Different Q1 necessary with or without upsampling

[listening]

I can play setting Q1=-2 with upsampling (double). But if changing to normal mode the sound is disturbed and I have to increase Q1 to -1. Is it an option to deal with?

Cheers,
Georg

[PeterSt]

Hi Georg,

Is it an option to deal with?

I am sorry, but -no matter how simple the question looks like- I don't understand it ...

Sorry !
Peter

[listening]

Hi Peter,

is there an interdependence between the Q1 value and the upsampling rate?

[PeterSt]

Ah, yes, sure !

The higher the sample rate (and bit depth), the more has to be "pushed through" per time unit. This is not easy to explain, but let's say that with a 22 sample buffer size, this is 8 times more difficult at 32/176.4 compared to 16/44.1. Notice though that the sample rate doesn't influence the (implied set) buffer length, the bit depth does though. I can't even reason out this myself, but it is correct ...

So in general : the higher the bit depth / sample rate, the more difficult it will be to have a low Q1 value (at KS Special mode that is !!).

Peter

[JoaquinM]

Hello Peter,

I am being able to work with adaptative mode now. But I have noticed the following: If (according to your recomendations) I set Q1 at 0 and begin selecting buffer size, the smaller buffer size I can reach is 256. On lower buffer sizes I get the message: (Communication with End device failure). Then I set Q1 value to 1. All is fine... but now I discover that I can lower again the buffer size, up to 96 with no glitches or error message! Is this normal?

Joaquin

[PeterSt]

Hi again Joaquin,

Well, I will try to explain ...

First of all, I never thought of the possibility you people doing that. But when you do anyway, yes that would be normal behaviour. BUT :
Although you used Adaptive Mode to get to a setting without ticks etc. (glad it works for you !), you won't be using Adaptive Mode when you change the set buffer in XXHighEnd. Keep in mind : this set size is actually used with Adaptive Mode, and when used "wrongly" you will be using Normal Mode again. No problem of course - and I say it again - you just used Adaptive Mode to find a low latency setting for Normal Mode, but the sound will be different *and* (!) you may incur for anomalies regarding cpu usage hence may incur for more buffer errors than necessary.

A bit similar is this : once you found your device's buffer size (I am assuming you don't know it !! and it is all about that), I tell to increase Q1 to 1 (or higher). However, you can also leave it at 0, but you will see more buffer errors because of it. Also notice that these buffer errors will be impeeded by other things happening in the system (look for the ChangeWP log entries). So, Q1 = 1 is the safe setting, but still it may create buffer errors. If so, set it to 2 etc.
At changing the set device buffer size via the backdoor (like what you did), you will always make it worse. But, just like Q1=0 could do the job, making it worse via that way can also (just) work. The most important thing though is : at doing that (changing the set device buffer size) you will not be using Adaptive Mode anymore. So, summarized, Adaptive Mode is for :

a. Finding the lowest possible latency with a fair chance of no buffer errors in a fast means;
b. A different nature of the sound again;
c. The possibility to *check* whether you are without buffer errors, which possibility Special Mode doesn't give you.

Lastly, where Special Mode too gives you another nature of the sound, you will always be guessing and wondering whether you listening to "bit perfect" sound, because skipping a sample here and there sure is not. This is very dangerous, because skipping a sample here and there is inaudible. Skipping many is though, and the nasty thing of it all is : the lower the latency, the fewer samples will be skipped once things go wrong (so, a latency of 5 samples theoretically allows for 5 samples to be skipped per time (probably inaudible), while a latency of 500 samples theoretically allows for 500 samples to be skipped (very much audible).
In general we could say that if no glitches are heard (Special Mode) at track (part) boundaries, your system will for sure be able to cope with "playback at nothing else going on". To be honest, my system doesn't allow for that at 22 samples, and maybe 5 times per complete album I'm sure I perceive a glitch (Special Mode). But once I let Adaptive mode do its job, there are 0 glitches throughout an album; only at starting playback, which I think is normal because of really too many things happening there in order to keep up the ultra low latency.

I hope it is clear a little !
Peter