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Ray P
Reply with quote  #46 
Looks like I'll have to try JPlay again. I purchased it some years ago but don't seem to have received any recent upgrades/notifications so have contacted JPlay to, hopefully, resurrect my licence and obtain a current version.

Reply with quote  #47 
It is the CPU and the memory with all the possible settings and combinations of settings (clocking, voltages, hyper threading, turbo mode etc) that can affect SQ of the player.

I don’t know if we need to call it “metaphysical”. We could also consider that physics of today is simply not advanced enough to explain these things sufficiently so that theoretical knowledge would perfectly line up with what we hear. Galvanic isolation and jitter might be just two areas of many more?

I think there is a point of satisfaction when everything we hear through our sound system satisfies our expectations which we have built up by our lifelong perception. It’s like listening to a piano player standing right next to a perfectly tuned piano. We might like or dislike his playing, we might like or dislike the piece of music he plays, but we would never have any issues with the sound of the piano itself. That’s my description of how it feels to be at this point of satisfaction. And it’s worth to strive for it.
Reply with quote  #48 
Yes, we must remember that nothing but the content in digital is digital - all steps of transmission, representation and reconstruction are analog.

So I guess that every change in the way and timing these steps are invoked prints through to what we hear.

Assume every push of a data packet upsets the circuits that receive and digest it.

Only natural then, that the cycle of these upsets is in the audio; and with standard buffers of 1,5 to 30ms that's a "transport jitter" frequency of between 33 and 666 Hz! A most important frequency range.

Indeed, quite intraphysical.

Ray P
Reply with quote  #49 
I've managed to recover my copy of JPlay 6.2 from a 2015 backup so it looks as though I'll be able to give it a try shortly. I'm still waiting on parts for my R2R project so will use my Linsley-Hood transistor headphone amp to do an initial check out.

Kelvin Tyler
Reply with quote  #50 
Ray, your R-2R project sounds excellent and is probably the way I should have gone if I was not involved in so many other other things. I may have been a bit impetuous in trying to take a quick, if expensive, way to the 'best possible'. Time will tell. Have not had any response from Soekris yet re raw take-off points on the 1541. Do I take it you have decided on the Soekris route despite your DSD considerations?
If the sound resulting from outputting a particular file from a personal computer varies with 'settings', then this has to be due to timing causing overall bit errors in the real time requirement that audio imposes. It is perfectly possible to transmit an exact copy of a file over USB, for example, if the timing constraint is removed. Were this not so the whole digital world would collapse. Presenting the data accurately to a DAC is, in principle, a straightforward counter/ shift register procedure. It would be nice to achieve this without all the extra 'baggage' required for a general purpose personal computer.
Now, to me, 'satisfaction' in audio matters is a very individual thing. I know that my perception has changed noticeably with advancing years. I require a bit more high frequency tilt than I did in my younger days. Others prefer a different tonal balance. The ear/brain combination is a very complex affair and is certainly not the same for everyone. Also, different pianos, however perfectly tuned, can sound quite different. Some will be satisfied with one, others with another.
As always all in the best spirit. Just my take on things.
Reply with quote  #51 

The piano example was meant as an analogy only. What I tried to say was that our sound system should reveal easily what’s part of the content and what’s part of the outer form so that we can focus on the content -the music- and our brain doesn’t have to decipher in the background what mixed messages it gets from the combination of the recording and the music reproduction part which would be the outer form in which the music is delivered to us. A good system should reveal all of this and show what is what so that the music can appear as what it is.

The importance of the “timing” in the digital format is obvious and if we try hard enough we can explain almost everything with it. But there are still some loose ends. For example: why does the sound improve if I improve the linear psu of my USB card (input and output still fully in the digital domain) and why does it affect some frequency bands more than others? Or why do I hear a clear difference between FLAC and .wav files (downloaded from  HDtracks or exported from my recording software) although there should be none? The difference is only in the timbre, not in overall SQ where time errors would show up.

Kelvin Tyler
Reply with quote  #52 
I cannot say exactly why the sound should change when the DC source for the USB card is 'improved'. No measured data to go on, but I do not feel any new physics is needed to find an explanation. I am aware I am stating the obvious, but any change you hear, whether in timbre or SQ, has to imply that different bit streams are presented to the DAC.
FLAC and .wav files may well be called lossless, but are not identical and are handled differently by software.
Reply with quote  #53 
Kelvin - try to post your question re raw output to Soekris on the DIYaudio forum, he told me that he answers there:

Kelvin Tyler
Reply with quote  #54 
Thanks very much, Achim, for your suggestion I put my question re raw output from the 1541 to the 'diyaudio' forum.
Since I last posted, I have carried out an investigation of the 1541 inwards and have identified several  points at which the raw output is accessible. None of these are particularly convenient and I would not recommend this approach as it could easily lead to damage. It is clearly not the intention of Soekris that the raw output be used in the case of the 1541, and permanently connecting to these points may well invalidate guarantees. Consequently, I will not be giving further details as I do not wish to fall foul of Soekris in any way.
Howsoever that may be, I did briefly apply the raw output to my audio setup. Result: very little, if any, discerned change. I cannot say why others observe such a noticeable SQ improvement when the output amp/buffer is bypassed in other cases. This would worry me if I did not reckon that my system gave  excellent sonic quality as it is.
For me, the only 'quantum leaps' have been moving to TS OTL amps in recent times, and the upgrading and the siting of speakers. I have two modes in matters audio: a 'testing' mode and a 'listening' mode. When in the latter mode, and listening to Nicola Benedetti playing Shostakovich, say, the last thing I wish to consider are the values of R14, C5 or the exact tube currents.
Reply with quote  #55 
Good on you! ;-)

Should you enter testing mode again - of the available factory filters I have checked between linear and minimum. Started with linear, as that was the default when I couldn't look into the setting as I had to build the serial connection.

When i had that, I changed to minimum, and have listened to that for 2 weeks.

Today went back to linear, which to me reveals most of the R2R qualities.

Have a good time listening!
Ray P
Reply with quote  #56 
I should be able to start on my Soekris R2R build this week; my power supply modules arrived yesterday and my chassis parts should be back from being machined on Thursday. I'll just be waiting for the transformer for the buffer stage then.

I'm looking forward to getting to hear it.

Kelvin Tyler
Reply with quote  #57 
Ray. It seems to me that another highly well crafted piece of kit is about to emerge. I will be very interested to hear in due course how the sonics compares with Wolfgang's splendid sounding setup. Things appear to be progressing rapidly on the Soekris front. How are you getting on with JPlayer by the way?
Recently, I have been trying a rather different, but not totally unrelated experiment. I have acquired a Pioneer XDP-30R hi-res audio player which can handle high-rate FLAC and double-speed DSD files. This closely resembles the Astell and Kern AK70 device, but is a bit cheaper at ~£350. I have been feeding the line output from this directly into my Masterpiece/300B PP/Heresey III setup. Playing a high-rate FLAC sample of the Mozart allegro of his 4th violin concerto is, to me, quite breathe-taking.
Some will no doubt attribute this to Mozart rather than to my kit! [smile]
Mentioning this totally solid-state source here is, I realize, tantamount to blasphemy, but perhaps this approach does bypass some of the issues which may arise with the PC/USB approach. Obviously, I am not using my Soekris DAC when doing this.
Apologies for this being a bit 'off topic'.
Ray P
Reply with quote  #58 
I finally found time this evening to install JPlay and have been listening via my headphone amp - extremely good SQ. Next I plan to adopt the two PC JPlay configuration, then start to seriously explore the JPlay settings. Looking good to have it all sorted for the Soekris DAC build.

Reply with quote  #59 
I am still using a single PC setup for JPlay so I am highly interested to hear about any SQ difference if there is one with a 2 PC setup/streaming.

My PC doesn't make more noise(HDDs, PSU fan) than the fans of my PS audio P3 which I use for setting the voltage and stabilizing the mains power for the OTLs for almost zero Ip fluctuations with closely matched tubes (next best thing to regulated HV without the side effects). So no problems with noise in my case. But the SQ could be an issue if it truly exists as some JPlay forum members have reported.
Ray P
Reply with quote  #60 
Wolfgang I think the main advantage of the dual PC set-up is that you can then run the Audio PC element in hibernate mode and it is shutting down all of the unnecesary processes (that hibernate mode allows) that delivers the SQ benefits.

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