No Gerard, that's unrelated.

Gerard, note that theoretically you can do this now (0.9n) by means of creating a "main" Galery "All music", under which the data from both disks are created (2 minutes work I think).
Also note that you can "copy" again from a Galery to another Galery, so that this "All music" Galery really can serve you here.

Whether this is awkwardish for a solution I don't know. But anyway please note that another "more real" solution will give me sorting problems, hence that takes time to develop, and it can't have priority IMHO.

Peter

I forgot to tell ...

In the textfield besides the field where "blues" is entered (see screencopies in the first post) besides a time limit, now also an A (or a) can be entered. In that case a search will be applied to the "Album names", which actually is about the full path up to where the Track Name begins.
Virtually this allows to search for Album Names indeed, but also Artist Names.

Also, the new text field at the bottom, is now leading for where the search is applied.

Hi Gerard,

FLAC ... soon now.
MP3 ... later and when no other more important (sorry) things have been arranged for.

When the MP3->WAV doesn't work, something is not "official". Engine#3 is very critical to that.

Peter

Hi Gerner,

Although there is much truth in what you say, including the fridge thing , no, at judging things I'm listening right on-axis which is a necessity especially in this case (the more things could be harsh or unevenly loud, the better it is heard on-axis obviously.
Btw, how can I be offended when it is me stating that the whole idea is rather stupid ...

Talking about offending ...
For you and of course for Bert it may (or does) look like I am trying to solve a problem. A problem with my room ... a problem with the speakers, my ears, listening with my head in the fridge, whatever. But please read more carefully then. There is and was no problem.
I am just trying to "improve" further with say, completely illegal means. I know that, although I have some reasonings (not put here) which could make it legitimate. Could.

I think it is throughout my posts here that it is so as I described. Why do you judge it as solving a problem ?
May you not know it, I'm a guy of experiments. XX tells enough I hope. But also - and this is much more in the line of the subject here - look at my lamphood experiment with the Infinitiy's. If something is "illegal" it would be that. Infinity R90's being well respected speakers known for their VERY fresh highs, showing harshness immediately when something is wrong. So what did I do ? change the 180 degrees radiation of it and changed it into 20 degrees or something in that area. Now do the math on the dB increase ...

Bert, you were witness of that "stupidity". You also were there when we once removed them for fun. The well respected R90's became ... well ... R90's again. With the lamphoods it was a completely different speaker.
Did I try to solve a room problem ? (uhhm, theoretically yes -> less reflections) ... no. I just wanted more fresh highs (and again, do the math 180 degrees versus 20).

Today I act the same. Look back and read again. I said "it can have it".
"It" is everything. The speaker itself, and incurred by the player (since 0.9k or whatever) and the amplifiers.

Not to forget the reason why I started this thread : the generally perceived "too loud bad highs" in old recordings, which actually appear to be good highs with today's XXHighEnd and mainly : my "discovery" that highs must be squeezed in today's (but also 30 year old) recordings. Could be because of compression ... I don't know.

So remember, no problems, just experimenting, and what I hear I like.
And that's subjective ...

Peter

See 0.9n and Galeries.

Besides as explained in the above link, wrongly applied coverart can be deleted. So be careful, rightclicking on a picture in the Library Area and choosing for "Delete Image from disk" will actually do as promised (but a confirmation message will come forward).

The greyed out options shown after rightclick on a picture in the Library Area are not yet implemented.

No bugs fixes in the version.
No change of SQ to be expected.
Working with Galeries with the Grp checkbox ticked have *not* been tested.

Edit :

In the small textfield where the time limit can be entered (Library tab), now also an A (or a) can be entered. In that case a search will be applied to the "Album names", which actually is about the full path up to where the Track Name begins.
Virtually this allows to search for Album Names indeed, but also Artist Names.

Also, the new text field at the bottom, is now leading for where the search is applied.

ToolTips for the new fields have not been added yet.

Important : From off this version, 0.9n, non-licensed downloads (hence Demo versions) will be restricted to have output of 100 items in the Library Area only. IOW, the "Test" checkbox will always be ticked when the Demo version is used.

Here's a brief explanation on how to deal with the Galeries of the 0.9n version.
Much is to your own imagination, but here's the general outline.

When there's a result in the Library Area, you can select the Items you want, and with rightclick you can appoint them to a Galery.
In a future version you will be able to do this from within the Playlist Area. Not so now ...


So let's go.

1.
The first thing you'd do is create a folder similar to "Galeries", as shown in Galery01 below. Click on the button besides the text field as shown, and within there you are allowed to create a folder for it.
This will be your "root" Galery Folder.
Do this on your fastest drive, which is not necessarily the drive where your music data is held (but hinks of backups !).

2.
Now go to the Library tab, and let's assume from within there you can select music which suits a Galery. Look at Galery02 below;
It implies that "blues" music was selected.
In my case the Library Area got filled with "blues" music by means of the selection of "blues" in the Track Names.
How it came there is not important, but whatever it is, you are going to store it in a Galery.

3.
Select the Albums or Tracks (in my case Tracks because of the means the Library Areay got filled) you want to store in the Blues Galery, and press rightclick with the mouse ... See Galery03 below.

4.
Assumed that a Galery "Blues" did not exist yet, you can create it from within the selection as shown in Galery03 (Create new folder).
In Galery04 below this happened.

5.
Actually clicking the "Blues" Galery from Galery04, will save the selection from the Library Area to the "Blues" Galery.

6.
Now look at Galery05. With that button we can select the contents of a Galery. In my case I applied a sub-selection of all tracks of 6 minutes and longer.

7.

Whatever now is in the Library Area, select what you want from it, and press Load (Playlist Name empty !) to get the selection in the Playlist Aera.


Ok. Note that you can save selections from Galeries to other Galeries again.
And careful, a next version will allow for Keywords in a similar manner. For now : try to think of the difference between Keywords and Galeries. Make a plan.

But Klaus ...

I do not have full range drivers ... I have the same as you've heard at Bert's place ... 

This is not about a too much damped room;
This is not about drivers actually being incapable of expressing the highs;
This would be, though, about some "Traktor" effect. So, I must admit that this is actually about "DSP-ing" ... 

Please note, for all who think I am solving some "problem", that without the boost the in-room frequency response is, say, flat. So for Bert : you know that it is so, and actually it is just by your design. No problem with that, and nothing wrong with that. And thus :
after the boost the response is just as I told : an upgoing rise of 16dB starting at 5K and ending at 20K. In-room, anechoic, whatever.

If somewhere else the highs start to sound harsh, it can only be my conclusion that at this somewhere else the "base" for it is wrong. If the base is right, the only thing which happens, is that the SPL of whatever it is, is boosted. Same like cranking up the volume. BUT :

Of course this is theoratically about frequency ranges being out of balance. If that sounds like being out of balance, don't do it. But here it just does not. Keep in mind why I started this : cymbals sound way OUT of balance if the (measured) frequency response is flat. And of course, since we're all used to that, we leave it be. Or, possibly you just don't recognize it. Or ... you recognize it, but kind of know that when highs are cranked up, things indeed become out of balance. Or harsh, because of whatever it is (including theories ) that makes it so.

For Bert : you actually know how this started ... by means of an XX version which suddenly did not show no sissing/hissing from the tweeter, while coincidently I was looking for a perceived out of order tweeter (which was not the case). Before, the sissing/hissing *was* coming from the tweeter, and so easily heard at disconnecting/connecting it. Conlcusion : it just can have it now, and the only thing what happens at boosting those highs, is that NORMAL highs get a higher SPL. This is very very different from plain WRONG highs getting boosted, resulting in all the color going away, and sissing indeed.

Now, since it is looking like that improvements opposed to reality get hard to realize (it becomes hard to recognize what's wrong now), there's a very apparent thing lacking : the SPL of high frequencies opposed to reality. Look in this topic : you actually all agree with that (ok Bert, I did not see it from you I think). When a tom is hit at a realistic SPL level, I expect a similar hit on a cymbal to hurt my ears in the manner they should. And no, this is all not at 50cm distance being the drummer.

What I did, with 5K as the lower boundary, is boosting just the necessary to create realistic SPL of (graduate) frequencies above it. But mind you, at 7K this is already 3 times more SPL (6dB). This is (apparently) so much no sh*it that you think my system or ears is/are wrong and the speakers are under thick blankets. Fine ... 
And Gerner ... you know that I reasoned it out BEFORE I applied it, and it happened to work out as how (and why) I reasoned it. So no explanations after the happenings, but, say, well thought ...  (I was looking for a fishy smiley here, but it isn't there hehe).

A lot is related to the perceived grainyness, which by now has become a too difficult subject, because this is not there anymore (per Gerner's post somewhere about "infinite resolution"). However, this does not say it is not there in the old recordings (which are imcomparable with today's recordings, including those from 30 years ago (!)), and where all actually started around this phenomenon, all is solved by the same means.

All being said ... I *do* emphasize on the out of space stupidity on the idea itself, but since I keep on listening to it ... I must be out of space, or just go there.

Deaf Peter

Hi pedal,

Thank you very much. Very nice to hear.
But ... you did try the Q1 = -3 with Invert ticked, right ? because IMO (and of others) it just needs that ...

Peter

Yes Bert, I know. There is *no* logic to this, except then for what I said.
I could be very much wrong, but since I cope with this for such a rather long time ...

Peter

Klaus, no ... just the very capable Orphean "BMS" ...

Bert, of course I have expected a remark like yours. But although my writings are confusing as always, please read again;
There's contradictions all over, I know. The old recodings contain MORE highs, and by boosting the higher frequencies they turn into GOOD highs.

Btw, nobody told me microphones from those days could not reach frequency levels this is about (which is not about 20K but much much lower). It is about nothing more and nothing less than the grainyness I perceive from ANY recording, but since the old ones have more of it (SPL of high frequencies), they just express worse without filling those gaps.

But as said, my post is about later reference and the (honestly) sense and nonsense of it.
OTOH you could trust me just a little bit on my hearings. I mean, so far I will never let go of this high frequency boost. You (Bert) know what does and what does not hurt my ears. This just does not BECAUSE of the boost.
Isn't that strange ? And I'm using it for a month now ...

Well, maybe not exactly, but I thought it might be "fun" for (much) later reference to see what was right and wrong about some findings of me.

Already last year's Christmas I was so much surprised that the, say, really old recordings could give so much joy for sound quality. Think about the time recordings started to be in stereo for standard ... which was after the beginnings of The Beatles and the period so many Christmas songs were recorded.

Without now playing such recordings and without reading further, try to imagine why these recordings are perceived as bad and old recordings ...
...
Do you have a clue ?

Ok, because recording techniques weren't on par with todays means, you can say ...
Wrong wrong wrong.

I dare to say : because the recordings techniques from back then implied a "resolution", or a possibly other phenomenon we don't have wordings for yet, were so that they could not be played back 1:1.
And back to the clue I asked about above, this would be my answer from today :

Because of badly represented highs. It sounds raw, roughly smeared, and generally expressed : as bad old recordings.

For a normal human being this leads to nothing else than bad old recordings, but since the earlier versions of Engine#3 showed me at some stage that there's much more in the old cr*p than the old cr*p implies from it's description (old cr*p), it started to intrigue me. Possibly the main reason it started to intrigue is I could not get that back at later Engine#3 versions ...

I think it is one month or so ago, that I started to hear (per the then XXHighEnd version) that something in the highs was changed in such a manner, that highs could be added. Or better : *had* to be added in order to fill up gaps in resolution. Try to imagine : highs started sounding grainy (which might be the opposite of smeared), and the grainy to my perception was caused by missing higher frequency waves. So gaps, where there should be sound. Difficult of course because this is at the micro level, but anyway I perceived it like that.

Well, the logical solution to filling the perceive gaps was easy : crank up the higher frequencies for dB level; if something is there it should come out ...
Note : So this is about mangling with the frequency response curve of the loudspeaker, of which we all say it must be as flat as a pancake.

Not to forget : we are talking about perceived bad old recordings, which at least to my ears could be made concrete by the expression "raw etc. highs".

Ok. No matter how illegal or plain stupid it might be to start such an adventure, I let the frequency curve rise from off 5000Hz so that at 20000Hz there was a 16dB rise ... From 5000 to 20000 quite linearly. Mind you, this implies that the 20000 output is over 5 times as loud as the 5000 output ...

I must honestly say that apart from trying to get away the perceived grainyness, I was attempting to let cymbals and hi-hats output at a level my own drum set creates, keeping the general SPL coming from that in mind, compared with other instruments playing at their normal level with the toms from a drum set as the first reference (those not expressing highs as implied here).

I can tell you ... it worked.
Now an idiot drummer smashing around on the cymbals was perceived as an idiot smashing around on the cymbals. And mind you, listening to the normal frequency curve, this is perceived as -and as you all know- as some cymbals in the back, at far less SPL than reality tells you.
Is that in the recordings ?

Now a few things must be looked at together, and they are not easy to come with a decent explanation, let alone a scientific one :

The increased highs indeed filled up the grainyness.
That by itself *allowed* for increasing the highs in general. I mean, where highs are not good, they should not be over-expressed, right ? In this case they are technically over-expressed, but are perceived as just natural.
There is *no* layer added to voices, as you might expect when highs are well over expressed. This by itself shows that there's just no layer there. But also : the over-expression of highs allowed to happen only just *when* this layer is not there in the base. Appearently XXHighEnd now allows for it ... (and this was not the case in older (one month ago) versions).

To the above list I could add that I cannot imagine that "recording techniques" deliberately squeeze away higher frequencies, although everything is possible.
But now back to the old recordings ...

The first thing old recordings express (normal frequency curve, normal (CD)player) is too much of bad highs.
Well, with the experience now of one month at listening at the uplevelled highs it started to occur to me that old recordings express even more highs. They allow "more fluently" to smash the hi-hat in your face without perceiving it as boosted highs. But the highs are still boosted.
I did not go back to the normal frequency curve to test how the old recordings sound without the high boost, but I would not even like to go back;

Currently I am searching for all the old recordings I have because they sound sooo good and so beautiful. So much more realistic.
The only thing I can think of is that indeed somewhat after the day of poor old recordings, recording engineers started to decrease the level of highs. Mind you, it all fits. With normal playback means, the highs of old recordings sound worse the louder they play. So they really could have done that. But with the "newer" recordings playback of today (well, via XX) you start to hear the gaps. I do. Fill them up again, and all is back to normal.
Play back the old recordings at that same boosted highs level ... to me this doesn't give problems. It is really perceived as a somewhat harder hit at the cymbal.

All is *not* perceived as uneven highs. No hissing, no colouring going away, no layers.

As often, call me crazy ...

Peter


PS: For those who can tweak their frequency curve as indicated, try it ...

PPS: I have no real indication that all should start at 5000Hz. Important or not, no instrument has its fundamental frequency above that (except for maybe a synthesizer and pipe organs).

PPPS: Do note that I apply this at the passive side of the loudspeaker filter, hence it very well can be so that if you pump this kind of crazily boosted highs through your amps things might go wrong *there*.

Although in the anouncement of this version I told "No changes in SQ to be expected", this now appears to be far from the truth ...

Thanks to your feedback it got me thinking what actually could be happening at those places where now Q1 can be slided further down than before (and that's how important your feedback is !);

To me it already occurred earlier that with the same DAC, the USB connection did not allow Q1 to be as far down as with the SPDIF connection. I never talked much about it, because it was beyond my comprehension why.
Today, similar to your experiences, I too now (since 0.9m-1) can use the USB connection with Q1 at -4. But why ?

As a sidenote, but possibily as important, I wondered how people could not go further down than, say, 12, with a "normally looking" connection, whereas others can go to -4.
Also, in all cases where the boundary is crossed, people report real distortion like corrupted data. With my USB connection no difference.

Now, at the ever so long looking for the culprit in "Engine #3 did not start ..." I found another "issue" in my software, which in combination with now obtained better knowledge of things because of all the debugging - leads me to think about something "wrong", or to anticipate on anyway :

I am fairly sure that a DACs buffer can be overrun with a few bytes without it complaining. This is difficult to explain, but is related to the communication of the DAC with the outer world, and her telling about her buffer being empty to a certain extend, which implies to what extend the buffer can be refilled - hence data can be added;

Although I can't reason it out for 100%, it must be so that "conversion means" - one type opposed to the other - destroy the proper communication. Think of one particular sound sample causing the DAC to tell it reached a certain "buffer empty" state, that sample representing 0.000002 seconds, but the communication (by speed of light by itself) can be more or less direct because of less or more conversions (conversions : e.g. from USB to SPDIF, the other way around, etc.). Now, if something isn't working 100% properly this can cause buffer underruns (meaning : the buffer really gets empty and then sound stops). Btw, in effect this works out as sound stopping for only "some" number of samples, because the process as a whole will refill the buffer anyway when this happens, but it is the speed of light and the induced latency (oh yeah) of the software making this duration more or less samples.

Where in the above the underrun principle was taken as the example to make things clear, there's also the overrun I talked about earlier;
When, say, one sample too many is given to the buffer, it will ... no, might be lost. And this is what I talked about in the beginning of this post;
Looking at the Fireface, there's an official "safety buffer". I always wondered why, because it should not be needed when everything works ok ...
As I learned now (well, as how I perceive this now), this is about the DAC telling to re-fill the buffer with a certain amount of samples, but when -because of whatever (time) reasons - this is not correct, the DAC could communicate a few samples too many (to fill the buffer), and with the Fireface those arrive in the safety buffer, the hardware (Fireface) being able to deal with that anyway. Now watch this :

The debugging told me that where the Fireface allows to set the buffer to "a" number of samples (like 48, 64, 96 etc.) this is really respected by the Fireface at telling for how much to refill the buffer. And I mean, exactly. Stupidly enough, my DAC is not the Fireface ... it is behind the Fireface, connected via SPDIF ... Thus, where XXHighEnd communicates with the Fireface (read : the soundcard), the Fireface takes care of the communication with the Audio DAC. Read : when XXHighEnd is told to give 48 samples, within that implied time (which is 1ms) the buffer in the Fireface to feed the real DAC should not get empty.

With this connection (Fireface -> SPDIF -> Audio DAC) it is out of any control whether that succeeds or not, and really nobody would know how good or bad it goes. It largely depends on the Audio DAC ...

But now use the USB connection, with this same DAC;
Now XXHighEnd directly communicates with the DAC, although now the "fuzzy" USB is in between. Where the Fireface has a Safety Buffer to overcome anomalies (for overrun) the Audio DAC might have not. Or it just has ...

The latter can be about implementations of the DAC as a whole. Mind you, there's a DAC chip, and there's everything around it. The "around it" determines whether Q1 will give you anomalies at lower levels.


For those who are completely lost by now (not your fault) ... from the issue I found in the software, just creating the anomalies (soft cracking), compared with that now being solved and at least one user still reporting the anomalies, I can derive that those who still have the anomalies with 0.9m-1 must be dealing with something wrong around their PC->DAC communication. Generally (well, with my current knowledge of it) think of something being too slow to cope with the latency XXHighEnd can work with, hence that the latency of hardware products (like SPDIF->USB conversion) is higher than that of the software.

Lastly, note that before this day we always talked about "your system might be different for the workout of Q1" ... Today, I don't think this is true anymore. Today, too many people end up at the same Q1 level to be treated as coincidence. Now, if this is taken for being true, those who can't reach that level, hence have the best SQ at a higher level (but only incurred by otherwise loosing samples etc. !), must have something wrong(ish).
*If* this all is true, this is very convenient, because it means that I can concentrate on the single best sounding combination of settings, and they should workout the same for all of you.
Additionally I can tell that the Q1 = -4 is far, far from what the software is capable of, and at some stage the hardware response (latency) will come into play for everyone, that being different (for crossing boundaries) for everyone with different DACs, cables, etc.

Peter

Edit :
PS : Let me add that of course any comparison of software with hardware (like in chips) it moot. But do not forget : a DAC just runs on software too; though highly efficient, at some stage that might be less "fast" as the xxGHz processor from the PC ...

Yep, still planned. Also about extending the slider to below -4.
I've already been working on it, but technically it is not so easy because of running into (clock) stability issues. I think it can be done though, but it takes time to find the best way.
Why do you need "twice as good" each and every week anyway ?

Good idea Gerard. But be careful that this doesn't bother. -> just check something briefly, but since the checkbox is ticked ... nothing briefly there.
So all 'n all I'm not sure, but it can't harm *me* to make it.