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 1 
 on: April 16, 2014, 01:46:39 pm 
Started by charliemb - Last post by PeterSt
Ok, I have decided to let this topic be a sort of "progress made" topic which can be useful for myself already, but which also could be informative to those interested.

First progress made :
a. The "AI" filtering means can now (Redbook) upsample to 24/705.6 so I can compare with Arc Prediction like apples and apples;
b. It sounds like sh*t.

swoon

Of course this is only a first attempt but it can't make me positive. Actually it is so bad that after two hours of listening to it I was quite dead;

The last time I really tried must have been with the predecessor of the NOS1-USB, when I developed AI to my best knowledge. Now, this is long ago and since then I can be called spoiled with so much better sound since then, but this really looks way beyond "capabilities". So with nothing much more changed than the upsampling to 24/176.4 as we know it (but thus now to 24/705.6), this is what I hear :

1. The very worse : all sounds the same, and if there is one thing I can't stand it is that.
2. I thus hear the filter (and filters do make the sound).
3. All grey mush.
4. There is now life in anything and this is because
5. All sounds totally congested. Compressed if you want. Flat. Processed.

Before it is misunderstood, please keep in mind that the NOS1 I listen through does "nothing" to the sound that I know of, nor it has been reported by anyone that it does. It is just totally neutral, and if more neutral than that can exist then I don't know of that.
But therefore it is obviously the perfect machine to observe the filters through.

Also good to know : All the processing done is through a "pre-process", so what comes from that are giant 24/705.6 files (4 minutes or so is already 1GB) and they are played by the very same means as I do otherwise. Yes, I had to increase the RAMDisk size to 12GB to be able to play a few tracks, but this doesn't matter. That all is apples and apples does.

For this standard configuration (as said, as we know it from upsampling to 24/176.4) it is also nice to observe that the more lower frequencies measure a tad worse while the higher frequencies measure better. This latter is all logic because that's what such a filter it for. However, measuring (THD) is about periodic signals (sine waves) and this by far is not music. So the transient response is nowhere and I guess this is the complete smear I hear and which makes all sound the same and grey. Btw, this is linear phase and thus pre-rings as much as it post rings, and with a fairly (or very) steep filter (roll off starts at 20Khz and is done at 22.05Khz).

Next thing I am going to do is provide a nice set of settings for a few major response changes, so I can already use them myself to easily change things and listen. But please be warned : I see no single really helping solution; the revelation was just too big after changing back to Arc Prediction.

Regards,
Peter

 2 
 on: April 16, 2014, 10:56:24 am 
Started by PeterSt - Last post by PeterSt
This topic is meant to share DSP Configuration Files we as users of the Orelino / Orelo MKII Loudspeakers found for certain "applications".

I can be concidered one of these users and of course I created a "best base" configuration the speakers will be shipped with. However, all this means is that the DSP firmware in the speaker is loaded with this base configuration, ready for you to change if you want. But wat you actually change is the configuration file by means of a PC program and you upload that configuration to the speaker (different per Left / Right channel if you want).

Applications

First off, an application can *not* be about room corrections. That is, this is obviously nothing to share amongst us because if we apply room corrections this will be dedicated to our specific rooms and it won't work out in others.


Advice anyway : Try to avoid room corrections because only in very rare cases it will be necessary. But also : If we start trying the settings of others, it will not be convenient to
a. share any configuration because first the room corrections must be taken out;
b. load any shared configuration and try it because we would need to apply our own room correction first.


One of the applications could be the way the bass-mid crossover is set up; Imagine a "slope" coming down from the mid and how you can let ride the slope of the bass against that; officially there is one good way for that only, but with some experience more ways exist if first the slope of the mid has been made for that; Notice this is too detailed to work out in this post, but also notice that the forum board is OK for this when needed. Anyway, the one means to do this works better than the other, and all is about thevery best way the crossover works, which is as we know crucial.

A very different application would be about the balance between wanted SPL and the lowest frequency to support for that SPL. For example, while the Orelo MKII's standard configuration has been made to go deliver a straight to 19Hz (+/- 0.5dB) this is related to the reference level which is 88dBSPL. Now, 88dBSPL is more than enough for everybody with the larger room and normal usage and where talking will be difficult. Still, we audio freaks may want a crazy 100dBSPL with undistorted sound. Now if that freak also likes to play music containing those very low frequencies (because the music just contains that in the more rate cases), it can be a better solution to change the "straight to 19Hz" into e.g. "straight to 23Hz".


So just envision the downward going line at 19Hz to be at 23Hz and all further being the same. One difference : the speaker will now play (audibly) distortion-free at 100dBSPL (that assumed - I just took a figure).

While the example of "100dBSPL" possibly is not very realistic, the other way around can very well be : Create a response for e.g. 85dBSPL and let go the speaker straight to 17Hz.
This *is* realistic, because my tuned reference level of 88dBSPL the speaker is shipped with, is for my larger room - and any way smaller room will never utilize 88dBSPL (it will be too loud).

An application - though a strange one - in the same direction could be "movies"; Movies tend to show sub low, but what we will not be used to much is that this sub low actually is inaudible. So really, any 19Hz can not be heard. Still we are used to hearing it because our beautiful subwoofers distort so heavily that it just will be audible as distortion, never mind we don't notice that per se, and it is about the effect. And so, for example, the roll off could be made at 17Hz instead of the 19Hz as you see in the plot above and the movie putting out that 17Hz while say the level played is still 88dBSPL will now nicely "shake" the room. All what happens is that now maybe 1cm (peak peak) excursion occurs and while nothing will break yet, your 17Hz will be there (in distorted fashion and just like you are used to).
But of course we don't like this to happen with our music; there any 17Hz if present should pass by without even knowing that it's there (meaning : cut off at 19Hz when you play at 88dBSPL).

Another application is again in the XOver area, but now its "technical DSP". What I mean is that the offered various types of filters imply different (also cascaded) processing in the DSP (btw Digital Sound Processor). Now I don't like to turn it into XXHighEnd facilities, but in the end it is a kind of so;
We can well say that a more lean filter (less filtering means applied) sounds better. Also, the one type of filter just *is* more lean than the other, but obviously has a different target. Still it could do the job we hunt for, like "create that slope needed". And so as we say, more routes lead to Rome.

Lastly on this, people who obtain the Orelino will see that all said above sort of multiplies because the Orelino's specs on the low end are obviously somewhat less than the Orelo MKII. For example, I use the Orelino with a slow roll off from 25Hz so it turns up at -9dB at 18Hz with a reference level of 86dBSPL. In practice this means that there's now all kind of variations possible from letting it steeply roll of at 25Hz so a way higher SPL is allowed, via a slow roll of from 25Hz to e.g. 23Hz and let it steeply roll off from there and still allow for way higher SPL - through ... etc. etc. etc.



Of course, once you like to go working on your DSP filters you will need some ADC means, some (free) software and a measurement microphone; Once you have that (if at all) you will see that it needs quite some experience to get all where you want. But this too is what this forum board is for and I am there to help.

In the end, no worries; I too will be experimenting further and while I already have a few options ready for you, more are bound to come from my side already;
All together I'm sure that an already great speaker will be even more great just because of these software upgrades and because again your ear count will be more than mine alone. So what also is allowed is that you just ask for something like "hey, I want more bass". Well, since I can do that fairly easy I can make you such a config and upload it to here and you can try it out. But meanwhile I will tell what I did, how it looks (plot of the DSP response) and you will learn leasily how to do such a thing yourself in a responsible way.

I will add the first base configuration(s) when the first speakers arrive at customers and this will be soon.

Peter

 3 
 on: April 14, 2014, 09:48:12 pm 
Started by PeterSt - Last post by Scroobius
Quote
Djeezz, what an effct *that* gives !
Man, you just feel the "wave" going back and forth. Very special.

Mouthwatering indeed!!

 4 
 on: April 14, 2014, 06:31:06 pm 
Started by 2glory - Last post by 2glory
So I have read about the Clock mod and cap mod and I do have a thought about all that.

Take the cheap A$$ PC AC cable off and place a Dynamic Design Heritage/Nebulas or Stage 3 AC Zyklop cable on it and hear the sound change in the whole system turn for the better. Better bottom end? Yes!!  Do the same on the NOS1. Better bottom? Yes. Now place both on Star Sound Sistrum SP-1 stands. Focus and better deeper bottom end!!

What AC cables are being used by those who have done the mods or those who use the full Monty Phasure set up?

I for one would not want to change the design of the NOS1 with any mods on the inside but that is just me so come vacation here for free Happy

 5 
 on: April 14, 2014, 06:13:23 pm 
Started by 2glory - Last post by 2glory
So I have read about the Clock mod and cap mod and I do have a thought about all that.

Take the cheap A$$ PC AC cable off and place a Dynamic Design Heritage/Nebulas or Stage 3 AC Zyklop cable on it and hear the sound change in the whole system turn for the better. Better bottom end? Yes!!  Do the same on the NOS1. Better bottom? Yes. Now place both on Star Sound Sistrum SP-1 stands. Focus and better deeper bottom end!!

What AC cables are being used by those who have done the mods or those who use the full Monty Phasure set up?

I for one would not want to change the design of the NOS1 with any mods on the inside but that is just me so come vacation here for free Happy

 6 
 on: April 14, 2014, 04:58:17 pm 
Started by charliemb - Last post by PeterSt
Hahaha, yes. Since yesterday that's in my ToDo.

So when the AI button is clicked it actually sees this as a real time change and it start to convert and play.

You report the other way around I think (change to AP) but it will be the same thing.

Will be solved - and thanks ...
Peter

 7 
 on: April 14, 2014, 04:52:18 pm 
Started by charliemb - Last post by PeterSt
Well Charlie, I always hate it when I can't explain something, so here's an attempt undoubtedly full with idiocy :

As you may hace read elsewhere I measured HQPlayer (mind you at 1KHz only). And, it measured worse for all the filter settings (THD I mean). Not by miles, but still. Also, all but one leave more HF beyond the audio band (22049 Hz) than Arc Prediction does. Remember, this is measurement only and I never listened to it. Now :

What if that dither or noise shaping just matters for that poorer THD to begin with; I don't recall I tried with and without dither types, but maybe that is not important. So, if dither makes the THD better, it should also be audible; All what measures better is also audibly better (to my ears), which does NOT mean that when it measures the same there are no audible differences. But this is for another many days. Wink

So what's really relevant is whether I can improve by means of dither. And here another "watch out" : I never listened to my own attempts on this either - only measured (and saw nothing change - obviously as explained in my previous post).

And so we'll just provide the options and then we can all see for ourselves.

Something else :
I engaded that multi threading for AI the other day and had the idea it worked now. But while I never listened to it, this morning (which is 8 hours ago) I started measuring and noticed that the signal dropped after 15 or so seconds. So it took me that 8 hours to find out that this was caused by the parallel processing and things not being ready yet while some piece of the software did not take that into account. So yep, that's software, and all day I was looking for yesterday's changes which had to have caused it.

And then another thing you won't know about because it was prior to you being around :
Ever back AI worked with sneaky stuff so it could start playing while the remainder of the conversions (even of the same track) were still going on (quite normal for the other players because they convert real time). This caused so many troubles that eventually I disengaged it. But I will reactivate it, so we don't need to wait for ages before music starts. And hopefully make that decent now of course.

Regards and thanks,
Peter

 8 
 on: April 14, 2014, 04:47:45 pm 
Started by charliemb - Last post by charliemb
From another thread:
Of course, that post in your topic about 48->176.4 could have been a hint, but still not because you found a dozen of these kind of bugs before. Still, when I started this little project I was thinking of you ...

Thank you, thank you.

And this reminds me that I've forgotten to report another situation where I think some test code was not commented out prior to compiling.  Now mind you, I'm running 0.9z-9b (otherwise prior to 1.186a).  And I think it is related to the above.

If I am playing around with 1x 2x 4x upsampling and choosing AI vs. AP vs. nothing (repeating bits) filter, I usually need to change two of these settings for each A/B comparison.  All of this works normally under attended.   But under unattended here is what happens:

Say I want to compare AI at 2x against AP at 4x.  I start with AI at 2x.  Then:

1. play a track unattended.   Listen for a while
2. hit alt-x
3. once xxHE comes up I click on stop or hit alt-s
4. I change filter from current setting to AP.  At this point I also want to change the upsampling ratio from 2x to 4x,  BUT I CAN'T!!!
5. Instead, xxHE immediately goes into play mode as if I had hit play.  I can't select 4x

And so i now am forced to hit alt-x again, wait , wait, wait, and then hit stop and finally change the ratio to 4x.

The only way I can work around this is to click [c] between changes.

So  I believe that this is test code left over for you to be able to quickly switch between AI and AP at 4x.  ...and should be commented out before final compile for the version that goes out non-beta.

 9 
 on: April 14, 2014, 04:15:31 pm 
Started by charliemb - Last post by charliemb
Hey Charlie,
...  maybe also there's something special going on between us. I mean how big are the odds that I start working on this on day 2555 of XXHighEnd's life (which was yesterday) while only 16 or so hours later you ask for it with a couple of "please"'s.
Of course, that post in your topic about 48->176.4 could have been a hint, but still not because you found a dozen of these kind of bugs before. Still, when I started this little project I was thinking of you ...

Wow!  That's bizarre.  I think this might be the second time in about 8 months.  Maybe we have a telepathic connection. 

Quote
So I guess something of this will show up in the next version.

Thank you, thank you.  That will probably be freakishly good for us OS delta-sigma users, especially unattended.  And I predict that those who can choose a slower rolloff in our DAC settings will see bigger improvements.   And this will work in your favor because it validates the NOS1 / xxHighEnd as a concept because most the work is being done by the software filter in the PC,  and not by the filter in the OS delta-sigma DAC.  That is, your filter running under xxHE takes over and becomes dominant by far.

Regarding noise shaping,  who knows why each one sounds so different.  The only thing I can tell you is that it's audible even if it theoretically is too low in level to bother with.   Miska's NS9 sounds more airy (higher in frequency) than NS5 or Gauss1 at 4x.   Gauss1 is more evenly spread and in some ways sounds more natural.

It we can't hear the bit data influence,  then it must be the computational influence on the jitter.  Who knows.   Bottom line: it's audible. 

Quote
PS: I think I will move this to a separate topic.

I agree.  It is a different topic and also was thinking about quoting Mani and starting a new thread.  It was just easier to just hit reply and start typing.

 10 
 on: April 14, 2014, 03:21:22 pm 
Started by PeterSt - Last post by PeterSt
When I'm typing in here anyway again (previous post) ...

Day before yesterday I have experienced something quite new and exciting;

I was playing Rhoda Scott / Kenny Clark - Jazz in Paris.
Already at the end of the first track I noticed that the ending went super spacious. Ok, nice that was. But at the end of the second track - same story. Then I started to watch out for what actually happens and I think in the middle of the third track it happened again. So what ?

She switches on the Leslie at very low frequency (at also lower frequency of sound)

Djeezz, what an effct *that* gives !
Man, you just feel the "wave" going back and forth. Very special.

Peter

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