Blue or red pill?

[lmitche]

My two cents. . .

 

Having extensively applied the so called "JSSG" and "JSGT" treatments to my USB, Ethernet band and DC power cables, I hypothesize that moving switching DC current and digital signals (KHz, MHz or GHz) through any cabling creates radio interference that gets to the DAC through either the ground plane or the air or both. By playing the same file from a NAS or through local storage you are creating a different RFI pattern that will impact the SQ of playback in different ways. In the case of the NAS, the transmitting antenna is much longer then playing from local storage with, for example, a short sata cable. Reducing the emissions from the cabling through grounding and shielding, and shorter cable lengths, has great impact. Vary the length of the antenna and strength of the RFI signal and you vary the quality of playback.

 

This also explains the differences we hear in USB cables.

 

With about 100 foot of tinned copper shielding recently added  to my cable plant, one only needs a pair of ears to hear the difference, even on relatively short runs of cable.

 

Anyway, just my two cents.

[lmitche]

10 minutes ago, beerandmusic said:

^^^ Does anyone NOT agree that if there are audible differences that the recording will show differences?

and wouldn't you be able to see the differences on an oscilloscope on the input to the D->A circuitry?

 

You mean an objective measurement like this one?

 

[lmitche]

43 minutes ago, mansr said:

Those are mostly products of room reflections. A normal stereo recording doesn't convey depth or height information.

Mansr, this could not be further from the truth. I hope you get a chance to listen to some better systems. Height and depth are always there. Plus one can hear naturally occuring room reflections from the room where the recording was made. This occurs in almost every recording.

[lmitche]

8 hours ago, mansr said:

Please explain how any conventional recording technique is able to capture directional information, let alone represent it as a stereo signal.

This says it better then I can.  It doesn't address height.

 

https://en.m.wikipedia.org/wiki/Stereo_imaging

 

Here is an interesting writeup, first read on Absolute Sound, on height perception and sound quality.

 

http://www.enjoythemusic.com/magazine/viewpoint/0716/Why_Do_WAV_And_FLAC_Files_Sound_Different.htm

 

Lastly, I just need to listen to my system to hear natural height from recordings. If it is not there, something is wrong.

[lmitche]

6 minutes ago, mansr said:

Sorry, I should have said directions other than left/right.

 

Hahahahahaha!

 

Mansr, do you listen to recordings through headphones or speakers?

 

Also, do you perceive height in natural surroundings? Like if someone is talking from a balcony or hill in front of you?

[lmitche]

5 minutes ago, mansr said:

Speakers.

 

Of course I do.

Ok, so how do you perceive height with only two ear drums?

[lmitche]

6 hours ago, mansr said:

Through the varying effects of the outer ear and head on sounds arriving from different directions. A stationary source is difficult to localise without slightly moving your head. Regular microphones capture none of this. Binaural recordings heard through headphones can get partway, but they are still quite limited, just like a stereoscopic image is a far cry from a full hologram.

So the Superposition Principle works differently for microphones then eardrums? 

 

It is perfectly clear from listening that, for lack of a better term, "height cues" are captured by microphones and are reproduced by most recordings. Sure, microphone position matters, but usually, no always, height information is there.

 

I am happy to agree to disagree in this one.

[lmitche]

1 hour ago, Kal Rubinson said:

But the ceiling (or lack of one) is responsible for a portion of the ambiant sound field.

Yes, the difference in the arrival times of first arrival, floor and ceiling bounces likely account for the perception of height.  I'll have to listen to some outdoor concerts to learn what happens there. Similar timing differences likely add spatial perception on the horizontal as well.

[lmitche]

9 minutes ago, fas42 said:

 

What matters is that our ear/brain is very, very clever at sorting out what the sound field means - vastly more than the simplistic sound reaching the left versus the right lobe at different times explanations always put up in audio forums, ever give credence to - we're not robots when hearing! What we perceive can be manipulated by knowing how that extra cleverness works - some interesting experiments are being done in Auditory Scene Analysis, exploiting those "higher levels of interpretation".

 

With recordings, this cleverness can only be useful if all of the key information is not so blurred and confused as to be counter-productive. The latter is what yields messy, unlistenable to recordings, when playback quality is not sufficient - the brain gives up trying to make sense of it.

 

While one attempts to explain everything using simple arithmetical explanations of the sound waveforms, then one will always fail to appreciate what the mind is capable of - and better subjective quality will still remain far out of reach, no matter how hard you stretch to grab it ...

Well yes, of course. Just think about the fact that we can record vibrations from multiple instruments using a diaphragm that is capable of being in one only position at time, convert it to an electric signal that can only be in one position at a time, and then pump it to speakers that can be in only one position at a time, reaching an eardrum that can be only one in position at a time, and somehow the ear and brain can unmix the signal into the original multiple vibrating sources.  Completely amazing!

[lmitche]

57 minutes ago, gmgraves said:

I've answered that query before. I don't record commercially. I record for civic orchestra organizations, wind ensembles, "big bands", individual musical groups such as small Jazz bands, and soloists such as pianists. The only commercial recording that I ever worked on was for the Musical Heritage Society. It was a lute recoding. It hasn't been in the catalogue for more than 20 years.

George, is this you?

https://www.discogs.com/artist/381277-George-Graves

 

[lmitche]

So is the Mans and Mani meeting still on? Did I miss the results?

[lmitche]

@PeterSt, could you please give us a clear explanation of the SFS parameter? It sounds like "sector size" in disk io.  Is that correct?

[lmitche]

This trial went exactly as expected. I'm not surprised in the least with the result. By changing the SFS parameter, you changed some characteristic of the noise generated by the PC during processing.  This noise gets to the DAC which changes the sound. . . diifferent noise, different sound.

 

We know from the impact of the JSSG, JSGT, dual rail PC power, ISO Regens LPS1s and many other tweaks that this is so. To think argue otherwise shows a lack of experience with computer audio.

[lmitche]

3 hours ago, manisandher said:

Ambient RFI noise may well have some influence at the d-a conversion stage, but in our case it would have remained constant during the A/B/X, so can't account for the audible changes I heard. Unless I've misunderstood what you meant.

You likely changed the frequency of that noise and it's impact in the DAC with the SFS parameter change.

[lmitche]

6 hours ago, PeterSt said:

 

Actually this is not correct in the context you (seem to) put it :

 

 

There is no such thing as an "SFS various filter" and SFS is not a filter as such as well. It is a buffer with a virtual "analogue" means of lengths (to set), hence within a predetermined range the number of settings is virtually infinite. It also doesn't work out as a filter. It implies "noise signature" (together will all the other dials).

None of these alter the stream, so all these millions of settings possibilities will produce bit identical streams.

 

This is different from upsampling filters (which are for reconstruction), which do change the streams. But these were left out of the equation (were always set the same). Just like volume was not changed in any event. This is all the most obvious for "us" and therefore is not explicitly mentioned.

Peter,

 

So is the SFS parameter kinda like the thickness dial in a meat slicer?  You get the same amount if meat at the output, but the numbers of times the machine makes the slice (frequency) changes at the two settings?

[lmitche]

3 hours ago, manisandher said:

For example, I didn't hear more bass, or more highs in one over the other, or anything like that.

FWIW, I had the pleasure of hearing Patricia Barber live on Monday night at the Green Mill in Chicago following the Axpona weekend. I sat very near the piano inside the zone of enhancement with the PA speakers behind me. I heard a clear direct point source of sound to every instrument and voice on the stage.

 

The sibilance in Patricia Barbers voice is natural, I could hear it directly, but not really noticable unless one focuses on it. But it is there nevertheless so don't expect zero sibilance on her recordings. That would not be right.

[lmitche]

1 hour ago, manisandher said:

 

OK, perhaps my idea of slaving the Tascam's wordclock to the MOTU's wasn't very good after all:

 

 

Perhaps I should have slaved the MOTU to the Tascam? But I felt that the wordclock should sit as close to the ADC chip as possible.

 

In any event, the analogue captures should be totally unaffected.

 

Mani.

How about we record the DAC output to analog tape and later cut vinyl for remote particpants? We could even put track A on one side and track B on the other! 

 

I heard somewhere this has been done before. It should be feasible.