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Author Topic: 0.9u-6 early thoughts  (Read 58768 times)
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manisandher
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« on: March 12, 2008, 12:26:28 am »

Great, absolutely no L/R phase problems, with or without upsampling smile...

... but my preference is still no upsampling. Sorry Peter unhappy

For some reason, the best SQ seems to me to come for selecting 'DAC is 16 bits 44.1 KHz'. I've just been comparing with Foobar/ASIO... and in this configuration, there is simly no comparison. XXHighEnd is simply more musical... almost more 'stable' and 'solid', if that makes sense. The way I described it to my wife was; XXHighEnd is 3-dimensional, Foobar/ASIO is 2-dimensional.

I could listen to it all day long... and still believe it is the best investment I have ever made in hi-fi!

OK, as far as this upsampling melarky is concerned, well it sounds very 'nice' and 'listenable to', doesn't it? But even though it is now properly implemented (well done Peter), I still believe it robs the music of dynamics, transients and presence. Yes, no upsampling is definitely more 'in your face', but I prefer that to 'laid-back'.

That's my subjective opinion, though I'm probably heavily biased, having never liked upsampling on my P70 transport either.

Now, objectlively, I have even more issues with it. As I've mentioned before, aliasing has a definite effect below 22.05KHz.

Sorry, but I couldn't help myself - have a look at the following graphs.

Would love to hear your subjective views on the sound though - please feel free to tear me down.

Mani.



* 20Hz to 20KHz DAC is 16 bit 44.1KHz.jpg (152.03 KB, 1189x649 - viewed 895 times.)

* 20Hz to 20KHz double upsampled.jpg (162.43 KB, 1259x648 - viewed 998 times.)

* 20Hz to 20KHz quad upsampled.jpg (172.79 KB, 1333x651 - viewed 1002 times.)
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« Reply #1 on: March 12, 2008, 12:42:46 am »

For those of you not familiar with aliasing, any signal above 22.05KHz is not 'real'. There is no information above this frequency (known as the Nyquist frequency) on a 'red book' CD.

Any signal above 22.05KHz is simply a mirror reflection of the signal below 22.05KHz. The problem is that it tends to interfere with the signal below 22.05KHz, if an anti-aliasing (AA) filter is not used. This leads, in my opinion, to a 'laid-back' though very pleasant sound.

Don't get me wrong, there are some very good reasons why you would want to avoid using an AA, but ultimately, it's all a trade-off.

IMO, the only real solution is getting hold of some high-resolution files... though with XXHighEnd sounding so good set to 16/44.1, I'm not sure if I'm going to bother for a while Happy

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« Reply #2 on: March 12, 2008, 04:13:22 am »

Well, what can I say ...

I based my enthusiasm (in the release notes) upon more perceived detail hence things there which I didn't hear before, things not getting wrongly to my brain, and an important factor - listening though the Fireface as the DAC with the pre-amp in between again.
I did not A-B anything, knowing that the Fireface as DAC already would make the situation a loosing one, BUT, I don't recall ever being able to listen to/through the Fireface actually hearing better elements.

After a couple of hours it started to loose its interest, but (for me) more importently : when I shut down the music at dinner time, my wife said she was glad I did. In the mean time I had been turning down the volume at some tracks (also an indication of the very wrong), and all 'n all ... it can only be so that this is not it. No matter how I want it (like the movie thing), it probably can't work.

Maybe if I do build in an AA filter afterall (of which I'm near sure it will be for the worse).
First some other things though. Happy

Quote
For some reason, the best SQ seems to me to come for selecting 'DAC is 16 bits 44.1 KHz'.

Can you please confirm that this is actually what you do and want to say ? I mean, this is different from "setting" the DAC at a higher bit depth (the frequency is unrelated), still listening to the native track (hence not doubled etc.).
The answer is kind of important (will tell later why).
Please, I don't ask you to sit down and (re-)listen (actually you should not), and only want to know what you meant to say.

Peter
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« Reply #3 on: March 12, 2008, 09:17:48 am »

Yes, selecting 'DAC is 16 bits 44.1 KHz' is what I do and what I want to say.

To me, this gives the best SQ for 16/44.1 files. I haven't done extensive listening, but so far, this seems to make an obvious and consistent difference. I will admit though that I have only really compared it to 'DAC is 32 bits 192.0 KHz'. I have not listened to any other bit depths.

For hirez FLACs, I select to 'DAC is 32 bits 192.0 KHz'.

Would love to know what you believe might be causing what I'm hearing.

Mani.
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« Reply #4 on: March 12, 2008, 09:28:41 am »

Quote
Would love to know what you believe might be causing what I'm hearing.

Haha, that's why I'm asking (and why it is important). Two other questions please :

1. Do you use the volume control (either case) ?
2. Could you describe the difference between both cases (always playing 44.1/16 of course) ?

Btw, I already know my answer, but want to learn the perceived difference in the end result of things. Also note that I never listen to the "DAC is" 16 bit anymore, because I use the volume control (and really want to use my extra 2 bits (from the 18 bit DAC)).

Thanks,
Peter
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« Reply #5 on: March 12, 2008, 09:35:49 am »

'DAC is DAC needs' only gives the possibilitys of the DAC in XX isn't it?
When no upsampling is choosing and playing 16/44.1, it doesn't change anything to the output when choosen 'DAC is DAC needs' = 16/192.

My experience with upsampling is that the advantage of upsampling is also related to the noisefloor coming from the powersource wich feeds the equipment and the sensivity of the speakers. With high sensivity speakers and clean powersource, i prefer 192khz upsampling. With lower sensitive speakers (<90db) en bad powersource Non oversampling sounds more dynamic, more 'raw'.


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« Reply #6 on: March 12, 2008, 09:39:38 am »

1) No. At the moment, I'm using either my passive vol. control with my Genelecs or the vol. control built into the Stax driver unit. XX vol slider remains at -0dB in both cases.
2) More difficult. It just seems more 'alive'. I tend to tap my feet more... if that helps. I'm happy to do some more serious listening... when I have more time (maybe over the weekend).

I've never used the XX vol control. Let me know if I should try playing with this...

Meanwhile, why does selecting 16/44.1 sound different to selecting 32/192 when the vol slider remains at -0dB?

Mani.
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« Reply #7 on: March 12, 2008, 09:47:42 am »

Well I am sofar pro-upsampling. I listened to it with exactly the same system as I use for 16/44 and I hear more. And don't find it giving more fatigue so far.
Wasn't this perhaps the longest time that you ever listened to the Fireface as a DAC Peter ? So in your situation I think it is remarkable that Twindac + digital volume is even challenged by Fireface plus passive, when using upsampling.

I went between the sampling rates quite a few times and for me the preferred sound was with quad. I will do some more listening tonight. I don't mind that at all.

And Mani how are the 'gonio's' with the version 6 I was wondering.
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« Reply #8 on: March 12, 2008, 10:10:33 am »

The genios are perfect with 0.9u-6 - the correlation remains at +1.0 througout... as it should Happy

Upsampling/aliasing has no effect. And of course, it shouldn't - aliasing won't introduce any phase differences between L and R channels... just within each channel and hence the interference above 5KHz or so.

You know, this discussion reminds me soooo much of a discussion I used to have with a 'hi-fi nutter' friend of mine. He has the full monty dCS digital rig. I used to have a Sony SCD-1 and Marantz SA-1 - I jumped on the SACD/DSD band-wagon as soon as I saw it.

But DSD never seemed to be right to me in the top end. So much like I'm hearing here with upsampling (especially quad). You probably all know that DSD has less resolution than CD above 8KHz. What do you think that does to transient edges... even of something like a bass guitar?

And here, quad upsampling (without AA) is down some 6-7 dB at 20 Khz...

Mani.
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« Reply #9 on: March 12, 2008, 10:13:52 am »

Quote
And Mani how are the 'gonio's' with the version 6 I was wondering.

Haha, I can tell you, flat as a (vertical) pancake (I tried that same plots to find all the bugs yes).

Quote
Wasn't this perhaps the longest time that you ever listened to the Fireface as a DAC Peter ? So in your situation I think it is remarkable that Twindac + digital volume is even challenged by Fireface plus passive, when using upsampling.

Well, you got my message right ...  Happy
On that matter it is my idea that anyone who compares apples with apples (which I can't, and I'm not sure whether Mani uses the same DAC in all cases) could come to the conclusion that Quad/Upsample (but into 18 bits at least !) is better. And as I said earlier, the sole fact that I heard more information on known records, kind of makes me urge to explore it further (but see next post !).

For others : keep in mind that we all don't use the same amps. However, when the battle has to be fought over Leo, Mani and me, they are equal regarding the speed (which IMHO is an important factor to the subject).
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A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #10 on: March 12, 2008, 10:51:52 am »

1) No. At the moment, I'm using either my passive vol. control with my Genelecs or the vol. control built into the Stax driver unit. XX vol slider remains at -0dB in both cases.
2) More difficult. It just seems more 'alive'. I tend to tap my feet more... if that helps. I'm happy to do some more serious listening... when I have more time (maybe over the weekend).

[...]

Meanwhile, why does selecting 16/44.1 sound different to selecting 32/192 when the vol slider remains at -0dB?

"Tap my feet more" is a perfect description. In fact the best where other descriptions fail.

Ok. Why do the both sound different ? well, for me this is perfectly explainable, but let's say for now that the code for both situations is different. And -so far- (see later) this had a purpose;

The 44.1/16 code has remained exactly the same as before to the point of sound quality. I did this on purpose, in order to not dissatisfy people which were satisfied. By itself this is related to the necessary code changes to allow the larger bit depth. Thus, actually the latter should have urged for different code for everything, but I just retained the "old part". In fact I was hoping for someone to notice the difference (and remember, I myself won't use the old code because I want the larger bit depth to be active, that by itself because of the volume control).

Now, because the old code too is (has been made) subject to volume control, *and* I wanted to pertain the old code for SQ, this is pre-processed for the volume control. After that has been done, te playing code is 100% the same. This is similar(ly woking) to how I can guarantee FLAC not making a difference.
Once a converted bit depth comes into play it was more easy to do this real time, and the necessary code for that influences sound.
And thus, we already can see the next sound improvement coming up yes for those who exploit the additional bit depth (which already is in order at 96/24 files, no matter your DAC can do 24 bits only, and which is related to all being transported over 32 bits ! ("DAC Needs")).

Quote
I've never used the XX vol control. Let me know if I should try playing with this...

For a few reasons this gives the major impact on SQ. But careful though, because your Hypexes may respond to the vast improvement on transients, and so far accepted sibilance could become unacceptable. I don't know really, since the only D/T amp we skipped at the time (about one year ago) was the Hypex, but all the others had unacceptable sibilance to begin with. According the principle used in there, I expect the Hypex not to be different, and I only want to say : be careful in your judging on what you actually hear / listen to.

When you use the volume control, you should not avoid the additional bit depth, unless your attenuation will not be more than, say, 24dB.
For now this means using the "bad sounding" (hehe) code, but I will change that since I now know that it makes the tapping feet difference indeed (no matter it is only you saying it Cool).

On the digital vs. analogue volume control much more is going on than allowing for the pre-amp to be eliminated. What I found on this remains a secret so far (The 0.9u what actually happened topic ... ).

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*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #11 on: March 12, 2008, 11:07:28 am »

Thanks Peter.

Just to make sure I understand correctly:

1) DAC is 16 bits 44.1 KHz -> data is pre-processed
2) DAC is 32 bits 192.0 KHz (or any bit depth > 16) -> data is processes in real-time

And this is what is influencing the sound, right?

The next improvement will be to pre-process data for all bit depths, correct?

Mani.
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« Reply #12 on: March 12, 2008, 11:07:36 am »

My experience with upsampling is that the advantage of upsampling is also related to the noisefloor coming from the powersource wich feeds the equipment and the sensivity of the speakers. With high sensivity speakers and clean powersource, i prefer 192khz upsampling. With lower sensitive speakers (<90db) en bad powersource Non oversampling sounds more dynamic, more 'raw'.

Allow me to say this latter is a good remark, or one I try to deal with explicitly anyway. On that matter, keep in mind the explicit change I wanted in 0.9t, which was about "more metal" where things had gone too plastic.
In all cases this can be compared with (digital) imaging, where added noise perceives (nothing more than that) more sharpness, while the opposite - denoising makes the image litterally plastic. Ok, this is not abpout (de)noise, but for "sharpness" things come down to a similar matter. The key to my last referred to link above, just *is* in the area of sharpness ... grazy

Quote
'DAC is DAC needs' only gives the possibilitys of the DAC in XX isn't it?
When no upsampling is choosing and playing 16/44.1, it doesn't change anything to the output when choosen 'DAC is DAC needs' = 16/192.

DAC Needs is a technical setting by itself, but when set to a lower rate than actually possible (which would come down to choosing a 16 bit setting for DAC Is (!)), higher bit depth files will be (explicitly) cut.

Your suggestion that it doesn't change anything would be true for your context given, but you are forgetting the volume control. Only for that, a higher bit depth setting is of life importance. yes

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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
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For a general PC :
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« Reply #13 on: March 12, 2008, 11:14:06 am »

Just to make sure I understand correctly:

1) DAC is 16 bits 44.1 KHz -> data is pre-processed
2) DAC is 32 bits 192.0 KHz (or any bit depth > 16) -> data is processes in real-time

And this is what is influencing the sound, right?

The next improvement will be to pre-process data for all bit depths, correct?

All correct Mani. Including the last one. grazy
There are some more combinations though, like playing a 96/24 over a 16 bit (set) DAC. That too is processed in real time, and that too can be setup differently.

Note that at some stage the advised 2GB of internal memory will go low ... swoon
(just think of 96/24 files being pre-processed; they are really over 2 times the size of anything which I had  in mind of needing to be in memory ONCE MORE hehe).
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #14 on: March 12, 2008, 11:32:12 am »


... but my preference is still no upsampling.

I'm in the same camp as Mani on not using the upsampling option. When activated, the precision looses a little something. Mellower and not as articulate.

As a typical test track, I refer to "In The Hollow" from the album "Asian Roots" by TakeDake with Neptune. For those who have this album, listen to the shakers just left of centre stage at the beginning of that track. In normal playback there is a crispness to the sounds of the contents of the shaker as it gets moved backwards and forwards. Go with the upsampling and the sound loses it's clarity a tad, almost as if they changed the material within the shaker itself. I could quote other examples, but I spent several hours this afternoon switching between non upsampled & upsampled and my preference was the same each time, regardless of music type.

Nevertheless, it's a good option to have within XXHE for those who have a preference to use it. I daresay different DAC's and equipment paths may well sound better with it activated.

Cheers all, and continued thanks to Peter for the effort being put into XXHE's development,

Russ
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Parameters (0.9z-7-4) ->Coming soon...
Parameters (0.9z-6-1) ->Same as for 0.9z-6
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