You are completely correct Edward. In the heat of the battle I missed out on a few. But I tried to indicate the essential : one does not connect to the PSU of the PC, and the other does (or may do, depending how things are separated inside the DAC box; in the TwinDAC's case it's PC->USB->SPDIF->DAC).

My too fast writing (or better : your correction ) also reveiled another PC-PSU driven device : the Fireface.

Again, interesting ...

But wouldn't it be hard to prove what you actually hear in this case ? I mean, a DAC running on batteries is generally appreciated as sounding better, although that too might not be because of the more clean battery power itself (if true already, but that's another thing).
I mean, when the DAC runs on batteries and is USB connected, it will always receive the power from the PC's PSU (I think). With Firewire it would be the same story. Now, what about the DAC not being fed by batteries but by its own PSU. Still the influence via the USB/Firewire cable would be there, but now it's mixed with the own PSU.
Btw, nothing tells me (for all of the DACs existing) that the DAC using an own PSU would use parts of the USB/Firewire power.

What to do with this ? well, I can imagine that I detach the power to my soundcard (Fireface this case), same with the DAC (TwinDAC+), and feed them by the PC's PSU. That would be new, right ? hahaha


All brings me to another serious question (for myself anyway), and that's about the kind of huge difference I perceive between SPDIF and USB towards the TwinDAC+. I talked about that earlier (maybe not on this forum), but this is about the both connections (which the TwinDAC+ swallows) being so different, while at the same time it is nearly impossible to choose one for being the absolute best. Here too I dedicate that to jitter, USB being better in the bass, SPDIF being better in the highs. But now, what about this :

The SPDIF connection goes through the Fireface (PC->SPDIF->Fireface->SPDIF->DAC) while USB goes directly to the DAC (PC->USB->DAC). So, I tend to say the difference comes from the rather different route. However, when it indeed would be so that the PC's PSU could pass on the signature of XX, it would do that via USB, and it would not do that via SPDIF. Or at least musch less, because there's no official power carrier in SPDIF.

At least for myself it is time to re-evaluate the USB connection, which I did some 4 months ago for the last time, when the performance of XX was quite different.


For those who jumped into this thread without background : Burning audio CD while XX playing.

But I don't think these devices can output (nor process) more than 16 bits ...
But maybe you did not intend to say that.

The TwinDAC+ is 18 bits and I don't think "twin" is related here. 


No no. On the matter of the digital volume I could agree, but my other argument was the playing of 24 bit material of which more and more becomes available (also see Linn Records 24/96 master material available for download ).

Peter

Hi David,

Well, it would hardly be a "commercial" thing to handout a player of which I think it is the best around, with degraded SQ to start with !
The funny thing may be that even when so (degraded SQ in demo mode) apparantly people like you still think it's okay, or anyway good enough to pay for un-demoing it.
So the commercial now is : it will only get better ! 


Oh, but I already know it, and that much that I don't even need to try it myself to believe you. But as said, it is unintentional because at the time of creating it I did not know it at all, nor could I expect it for the life of me.
But please let it be the secret of this player, and let it be a guide for further future improvements. Btw, that these things matter (whatever that is ) I only know myself for a few weeks.

There's more to come. I promise.
Peter


PS: I use Scheme-3 as well. Scheme-1 is (a lot) more digital, but with more (too much in fact) dynamics as a bonus.

Oh Chris, I feel honoured to give the one and only real answer. Ahum.


First of all, the term "oversampling" as the result of an activity, is not the same as the result of the term "upsampling". Both activities are the same (or similar at least) : they increase the samplerate. However, with oversampling the objective is something which is in the internals of the DAC, and with upsampling it is about a functional aspect, for you.

When a DAC oversamples 8x, you won't notice it at the ouput rate (is still 44K1 or whatever it was fed with).
When a DAC upsamples, you will notice it at the output, and e.g. 44K1 has become 96K.

What makes all the answers at the (by now derailed) AA thread confusing, is that the various causes for oversampling are mixed, not mentioned, or not mentioned all (in the one post). Example of this : when one says that oversampling is needed for the (brickwall) filter to operate, the other says that such a filter is needed because of oversampling took place. Both are right but it is not important to the one wanting to know whether he needs an nos-DAC or nor. What *is* important, is why either can be left out (if at all) and which actually causes what to be there/happen.
Yeah, I'm confusing deliberately, but I'll take that away as fast as I introduced it :

Everyting and all is about the type of DAC, which differentiate in two main characters :
1. Ladder oparating DAC;
2. Sigma/Delta operating DAC (or Delta/Sigma, never mind).

The Ladder DAC is the most genuin DAC because (I think) it's principles were used in the first CD players around, by itself because of the logic of the principle (which is dividing by 2 the same way bits for a byte with decimal value; the ladder DAC is also called R2R DAC (R becomes 2R).
An example would be the TDA1541 (chip).

Today many R2R DACs (as a chip) exist, but the 1541 has a specialty : it is so old, that it doesn't contain the logic of the necessity to have a filter built in. Might you be into digital photography, this is nearly the same as an Anti Alias filter in a digital camera; it needs to be there otherwise many photo's will fail (by moiré effects), but when not there, the photo's which didn't capture moiré prone subjects (like the roof of a house) are so much more bright and better.

So, the first CD players contained this lousy DAC without "AA filter". This was subject 1.

Subject 2 is about 16 bits not being enough in todays digital audio world, and 48bits being the kind of minimum needed when mixing/mastering audio. However, R2R DACs with more than (I think) 23 bits do not exist, so they invented another way of, say, emulating the R2R principle. This became the Sigma/Delta (Delta/Sigma) principle, which in fact is about fewer bits (could even be 1 I think) with which a kind of unlimited bits can be virtually created.  In your Fireface such a S/D DAC resides, and it can do 32 bits I think.

So this was subject 2. Now we must combine things :

A S/D DAC needs oversampling (aha !) in order to "create" its virtual bits. It will depend on the number of real R2R bits in there - in combination with the number of virtual bits needed in the end - how much oversampling is needed. But it can be 8 or 16 or anything (up to 256 I think). Maybe I saw 4 somewhere.

So we can come to one intermediate conclusion at this time : A Sigma/Delta DAC needs oversampling to operate.

While "upsampling" - that being the same as oversampling for the internals of the DAC - is a nasty thing to happen, because squares become sinuses (the higher the converted (!!) samplerate the worse it becomes) ... the next thing happening is that the oversampling principle *needs* AA filters because of the process happening (never mind the process itself).

Next conclusion : A Sigma/Delta DAC needs AA filters to get the sound back to normal after oversampling.

What's left now is the good old R2R DAC which exist without and with AA filter;
Those with AA filter (which are about all of them), have that filter for one purpose only : the idea of it being necessary to perceive good sound. Of course, one can measure "aliases" when the AA filter is not applied, but as often in audio, things are a tradeoff, and this is - or looks to be one of them : what the AA filter repairs, destroys something else. Why ?
Well, here it comes : in order to let these well thought filters operate, first a good dose of oversampling has to be applied. Roughly it comes down to shifting noise and other anomalies to the inaudible spectrum, do some work there, and then go back to the original sample rate, applying some (necessary) filters again.

Kind of conclusion : Nearly all R2R DACs use oversampling in order to be supposedly better by means of filters for various reasons.

I say "supposedly" because I don't think there's science which can prove it true to be "better".

Last conclusion : R2R DACs allowing for no filtering and not needing oversampling to achieve their number of bits, leave the waves in tact.

The latter is a fact (leaving the waves in tact), and the difference with the R2R oversampling/filtering DACs is in the area of distortion of various kinds. Another fact is that no-oversamping cannot cause additional (Aliasing) distortions. So at least no filters are needed for *that*.


Which type of DAC is best is subjective. But note the big apples and oranges contest, where the filtered DAC measures better, if you only forget to measure the shape of the waves and forget that there might be more squarish waves than sines around.

Also note the huge dilemma : already 24 bit nos-DACs (chip wise) do not exist (not to my own research). I think 23 bits nos chips exist and by combining two you you'll get a 24 bit functional DAC. A functional 24 bit nos (box) DAC existed in the past, but I don't think currently its still in production. Now also note the reason why you'd want a 24 bit DAC : digital volume control for 16 bit material (but that only allowing for 24dB attenuation without losses, which hardly is enough) and of course the playback of 24 bit material (that coming along with 88K2 or 96K samplerate).
Sigma/Delta DACs can do that easily enough, but they sound lousy compared to R2R DACs already by themselves, and additionally they oversample to get the (bit) job done.


Well Chris, was this something for an answer ?
Peter

uhhm ... This is quite some commercial !

Yes, I can imagine this. Well, anno December 2007 I can.
At the time I started the development of this I would never have thought that the things which cause this would be able to make a difference. Today I can predict it.

Not that I ever thought about that what's in there to cause the demo to be a demo sounding different. So of course it is unintentional, but also kind of unavoidable.

So what can I say ? when you Activate the sound is getting better ?


Thank you very much for pointing this out.
Peter

But ... I think Klaus was talking about xovering in the PC, which implies multichannels allright (output for bass, mid, high), but fed with two stereo channels ... 

Hi Dave,

Could you be a tad more clear please ?

As you might expect, that indeed would be the (my) ultimate goal and Engine#3 anticipates on that (no matter it can't even do 24 bits yet ).

What's holding me back (apart from the lack of time ) is the needed DACs en enough bits (say 32) in there for appropriate volume control.
Now, it could be that your XR-55 (etc., there are more) for an overall result are better than my current setup (DAC + amps), but as you can expect it's not a matter of "just trying". It would need :

1. Buying six channels of such chains (one XR-55 would be enough I think);
2. Creating the crossovers for my speakers (which would not be a one day job);
3. At least try it with appropriate (ASIO) playback software, or in the end integrate all in XXHighEnd.

I don't underestimate such a project because

a. The DACs should workout at least as well as my current DAC;
b. The amps should be as least as good for sonic results as my current amps;
c. Although such an "active" crossover is easier to create in theory, it's not said that the result for SQ is better from it.

Then another theory (from me) :
Where such a DAC/AMP would work similarly to DSD for digital principles (at least I think it is), this would come down to SACD quality - or in the very end to delta/sigma like working, an I don't think anyone thinks that's the best.
Whether the net overall result is better still, is another question.

But hey, leaving out the crossover thing as a project, would simplify things to buying an XR-55. Right ?
Peter

Hi,

That is because it wasn't implemented yet. If all goes as intended, after this weekend you will.
Same for MP3.

Peter

I have a hunch  ...

You as well as Johan use the DDDac, right ?
I'm not sure, but I recall similar problems a few months ago and I think that was (or those were) DDDac users too.

I think the driver (USB ? it's not in your sig) doesn't communicate all that well with the Dac.
Now, I'm not sure how to relate that to your "keeps on repeating" problem (which is SPDIF and another DAC) but it sure looks as the same problem -> Your DAC keeps on playing while the host (process) has gone ...

When this is Engine#1 or #2, during playback, kill XXHighEnd.exe (Taskmanager).
When this is Engine#3, during playback, kill XXEngine3.exe.

What happens ?

I did exactly as you said. ... All OK.


Your "click open" obviously is "click Load" I suppose ?

Also, exactly when do you click on the Playlist tab ? (it's not in your list but it should).

How many albums did you select ? And are that albums or tracks ?

Lastly, of course you are sure you run 0.9sb-2 and copied that over 0.9s-0 ?


Sorry to be a pain ... but so far I can't understand.

Leif, Johan,

Since Johan seems to emphasize on tracks loaded via the Library Area (and Leif in the end tells the same), could you please confirm that this is, or is not related to the problem ?

For instance :


I cannot see at all how the Library Area could be related to the problem, since the sequence would be :

1. Load tracks from Library Area;
2. Press Play;
3. XXHighEnd closes itself (Unattended Playback);
4. Bring forward XXHighEnd again;
5. Currently playing track is at the top;
6. Press Next.

There is no way the Library Area is involved here, so how do you think it is related ? (added to that I can't copy the behaviour).
Actually I think my presented 1,2,3,4,5,6 is not what you do ...

Also the explicit question : When you load tracks via the Library button, or drag tracks from Explorer, you don't have this problem ?

Peter

No Carlos, the opposite ...

The bass from vinyl, what I listened to the whole day anyway, was lacking of definition all over the place. The effect from that ?
No smoothness, but merely rumbling (no no, I don't mean rumble from the turntable).
When I made a remark about it, I was asked the question "do you want even more clinical bass  ?". And I did not understand that question ...

So since then I've been talking to a few genuin bass guitarists, and each of them came of with indeed the same remark : the sound of a bass guitar should not sound lineair and out of everything should not sound clinical. "That's why we love tube amps" they said.

My next question was how their basses get recorded, and the answer was "directly through the mixing console".
but also
"but we'd love it when all is recorded through a microphone in front of the loudspeaker" ...

So there you go ... there is no way to listen to how the players intended their playing.

Another things which surprised me is that they told me that there's no better dynamical instrument than a bass guitar. Similarly the remark about it being so difficult to outperform the sound of toms with a bass guitar, surprised me.

The whole subject is too complex to put down in writing here, but what it comes down to is that I bought a bass guitar, a good amp with it (on the asvise of one of these bass guitarists), and that I 100% can imagine what they mean by the mixing console vs. the microphone just because of what the amp can do to the sound (like overdrive and interaction with the speaker driver), but the other things (dynamics) ... no. The point is that only few people these days know what CD recorded (compressed) audio is capabale of during playback. Those guys just don't know that ... I (we) do ...

Le Snob