as good as I remembered them sounding.

I then inserted a fresh copy of the better sounding tracks into each of the pairs as a reference, (a total of 3 tracks) which then helped me to home in on which of the original tracks had been my original preferred copy.

I had a falling out with my friend (DIY Audio member Greg Erskine) however, when I insisted that NONE of his tracks sounded as good as the new reference track.

 

Alex

 

 

 

The mind does play tricks on one ... :D

 

The brain is far too agile, adaptive - and "fills in the gaps" with great ease. Which is not the same thing as not being aware of what's better, especially when no special attention is being paid to what's going on. A powerful tool is the annoyance factor - something nominally sounds OK, but the longer you listen the more irritable you become - the music is bugging you, but you can't really say why. This is because the replay is close enough so that the brain then "fills the gaps" - but it gets fed up with doing this, and sends out signals to your conscious mind, "Hey! Enough, okay!!" ...

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8 hours ago, PeterSt said:

 

Better don't pin it down on the SFS. The number of parameters as such are quite countless (there are also parameters active nobody knows about because they are not separately/explicitly settable) and they each serve the same purpose : influence the DAC. It is the way how they combine that determines the SQ direction. like it is really possible to imply more bass or more highs or more mid, etc. Of course it is not about more mid and such but about how the more realistic reproduction is dialed in. It's only that it takes a life time (Mani feels he runs short of that) plus that with a new (approved) (W10) OS build we can start all over.

 

Even using a simple, untouched laptop, with the inbuilt sound system, including speakers it's easy to hear the variation when altering parameters. My HP, Windows 8.1, originally built as a "multimedia" m/c, sounds pretty blah!! with foobar - I looked around, Media Monkey looked interesting, and turned out best. Big array of parameters to play with, along the lines of Peter's software - and I narrowed it down to a precise combination to give the best sound - quality starts being lost as soon as one moves from the optimum numbers, etc.

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44 minutes ago, PeterSt said:

 

 

Mani, ehm ... No.

It only will testify that what you perceive for difference can not be measured.

It would be a good idea if someone write out a course (we call that a webinar these days)  showing how jitter measurement goes, what it takes to really show differences, and what gear it takes to capture for real what we're hunting for.

Line one would show : what you want is impossible, forget it.

 

 

I wouldn't say impossible, just very, very difficult - I've had a go a couple of times, but ran out of puff and motivation to follow through - the software to do the job has to be exquisitely crafted, to ensure that you are really measuring what you think you are; all the slippery but benign behaviours of analogue waveforms have to be accounted for, and nulled, to expose the precious lode of significant differences, in the context of subjective SQ.

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12 hours ago, manisandher said:

 

So what makes something like the effects of jitter so difficult to detect in a 24/176.4 capture, and yet (relatively) easy for the ear to hear?

 

Mani.

 

A program already exists which nominally does this, DiffMaker - and it may prove useful in this situation. But it unfortunately has many issues in its operation, and can very easily produce spurious, completely wrong results. The great problem is that analogue waveforms can be audibly identical, yet show huge differences with a program like this; and conversely with noticeably audibly distinct samples, show extremely close matching by the program, with a tiny difference which has the crucial difference mixed in with benign aspects! - it's "separating the wheat from the chaff" that's hard, but absolutely essential for gaining understanding.

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7 hours ago, PeterSt said:

 

Frank, come on now.

 

Tsk, Peter, I couched my post to indicate that the program could possibly be useful - a person "skilled in the art" may get significant results - I lost interest in using it because it had too many issues.

 

7 hours ago, PeterSt said:

 

It's the other way around. People of perceived fame use it to show you that any -160dB of difference can't be audible and thus is no difference.

In the remainder of your post you seem to say similar, but the way you put it you in the end say "nothing". You can do better.

Use something like this yourself and make some explicit sense out of it.

 

So in the context you put this remark (see first quote) I claim you make this up. Or hope it works.

Etc.

 

 

 

I hope that we gain access to the captures of the analogue out - I wouldn't be using that program, except out of curiosity ... so far, I have done exercises like this manually; I may find something, or I may not ...

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1 hour ago, PeterSt said:

 

And that is exactly the problem. Those differences ARE there as a fact and you can't do a thing with it for that reason. This is apart from the impossibility to align two different takes (only infinite ADC sampling rate would allow for that (so Mani, now you know).

 

Not true. It is possible to align the takes, but it's not trivial to do - I got halfway through the exercise of achieving an alignment sufficiently precise, but lost the motivation at that point; if the interest is there for this exercise, then it will be taken further.

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For the amusement of some - and related to hearing "subtle differences", I was visiting the audio friend down the road a couple of hours ago  - and his SQ was "not there". I realised that a power cord was not optimally held in position, and it was easy to hear the variation by very slightly altering this - leading to doing some "furniture" hardware mods to stabilise this - problem solved! Many here would have a fit if they saw how 'dumb' this change was - but, it did matter ... doing captures of this variation and finding a significant difference would be of the same level of difficulty as this exercise, here.

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12 minutes ago, PeterSt said:

 

It is not about aligning the takes. It is about the sheer impossibility to start the take at an e.g. 1/192000 of a second so the ADC sampler will start at the exact same time of the analogue wave thrown at it. This is how I said that only infinite sampling rate can do it; then you can shift afterwards to any point for alignment as there are no points (it's analog now).

 

I also said (earlier on) that "of course" this can be done with connected clocks hence one clock for DAC and ADC.

When that is arranged for (doable) we must still take care that all happens in the bit perfect domain of the OS (quite difficult) and we, for example, notice that it must happen on two different PCs (never mind the why for now and go ahead if you think you can avoid that).

 

That then you still have trouble in the alignment itself which is an aftermath thing, yes, that can be a clumsy task. So the software does that too to gain many hours of time and try things without hassle.

 

It's all about the fact that there will be some remarkably small measure of time misalignment - even if the clocks are synchronised, there will be a tiny time delay because of the finite time for the signal to pass through the circuit - one way to get around this is to resample at very two high, different  rates, with only a single digit difference in the rates; at a certain time the two will be in perfect sync.

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2 minutes ago, PeterSt said:

 

Yes, I thought you would be funny to come up with this.

And of course you next claim that you wouldn't "resample out" these subtle differences, right ?

So again go ahead with that but I don't believe you will be left with some result (differentiation).

 

Otoh I am almost sure that Mans is - or has been trying this. So yes, I'd expect something in that realm.

Thus Yes, technically that is a possibility. Practically I'm afraid not.

 

Given sufficient accuracy of the ADC, and the best methods for processing the data then there should be something there - I haven't pursued this far enough to be sure; and you may have taken this further without finding anything worthwhile - but there has to be some method for identifying the variation, if the ears can hear it.

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