XXHighEnd

All,

Since Januari 6 (which is a whole long week already) I have this still unopened box with 10K worth of audio measuring equipment;
It has been a dream for me to have such equipment, which will allow me to see what actually happens within XX at certain settings. Also, it will allow me to find theoretical best settings, or even other influencing means. So, be ready for that. :teasing:

I post this in advance to ensure myself doing it in the next coming days or weeks, and to at last prove that what we hear is no placebo.
Right now I don't know what I will find or where to find it. Part of it should be in the digital domain hence jitter, part of it will be in the analogue domain only (right behind the DAC).

The reason to at last buy this equipment originates from the Phasure NOS1 DAC which should be top of the bill, and which should go along with some real THD etc. data. But of course first there will be the process of improving by means of measuring and applying changes.

Whether it be XX or the DAC, you can bet that I will be measuring different things opposed to what is commonly done and accepted;
The fun is, that by now truckloads of experience tell me/us what influences how, and what we like and do not like. For example, I could grab an old XX version of which is appreciated it doesn't sound good, and compare with another of which is known it sounds good. From that possibly the properties can be derived which make what happen.

But now comes the combination ...

What is currently done via software so preciously and which is very fragile and far from absolute science, most propably and hopefully can be mimiced by hardware. If all is just about less jitter, we're done very fast. But I don't think it is and otther influences may play a larger role than we expect. This is why I will be measuring quite different things than commonly done.
If all works out as I expect, think of a Q1 button on the DAC that changes sound ... :veryhappy:

So far for now.
Peter

Right, time to report a bit about this. :heat:

[Edit : I removed the pictures from this post, while the text was not changed accordingly]

First of all, the measurement equipment (officially there for these kind of things) brought nothing. I guess by now this a kind of known, partly (or mainly ?) because I ranted about it a bit elsewhere, and I seem to have one follower at least.
So, for weeks I have been thinking how to approach this, and while working on the "glitch" software two weeks back, I suddenly started to see some lights.
Two weeks later, which would be last Friday, the first part of a program to do it was ready, and this weekend I have been measuring and measuring and measuring. A brief view in the directory I use for it, shows that I took 133 recordings of an 80 second track, and below Compare01 may give an impression of what I compared.

The main conclusion is : oh yes, I can measure the difference !

The means used comes down to recording the music data from the analogue out from the DAC, and compare those recordings amongst eachother.
As you can see in Compare01 for each same version two recordings are taken, which is necessary for the interpretation of the correctness of the recording itself. More difficult to explain (so I won't even start that) but it is crazy difficult to obtain compareable recordings, which is related to offsets, sample rate and clock alignment(s). This summarized, it took me days already to get similar results from the same situations.

On a side note, keep in mind that for each difference measured, the bit perfectness is still obtained. This now can be checked by means of the same software very easily, with the setup of looping back the digital line into a digital in of the soundcard. This by itself is no exciting thing, but the fact that the same software is just, showing zero differences always, is a nice thing to know for good feelings (about some decent wat of working).

What do you see when the same settings are used ?

For XXHighEnd this would mean the same Q settings and all, and in general this is about comparing a player with itself, performing two runs with the same settings.

First let's see what's in that 80 seconds 16/44.1 file :
There are 80 x 44100 = 1,764,000 samples for one channel. Note all measurements are about one channel, which just is sufficient (for this purpose and for now).

To my findings so far, when a random *same* situation is compared, around 82% of the samples will be equal.
This means that for our example around 317520 samples are not equal.

Keep in mind that the maximum value in one sample (for one channel, 16 bits) is 65536, but that any situation which is officially equal will have a maximum off value in any sample of 1 (and more rarely 2). This gives an off value of 317520.
See below Compare02 for an example.

These ratios apply always and spring from general phenomena not under our control, but count for my DAC in this case. It will depend on things I currently don't know, but I have a most strong feeling somehow the clock of the DAC is influenced. I derive this from everything I saw at comparing and the patterns I saw. There is more to it though like the receiver chip, and of course jitter coming from the SPDIF connection I used.
If you take jitter as the example of something not being under our control (once the DAC is there), it is an explanation of not every run at comparing the exact same settings, ending up with the same figures. So keep in mind : at running the same settings, only 82% of samples will be the same, and the off value in one sample will be 1 only. I assume this is inaudible when you run a track twice, of which I just have proven it will not sound the same, because it just doesn't come out of the DAC the same.

What do you see when the settings are different ?

Simple : the percentage of samples being the same goes down, and the off value will rise. But there is a very profound other difference :
The off value per sample will be higher than 1 for many samples.
See below Compare03 for an example.

I'll spare you the longer list for now, but this is from a relatively mild difference in settings, and the percentage of samples the same dropped from 82% to 81% only, but the off value rased from 317520 to 361724.
This is a mild difference, but something more bad is going on : it happens in patterns. So what you see in Compare02 recurs throughout, and no matter how small the increase of the value per sample is, it recurs every few ms and it will imply an audble frequency by itself (think about dither and why it should be random and you'll understand).

Compare04 below shows a much more severe case;
Here the number of samples equal dropped to 40%, and the total off value rose to 1,980,277. This is not only because the spikes are deeper, but merely because of a more consistent off value, which you can see in the screenshot (the parts without spikes, but with a consistent +++ etc.).

Keep in mind : this is bit perfect. :swoon:

I have seen cases comparable to Compare03 but worse, which are beyond my capability of explaining so far. There's rows and rows and rows with the value just being of in a rather consistent fashion. It seems that something can get confused and won't recover for a while. But, it always does ... until it happens again.

Not to forget : when I'm talking about things getting confused, this has to be seen in the context of the phenomenon happening, can be repeated over and over. With this I mean, that the specific comparison of two (settings) situations showing this, do not show when the two files recorded from that same setting causing it are compared. Then it's just the known 82% of samples being the same, the others being off by a value of 1 (and rarely 2).
So whatever it is causing this, just can be repeated. Of course this is just the Q settings, or the player for that matter.

Further explanation

I explained the above without all the context needed, thinking it would be more easy to grasp. But what actually is happening, is this :

If you look at the screenshots again (03 and 04) ... you see the relative difference between two recordings. Or in other words, you're not seeing the difference between the original source and the recording. It all is crazy difficult already, and that is just another step for me. Besides that, comparing with the original recording would be impossible with an oversampling DAC which changes the data in the first place (I tried it, around 1% of equal samples are left then). Comparing recordings each from the oversampling DAC, however, just works
So it is about relative differences ...

This means that the pattern I talked about, and that by itself turning into an audible frequency, may indeed be so, when the setting I used for the base coincidentally is the absolute correct one. And I don't know that ...
Keep in mind : whaver setting I use and take two recordings from, show the same "being good".

What it comes down to, is comparing many settings, and learn from the relative differences in between those. If A-B shows a pattern, but A-C shows another pattern which also shows at B-C, you can bet C is creating that pattern. Once this is known, it is a reference by itself.
From this kind of operating I learned that a very high Q1 value compares very much to a very low Q1 value, which btw was the very last I expected.

To end this for now : Foobar WASAPI shows a mild pattern compared with XXHighEnd WASAPI (Engine#3) just the same, when compared with XX-Q-4-0-0-0-0 (which seems safe). So far, I don't know whether Foobar creates that pattern, or XX does it. We only know XX sounds better, but we actually don't know whether that's because an unauthorised frequency riding on things. So now it becomes more complicated, and the Q sliders can be used to simulate something which is going on in Foobar. When the comparison shows the normal 82% and off value, we'd be listening to Foobar ...


 :NY02:

But it's not quite finished.
Along with some other analysis stuff, for your benefit, it will be in 0.9y (will take a while !), although I guess some of it will be available in the licensed version only. Must see about that.

Peter

Wow, what a post!

I don't claim to understand everything you've written here, but it looks like you're finally being able to prove that differences between bit-perfect players do exist.

Please let the rest of us (who simply sit back and enjoy the results of your hard work) know how we can help you.

Good luck in your quest!

Mani.

True ...  :) :)

Currently I am working on some nice graphing of it all, and then the real work starts ...
You know, this takes so crazy much time that I kind of hope for - all of us (who like to contribute) to make some "world map" of this all, so it can be determined what is right and what is wrong. I am sure this is related to the DAC as well, but since usually everbody agrees on the software influence in general (outside Q sliders) being better or being worse (whatever is/was actual), there's common denominators over DACs to find. It really can be interesting I think, with in the end a possible conclusion about what actually happens and/or whether it is jitter sipling through through everything. In any case, my SRC can't stop "it" and the results are the same with and without SRC (in the DAC which I can switch on/off).

But I also saw that the Fireface itself is more prone to being stable (immune) for these things. Yeah, a device we use for SPDIF passthrough, so we never tested that for software influence (at least I never did). The difference is : a direct data (Firewire) connection ...
or ... a better clock relation to the capturing section ...

Peter

Hi Peter,

Re: Measuring XXHighEnd ... With results !

Amazing analysis here and fantastic insight, well done  :good: even though I can’t pretend to understand all your explanations and hypotheses of this complex phenomenon.

Although not completely understood, in super simplified terms, it seems then that a .wav file played back with XXHE (using its various Q settings) or Foobah, both of which are “bit perfect” with respect to the digital signal fed to the DAC chip, can “somehow” result in distinct differences in the analogue signal output from the DAC chip, which of course is what is amplified and what is heard.

Regarding your finding that “when a random *same* situation is compared, around 82% of the samples will be equal” and that in “a much more severe case; the number of samples equal dropped to 40%,” which I presume was when *different* situations were compared, I wonder whether differences of >82% of the samples using your measurement system (which I think your saying will be quite audible) might occur then between a .wav file (with the same checksum) written to HDD using a BluRay writer and a CD writer? Based on subjective SQ differences of such files, I’d be extremely interested in comparisons that examine this. Completely understand though if you’d need to clone yourself to find the time to look into this.  :grin:

As your analyses here seem quite revolutionary and might provide answers to many questions surrounding why HDD .wav playback systems differ is SQ, thought I’d get in the queue early before other requests come in.

Cheers..Jeffc

Well, I must say I'm a little proud on this. Long from finished, but here's something for :teasing: :

05 is a part of 0.7 seconds from the difference between Q1 = 4 and Q1 = -4;

06 is a part of 0.3 seconds from the difference between Q1 = 4 and Q1 = 24.

Unattended, other Q sliders at 0.
Both snapshots are not throughout like that, but they are characteristics for the difference.

Let's assume we can hear this. Hahaha

It is obvious from the above pictures, that Q sliders influences the sound and it is not placebo the differences that we all hear.

So with this technique you can visualize the relation between 2 different bit perfect Q settings. It is useful for finding some similarities between those settings, but how can this help to find the most accurate reproduction of sound? Is it possible to compare the results, in relation with the digital data before get inside the dac?  

Is there anything showing Q positions influencing 'phase' changes?  It seems to me that phase changes occur when messing with the Q sliders.

Yes Dave, and you can actually see that in both above pictures;

The red line is how things should be. Everything above it deviates to plus voltage, everything below it to minus. And since there are patterns, who knows how this is perceived. Oh, note you see one channel only.

I think I saw my DAC switches absolute phase, btw most do. So whatever you want to see in the above, think about that.

But it is a good remark/question about that phase thing Dave. I already forgot about it, did not look at these pictures with that in mind, but indeed below a certain setting of Q1 we have always felt "phase change". "We" = many.

Ok, I think I'll be spending the rest of my life analyzing this stuff. Look what I found now :

Below you see a comparison between Q1=4 Attended and Q1=-4 Unattended.
What you see could be explained as breaking up sound. The point here is that this is a comparison between Attended and Unattended, and what you see happens exactly each one second. What is the main difference between the two ? one of them moves a time cursor forward, once per second ...
It always takes 12/100 of a second before everything is in rest again. Edit : I don't know how I got the 12/100, but if I'm looking again this is around 7/10. Sorry. Edit2 : No, sorry again, but 12/100 is correct afterall.

Anything else ?

Peter,

Just for my understanding:

1. You are measuring from the analogue output of your DAC.

2. All comparisons are bit-perfect.

3. In the graphs:
x-axis = time
y-axis = no. of 'off values'

Is this correct?

Mani.

1. Yes.

2. Yes.

3. Yes. Note that "off values" is the difference between two recordings. Only thourough analysis between different sets of comparisons may unveil which recording is really off, and from that may come which one is the most correct. Only if reasons can be found for seen off values, like the one from the last example and the time cursor, one comparison may be sufficient to declare one recording as being off in absolute sense. And also :


With the knowledge of the last picture, and without examining it again, I dare say that this too comes from the time cursor in Foobar.
So, once specifics are known and can be recognized, it becomes more and more easy to qualify what is happening in other situations. This will be a matter of experience.

Peter

Hi Peter,

It would be very nice to have a foobar-wasapi comparison vs xxhe q1=4 attended (which seem to be the gold standard)

Nah, Q1=-4 Unattended is the gold standard... surely :yes:

But an XXHE - Foobar showdown would be awesome.

Mani.

Your both wish is my command ...
Below you see (don't look at the window title !) :

(1) 1st trace 4-0-0-0-0 Unattended compared to Foobar WASAPI;
(2) 2nd trace -4-0-0-0-0 Unattanded compared to Foobar WASAPI;
(3) 3rd trace 4-0-0-0-0 Unattended compared to -4-0-0-0-0 Unattended.

Sidenote : The white graphs are the transients, as taken from the first (left one) mentioned. Generally they are the same for each trace because it isn't about that. However, it may look logical that anomalies in the DAC are cause by high transients. I could not prove this though.

Some notices (beyond the pictures you see, so trust me) :

a. (2) and (3) seem to be of opposite phase, which most likely is because of different offsets used. Strange explanation, but right now I have no other. Otherwise (2) and (3) are very similar.
Try to imagine the phase-change-thing, and that the base of (2) and (3) is -4 vs. +4 (the left one mentioned is the base).

b. If you look very carefully at the first trace of the first picture below (Compare09) you see that where the mousepointer is a "1 cm" repeating pattern occurs throughout. This is fragile, and note you're looking at a time span of 20 seconds here !

c. Compare10 has zoomed in (time span is now near 2 seconds) and with some imagination you can still see the pattern in the first trace.

d. While both (2) and (3) incorporate Q1=-4, you can see it masks all, and makes comparisons worthless. Keep in mind this is my DAC !

e. Compare11 is generally showing that some consistent thing is going on in my DAC; Where the mousepointer shows, right below you see the same but expanded (this is Q1=-4 doing that !), while the third trace shows exactly the same though in opposite phase, but dead sure again Q1=-4 doing it. If you look closely at the (1), (2), (3) above, you see there is no common denominator (none of the three elements, -4, +4, Foobar is in each of them), so my DAC must be doing this by itself. However, it is emphasized or it is not (like in trace 1 it is not).

f. Knowing that Q1=4 seems more reliable, Compare12 the most clear shows that in between Q1=4 and Foobar a pattern is present. It is mild though.


In a year's time I may have more firm conclusions. Just learning here. :blush1:

Mani,

Right after Q1 was introduced a few of us were seriously testing it and we were shouting to eachother to notch down after starting at the default of 14 (IIRC this was merely off line). At 4 things started to happen, and at 1 we were shouting things got wild and crazy. This was all in a time span of a week or so (slowly getting the grasp), and while at first being under 0 we were all impressed, we later all came back on that. Now, some 2 years further down the line, try violins. They don't work at these low levels. Things seem more detailed, but violins sound digital. Now :

I have always known (and told about it when it was necessary) that the values below 0 are special because they imply some randomness. And although I was talking into the blue to some extend, I knew about that randomness, that by itself creating a resonance. As you can see in the pictures where Q1=-4 is involved, you can clearly see it does something with a pattern of 3/100 of a second. How that sounds ? I guess more detail is perceived but in the end it sounds like more digital. Watch the violins.

Peter

Just wanted to quickly say I agree with you above Peter.  I got Q1 down to -2 for a long while and liked that... then, moved it up to 0 and felt it gave it a not so see through crystal clear view as -2, but did add some weight/meat to the 'notes'.  Then Pedal posted about 4 and I moved it up there and liked the balance I heard there and have stayed there.  I've listened to 14 for awhile and to me, it is 'woolly'/fuzzy  compared to  4,(don't take those terms to too much extreme, just in comparison).  14 not as see through clear as 4.  I HAVE NOT listened  to settings inbetween 4 and 14, but we should no doubt.

Also, as my sig says, I haven't been able to stay at Q2-0 Q3-0 and have moved them both up to 26 - 30.  Keeping them at 0 sounds like low Q1 settings compared to 26-30.  Moving up to Q2Q3-26 starts putting meat on the notes.  26 compared to 30 =     26 makes it sound more like lower q1 settings and 30 makes it sound more like heading toward 14 ... i thinks
Also, moving from Q2Q3-0 to Q2Q3-26-30 changes phase.  At Q2q3-0 I like phase ~ (normal?) and at Q2Q3-26-30 I like phase I (invert).  Even so with 26-30 there are slight phase changes going on (also phase preference is also recording dependent, but most recordings sound correct at those above basic phase settings given those certain Q settings).

I'm sure you're both right.

In all honesty, I've never really explored any of the Q settings. I started with Q1=0 (I think) and then tried -4, liked it and stuck with it. Q=-4 seems to bring my FF800 (used as a DAC, not an spdif pass-through) more to life. But I will explore all the Q sliders further.

My comment about Q=-4 being the gold standard was a bit 'tongue-in-cheek' - it seems to provide the biggest difference with Foobar WASAPI... ergo must be the best setting. But of course, this is flawed logic :1eye:

I'll give some violins a play and try to hear the difference for myself.

Really interesting stuff Peter...

Mani.

Wait, wait, wait.

I remarked that Q1=4 is indeed the gold standard because that's the setting that i like the most, not the one most different from foobar. That would be just silly.

I experimented with several settings. Anything under 0 was plain cr*p (=digital)
Tried 5, 6, 8, 10, 14 and the max. Above 14 things went too fussy. Between 4 and 6,8,10 I ended to prefer 4. Between 4 and 14, it is more a matter of taste. with some recording 14 (more smoothing) can be more pleasant, but I believe to be less true to the original.

Now, my comparisons with Foobar, which started about one year ago are interesting too.
I told Peter that I felt xxhe to be a BIG improvement over it only from version v (or w now i dont remember).
So they come after, choosing the Q1 and the Q1 has not been changed after that. I used 4 since the first time Peter said that it was his favourite. I tested again with x1 and my preference was still Q1=4. In that occasion i briefly tested Q2 and Q3, but didnt like them particularly.

I have to say that I didnt do much tests with those, and after the comments on those, I will do more evaluations.

???
I dont understand. You must have done a typo with some attended/unattended in the list above.

Telstar, no ... but you may be missing the -4 vs. 4 ?

Dave, FYI : As you may recall I deliberately set Q2/Q3 back to 0 to get the real merits from the stopped services. I kept that up for quite a while. However, the other day at testing with these analysis, Q2/Q3 by accident got stuck on 30/30, and it was only the next day that I started to recognize just what you said in the quote. Yesterday I payed attention to it explicitly, and it just seems to be so.
Even "meat on piano notes" would be a good one to state. Like I earlier said that a piano would get less dry from it, I now hear the same for bass. It's more spatious, and to me this comes as a good thing.

and Peter sets his specs back at Q1/2/3/4/5 = 4/30/30/0/0 (must be that left ear) haha

Whatever I tried I liked Q1/2/3/4/5 = 4/30/30/0/0 the most !
0/0/0/0 is simply to much dynamics for me, like it for 10 minutes then always go back to 30/30/0/0.
But the dynamics are amazing!

roy

Yes, I did.
I have one more comparison request then :)

+4 attended vs +4 unattended :)

http://www.phasure.com/index.php?topic=692.msg6125#msg6125  :)

ok, i'm blind :D

i think som weird "echo/decay"  appears when I move above 20 on q2&3
sound gets a little unstable
not bothered by too much dynamics at any setting
best
Leif

Could be,,, this is kinda what I might describe what I hear going from 26 to 30 with Q2Q3

Ok, let me throw this out so Jeffc doesn't feel all alone, (btw, I haven't listened to bluray ripped vs non yet, I will try sometime...)
Could you measure this for me?
When an album doesn't have a back cover, XXHE mirrors the front cover and pulls the song info and tracks/highlights which song is playing in a generated list displayed on the mirrored cover.
I would like to see measurements of this situation above vs the old way of just showing the front cover and info in bottom and top margins ()without a generated mirrored cover song list.
did that make sense?

Well, I now have accomplished the most difficult task of it all : compare the DACs output with the original digital data. Took me two weeks only, and although not directly to the benefit of XXHighEnd, I hope we audiofreaks will benefit in general. Ah, no, I'm sure you will, because I'm also 100% sure we're al hoaxed by official measurements. 0.00002% THD blahblah ... prrrrt.

Ok, my DAC is Non Oversampling, so it will produce 40% THD (or whatever measurements try to fool me with), but one thing : NOS will be the most representative for what's in the WAV file, and what that brings you can see below. And you could say that already brings nothing much.

The red line is where the digital data is, and the black lines show where the DAC plays (or where it's off). Top is the left channel, bottom the right.

Btw, if you hear a kind of p*ssed voice here, yeah, that's a kind of how I feel. This all s*cks so much all over that it seems a better idea to make my reel to reel noise free, somehow. On the other hand, it shows a. the truth of digital, and b. the enormous amounts of improvement to achieve.


Ok, what do we actually see ?

The white small graph around the red line is the representative of the transients. Transients are the more or less steep changes in volume, like a smash on the rim of a snare drum will create a high transient. The second picture shows the transients better, because they are higher there. It may take looking at your monitor from more below or above to clearly see them (because of the white on grey).

These pictures weren't taken to show explicitly what I saw : there is a relation between the direction the transient goes and the direction the DAC is off. When the transient shows below the red line, the "distortion" (the black line) will be above the red line. Look at the first lower dip of the black line (both channels) in the first picture, and see that the transients are at the opposite of the red line.
What functionally happens is that the transient goes to the minus (volt) direction, but the output wave by the DAC stays behind and is more posititve than the digital file. Remember, the digital file is at the red line, so when the black line shows above it, its voltage is too high.
The other way around, when the black line is below the red line, its voltage is too low.

In general you can say that there is no - no - no - way the DAC can follow anything of what it is dictated.
And remember, this is unlike the earlier graphs which showed differences between settings and players ... this is just the absolute reference compared with the digital file.


Now let's look in more detail to the first picture. At the second time the black line (both channels) come above the red line, you can see that the transients are not in one direction only (like more and more and more positive amplitude), but in both directions (high plus, high minus, high plus, high minus) which might happen with a fast frequency like from a snare drum. Now look at the transients from left and right, which are just about equal. However, the "distortion" is not equal ! The left channel gets inconcistent, and the one few samples it is more plus than an imagineable average line, and the next few samples it is more minus. And oh, that imagineable line shows at the right channel (bottom).
The bottom picture shows this even more clear, and I can tell you this is throughout; The left channel of my DAC is even more incorrect than the right channel. Something is just wrong, and it shows clearly. Note though this is at the "unmeasurable" (hahaha) uV micro level.


Allright, this is just an example of what can be done now, and as you can imagine it is now possible to find the anomalies of players or settings in an absolute sense. However, thinking twice you can also see that it is not said that it is more easy this way, and this is just because the DAC itself is so much off in the first place.

On a side note, think about this : when the DAC would better follow those transients, various other things would go wrong, like bursting speaker driver diaphragms when the amp can follow in the first place. So, it is clear to me this can be much much better, but to a certain extend only. When things become real good, other parts of the chain will collapse, I'm sure.
This is also the reason why hires material really doesn't make it better. The inividual digital steps will be smaller, but the general transients will stay, and the DAC won't be able to follow anyway. Not by a mile !

Oh, obviously this is from my "super duper" NOS1 Phasure DAC, and I can tell you that a new version of it is coming up. This time it will have a kind of infinitly more "fast" analogue stage. Yep, that's the attention point now : very fast analogue parts. And we will see what that brings ...
(it has been a kind of quiet around the NOS1, right ? ... not for long anymore).

Peter


PS: Before one starts to compare with the earlier pictures ... the horizontal (time) scale is equal to some of the earlier pictures, but the amplitudes you see in the black lines have been devided by 10 here !! (or otherwise the graph would be outside of the picture all over ... so bad is it).

PPS: I must put forward a very small disclaimer : The ADC (Fireface800 in my case) is involved too of course. But I think this can follow everything, just like the digital data and the high transients in the WAV file at some stage came from an ADC. This is just measuring voltage from one sample to the other, and making that a digital number. I don't think much can go wrong with that, as long as the "measuring device" in there can measure fast enough for the time it has for the one sample; I don't think a greater (high transient) or lower (low transient) voltage difference between samples will influence the value the measuring device presents. But maybe it does, and if so we're looking at the ADC for a more or less part, and all doesn't say much.

My own experience and everything I've learned from reading about that of other people suggests to me that AD conversion is much more difficult to get right than DA.

I really hope the RME ADC isn't affecting the results. If so, what you're doing is amazing...

Mani

Yes! Too quiet.
And I'm waiting for updates :D

Very interesting.

So now you can calculate a digital correction signal to get perfect output from your DAC?
Like digital negative feedback, maybe not realtime but still?

frightening thought :(
imo
best
Leif

 :blob8: :biglol: :grin:

Very evil thought.  :soundsgood:
I like it!  :pleasantry:

Well, some of you may have noticed the QControl tab being there right from the beginning, and that it contained an "Input" combobox for selecting a capturing device (the opposite of the sound device under Settings). Now the Q sliders got there, and the Input combobox got removed temporarily, but yes ...

... that has been my dream right from the start.

What you don't know yet - but I will write extensiveley about that later - is that the whole lot is already self contained. So for the measuring part XXEngine3 does it all itself, and it records while it's playing, and what it records is known to itself and it could be used in real time.
A more sad point - and this is unexpected - is that the data seems to be more off than what can be worked with. On that matter I can only hope it is the ADC doing that to me, which is yet to prove. But if so, solutions to that are in my mind ...

Now you know.
:)

(Ava, good !)

PS: I don't know if I ever get around with this really, but it can only work for an NOS DAC, and which is inherently filterless.

Edit : I stuffed in the picture below, showing the Analysis section (which nobody can see in real life).