XXHighEnd

Hi Peter,

I am now using an Eximus CD player/DAC. It's USB input automatically detects and accepts signals between 16/44 and up to 24/96.

In your installation note, part 2, you state Rather important: Probably you are used to set your sound card to a certain amount of "buffer". E.g. for the Fireface that would be 48 (samples). Keep in mind that the below requires more of that buffer. E.g. with 96Khz/24 files (for me) that would be 96 (samples). Better higher this buffer at this stage, or *that* may give you crackles. Note that with 16 bits in either rate of upconverting (to e.g. 24 bits) my Fireface again allows for 48 samples. But start with higher anyway!

My PC-knowledge is limited, but as far as I understand, XXHighEnd is feeding the signal DIRECTLY to the USB com-port. So, there is no sound card buffer setting to change. Or what? Can you pls clarify this matter?


I have installed 09.U-0a and it plays music. The digital volume works. (I still use my active linestage for practial reasons BTW). I think it is a slight SQ improvement compared with XXHighEnd 0.9T-5. More fluid and "lighter" character, without loosing dynamic contrast or slam. (It reminds me about the positive SQ results of Q1 = -2, but without it's softening of the transients). I am using Q1 = 14, by the way.

However, I am not able to alter the XXHighEnd settings of "DAC is 16 bits   44.1 KHz". If altering to fex. 24/96 I get error code UNSUPPORTED FORMAT + FAIL (and no music). -I find this strange, considering that I have a 24/96 DAC. Whats happening? Can you pls clarify this matter?

Thanks in advance,

Hi

Just in case...
Did you try to set "32 bits (most often)" radio button on the DAC needs section

Yes I did.

True and No ! ... :rofl: About the Latency settings of your soundcard (http://www.phasure.com/index.php?topic=386.0). From earlier from today ... Last alinea.


The answer to this, I think, lies in here :


Or IOW, chances are fairly high that the drivers your USB DAC require do not list in the Audio Device's properties - Advanced tab (with the 96/24 option). Or ?
Btw, don't get confused ... the options in there do nothing for exclusive mode, it will show though what the Vista Exclusive Mode can deal with.
Disclaimer : I'm still learning myself in this area. :sorry:

Peter

This is consistent with my situation as well. I cannot play 16/44.1 songs when "DAC needs" is changed to 24 or 32bits. If I change to "DAC is 24bits 96KHz" and "DAC needs 32bits" then all I can play are 24/96 files. (Or if I change my M-Audio control panel setting to 88KHz, then I can play 24/88 files)

Please also read this post for the messages I'm getting:
http://www.phasure.com/index.php?topic=379.45

edit: I should also note that my M-Audio Device comes with a custom driver (does not use the Vista USB Audio driver) and therefore has it's own control panel. In that control panel I have to manually select 16 or 24 bits and 44.1 or 48 or 88.2 or 96KHz. And this is also where I select latency (the lowest which is 128). So my device is not "auto-switching".

FWIW The Terratec is consistent with this too.

Edward,



Can you recognize what I was suggesting : that these "custom" settings are not 1:1 with what Vista shows in Properties - Advanced ?
I mean, I can just as well be wrong here, because almost every audio device that's not USB comes with its custom drivers. And they show in Properties - Advanced (as possible settings). Otoh I have the hunch it's kind of generic. For example, my Fireface shows two options in there it will never be able to play (both at 24bit-3byte options, while the Fireface can only use 24bit-4byte). And mind you, I only know this because RME was so kind to tell me. You can't see it (3byte and 4-byte options show the same).

Thanks.

Hello again,

IN SEARCH OF HI-REZ - PART 2:
Continuing my efforts to play Hi-Rez, I have tested a 24/96 WAV file. No music. I tried all the various DAC settings in XX "settings". So whatever settings, I am not able to play WAV 24/96. The error code is "Can't play! - Wrong Stream Status (1)".

Then I asked VISTA to update my drivers, but they reply that my drivers was the last versions. See below.

So what can I do? -Is it possible to test out alternative USB drivers? Mayby it will work with USB drivers sourced from other companies?

---------------

I purchased the €5.000,- Eximus 24/96 USB CD/DAC in order to play hi-rez files. :veryhappy:
Now I realize it doesn't work. :cry:
Will there be more developements in this field, Peter St.?  :unsure:

After all there must be many users with hardware of same technical category as me.

Until it works for you. But wait a minute ...


This says the file isn't correctly organized. In the near future that will be attacked by me, but for now ... this just is so. IOW, try another (from another site !).

Peter

Who told you this? I would double check with your dealer.



I looked on the April Music website and no where does it imply that you can play hi-rez files. The CD player only plays CDs (not SACD). Right? The only thing they are touting about hi-rez is that their player/DAC upsamples everything to 24/96 or 192. That does not mean it can natively accept those formats. Dave (SeVeReD) has the Stello DA100 that uses the PCM2704 (USB input receiver chip) and I suspect yours is probably the same or similar. That is limited to 16/48.

Quote from edward:

All April Music D/A´s and Eximus CD10 can play 24/96 through the usb input.
This is Stello DA100, DA100Signature, DA220MK2 and Eximus CD10.
I am actually playing 24/96 with Foobar and a DA100S at this moment, but xxhighend does not work with 24/96.

I think the reason for your misunderstanding regarding this is that the usb receiverchip used in these units also contains a lowquality built in 16 bit dac.
This built in dac is not in use in any of these units.
The signal from the usb receiver is led directly to highend dac´s from AKM or  Burr Brown (CD10), and all the units can therefor play hi-rez.

Thank you for helping Terje.


Can you please try to indicate why this is ? it just should ... (but as you have seen things are rather confusing at this stage of time).
Also :

The picture from this post (http://www.phasure.com/index.php?topic=387.msg2612#msg2612) in this topic shows the normal USB Audio driver. Do you use a different one ?

Thanks,
Peter

I am afraid I can not tell you why, but I get the following error messages.
When trying to play wav24/96: Wrong stream status(1)
When trying to play flac24/96: Length error in flac file. (this error message also occures when placing flac files in playlist)

I use the same drivers as pedal. These drivers are automatically used when connecting Eximus/Stello products with usb.

As Norwegian importer of April Music´s products, I am really waiting for Xxhighend to be able to play hi-rez with our products.
I dont feel ready to recommend Xxhighend to our customers before it is a complete solution with the ability to play hi-rez formats.

Best Terje

Hi Terje,

Again thank you for your input. Also, a (late) warm welcome here ! :)

Then, the both errors you mention, really have the file format as the cause. Both coincidentally different (WAV vs. FLAC), but anyway. BUT : I must tell you that at this moment I did not check FLAC 96/24, so that could be my bad. But I will check later today.

As I said earlier, please try to get a WAV from another source, no matter it plays in other players. The format in the file you use *is* wrong, but I could tweak it (and so far I did not spend time on that).

But if the format appears not to be wrong at all afterwards, I'm very sorry in advance, ok ?
Peter

OK, I apologize if my information was incorrect. Indeed the datasheet indicates a maximum of 16/48, but I did not realize this to be only pertaining to the DAC. I did not mean for my comments to be "definitive". I was just offering a "possible" conclusion. I hope you are right and you all get 24/96 working (with XXHighEnd)!


Sorry, I don't mean to overstate the obvious, but can I assume you are using ASIO or KS (with foobar)? Because otherwise, I think, Windows is just downsampling back to 16/48 before it goes out to the DAC.

Terje is a seasoned high-end'er with very good system and listening experience. I hope he can contribute more here on the forum. Peter needs all the feedback he can get. XXHighEnd can be considered as a joint venture between him and all the native users. Remember we are still in "Beta-mode".

Also, what he told me, but forgot to tell you, was the following: His 24/96 file (downloaded free Mozart 2L.no sample, same as mine) plays perfectly in Foobar, but exactly the same track from the original CD (in 16/44 ofcourse) sounds better through XXHighEnd!  :yahoo:

In other words, XXHighEnd has ALLREADY reached 24/96 performance (relatively speaking) from playing 16/44. So thumbs up to you Peter!  :goodjob:

Keep up the good work Peter. I am sure you will solve the 24/96 related bugs in the near future. In the meantime we all are enjoying XX 16/44 quality second to none!

pedal

Haha, thank you pedal. Nice too hear !

Right now I'm working on some "analysis" function that can summ up the possibilities of the Sound Device. I think it's almost done. Putting it up is another matter because I was also in the middle of 44.1/16 -> 24/88.2 upsampling. It's not that much work anymore, but maybe not tonight.

:)

Ok, this is to get warmed up a bit ...

For the official reporting of Exclusive Mode what's possible, below you can see in the first attachment that HD-Audio is capable of ... well ... almost nothing. But *also* you can see that a 96/24 file must be possible to play when upsampled to 192000 with the DAC set to "Needs 24." Ok, XX can't upsample to 192000 yet. So dead end here.

The second attachment shows the FireFace800. It can do everything ...
Yea ? NO ! because it can't accept 24 bits. This is a "needs 32" device.

Then it's more complicated, because behind the Fireface is the TwinDAC+ and I can't see through the FireFace. So now I just have to know that this DAC is 96/18 max, and it needs 32 bits for it. Or anyway the Fireface arranges for that.

The third is USB Audio with the TwinDAC+ connected. Officially the TwinDAC+ can (USB connected) do what is says there. But, this is probably (already) because it's squeezed by the USB Audio driver.

Well, :wacko::wacko:

Hey Guys!!!!
What's going on here.
I have a Stello 100 USB-DAC (probably one of the first - non-signature model).  I've enjoyed the Stello quite a bit and with XXhighend is the best front-end I've had in my system (I'm coming from a Sony SCD-1 that I felt didn't need any mods, just a solid acrylic platform... excellent unit that wasn't matched until XXhighend .9D).  Sooo, do I have a chance at higher rez with this unit?  So far all my clicking around...243296.... have produced nothing but errors with 16/44.1 wavs... so I assumed I was SOL for higher rez or doubling/upsampling... I have to have 16/44.1 selected under XXHE settings tab to hear music... which sounds pretty good at the moment....(I've kinda gone back to Vol at 0 and using the xover volumes.... I haven't experimented enough with dig vol & xover volumes ratios to see if I can get a better sound.

This report makes SeVeReD a little sad :/

Start Audio Device Analysis Exclusive Mode support
040FFD38

    Supported : 2|16|44100|176400|4
Not supported : 2|24|44100|264600|6
Not supported : 2|32|44100|352800|8
    Supported : 2|16|48000|192000|4
Not supported : 2|24|48000|288000|6
Not supported : 2|32|48000|384000|8
Not supported : 2|16|88200|352800|4
Not supported : 2|24|88200|529200|6
Not supported : 2|32|88200|705600|8
Not supported : 2|16|96000|384000|4
Not supported : 2|24|96000|576000|6
Not supported : 2|32|96000|768000|8
Not supported : 2|16|176400|705600|4
Not supported : 2|24|176400|1058400|6
Not supported : 2|32|176400|1411200|8
Not supported : 2|16|192000|768000|4
Not supported : 2|24|192000|1152000|6
Not supported : 2|32|192000|1536000|8

 :drinks: :grin: :cry: :unsure: :)

Contact your dealer ?
:swoon:

This again is about what is to be expected from the USB DAC Audio device ... I vote for that this is just correct ...
Please keep in mind ... my TwinDAC+ reports exacly the same USB connected. It's in the specs as well. Still, SPDIF connected it's 96/18 and it really does that (to spec again).

Hi Terje,

On behalf of customers and users here, could you please re-check ...
Apart from (I think) only the specs of the CD10 mention explicitly 24 bit input being possible, it is my hunch that any device allowing for that would need a special USB driver for it. So, not the USB DAC Audio device.
Could that be true ? if yes, where can it be obtained ?

Also, by now can you please try the DAC Test facility from 0.9u-1 ?
If another player can do it, I be happy to try to get it to work. But as long as theories (and a little practice :)) say otherwise it's kind of difficult for me.

Thank you very much,
Peter

Hi Peter,

not sure where to place this info, but this thread looked as good as any.

Primarily a FYI posting, but might but helpful for others trying to decide what make/model soundcard to buy in they are in that situation.

Here's the results from the analysis log of the ESI Juli@ card

--------------------------------------------------------
Start Audio Device Analysis Exclusive Mode support
044AFD38

    Supported : 2|16|44100|176400|4
    Supported : 2|24|44100|264600|6
    Supported : 2|32|44100|352800|8
    Supported : 2|16|48000|192000|4
    Supported : 2|24|48000|288000|6
    Supported : 2|32|48000|384000|8
    Supported : 2|16|88200|352800|4
    Supported : 2|24|88200|529200|6
    Supported : 2|32|88200|705600|8
    Supported : 2|16|96000|384000|4
    Supported : 2|24|96000|576000|6
    Supported : 2|32|96000|768000|8
    Supported : 2|16|176400|705600|4
    Supported : 2|24|176400|1058400|6
    Supported : 2|32|176400|1411200|8
    Supported : 2|16|192000|768000|4
    Supported : 2|24|192000|1152000|6
    Supported : 2|32|192000|1536000|8
--------------------------------------------------------


Cheers,

Russ

But Russ ... You are not going to tell me that you can get the 24 bit possibilities actually to work ? or ??

Assumed not, this now leads me to the thought that this analysis thingy must dive deeper. Kind of right to the end and playing. If *that* works, then it would be actually true. One exception : crackles. But I can't believe it will come that far.

I will change the analysis accordingly. :)

Hi Peter ...

It all seems to be working as expected  8)

The control panel for the Juli@ card has indicator buttons that change green to indicate the sample rate being played. As the DAC test analysis is running, those buttons are initially lit at 44.1 then progress thru 48, 88.2, 96, 176.4 and finally 192.

Using XXHE's volume control seems to be doing its thing also. If the Vol is set at 0dB, the Sony processor I use indicates 16bit @ 44.1KHz on its display. When I drop the volume to -6dB, the Sony then reads 17bits @ 44.1KHz. At vol = -12dB the Sony reads 18bits @ 44.1KHz, and so on , until, at -48dB, the Sony reads 24bits @ 44.1KHz.

Playing back 24bit/96KHz WAV's has the Sony indicating 24bits @ 96KHz so that part is working as expected also.

Seems like the Juli@ is a versatile card :smile:

Cheers,

Russ

Peter,

This i do not understand. I thought the higher near 0db things where getting better.... Or does that not have anything to do with the bit thing?

:rofl::rofl:

I'm not sure what technique Russ uses to determine all is right, but I think I get the grasp of it.
His highly intelligent processor presumeably shows which bits are actually USED. So, with a 44./16 file playing (which Russ doesn't mention, but I hope he is doing that) and the volume at -0dB, indeed only 16 bits are used. There aren't anymore in the data AT THAT MOMENT.
When 6dB is attenuated, what happens is that the 16 available bits are are shifted one bit, and it goes into the available headroom of the more than 16 bit capable DAC. So, now 17 bits are being used. Note that actually this is not true, because the Most Significant Bit is out of use now. So it's still 16 bits.

Hey smart*ss downunder ! did you find this out yourself, did you just know it, or did it take a couple of days reading the Sony manual ?!? :prankster:
I must say, I'm really surprised ... Hats off !


And Russ, my "24 bit" question was about my suggestion that you could not use the "DAC needs 24 bits" setting (so, 32 only). Am I right ? or can the Jui@ take both 24 and 32 bits ?

Thank you !
Peter

LOL ... no reading of manuals down here mate. That's only needed when things get real desperate and as a last resort  :scratching: :scratching:

The Sony's display has a 'scroll' function that can be manually activated ( via one of its knobs ) that will show you the characteristic of the audio file being processed, amongst which is the bit rate and sample freq. I often use that feature to double check that things are working as intended when using different playback programs. I noticed a few days ago that when I had set XXHE's Vol at anything less that 0dB, that the bit rate figured increased accordingly until it got to 24bits.

And yes, I was testing the DAC analysis with a 16/44.1 WAV file. And yes again, it needs the 'DAC needs 32 bits' setting to work also. It's a no go if I try and set to 24 bits .... XXHE resets back to 32bits in that setting.

Cheers and beers,

Russ

Thanks Russ.


"Resets back" ?

ah .. I meant to indicate that when i tried to select 24bits, that XXHE rejected that attempt and forced the 32bits option. So perhaps 'reset' was an inappropriate word to use as it never got 'set' in the first place.

Hey  .. its late over here  :grin:

sleepy Russ

bump
Has Pedal or Terje come back in and posted Audio Device Analysis Exclusive Mode support Test:

I haven't contacted my dealer yet, but I have to wonder if Foobar is really sending HiRez data or is Win just down sampling for the dac to accept... going through the mixer?... I highly doubt this is XXHE fault.  I just think my Stello doesn't accept HiRez files through the usb port.... then again, I'd like to see Pedal or Terje do the Audio Device Analysis Exclusive Mode support test.

I'm about ready to get a big tax return and I'm thinking new DAC... but I need to really have the ducks lined up in a row.

Huraayy, my 48/16 USB DAC now can play 96/24 too !! All it needs is foolbar !

Dave, you're hoaxed ...


This is what Foobar does, no matter what I set the output bits to (0.9.4.4 here).
Also, resampling will take place, although I can't prove to which sample rate. On that matter, note that Foobar can work around the Volume Mixer of Vista, which is what I use here. So, that can't say anything today. What *does* say things, is that during playback (of that 94/24 file) I can "test" the sound device via Vista's "Test" (rightclick on the device), and the test sounds go right through the music. The other way around works too : first the test sounds, and right through that press Play in Foobar; then too both will sound.

At this moment I vote for the Stello (and what else) *not* being able to do 94/24. This is completely logic, as long as legacy USB can't deal with it AFAIK.
Terje, please grab a DAC of which you are sure it can do 48/16 max. Now take Foobar to testify that you shouldn't take Foobar to test this. Ok ?
You still can be right (and are certainly allowed to :)) but the means to prove it must be, say, acceptable.

Peter

Allow me ...

I don't think anybody will be against the possibility of playing a 96/24 file which plays as 96/16 for those having "DAC is" set to XX/16, or IOW for those who have a DAC that does allow for 96KHz but where the DAC supports 16 bits only.

This will cut the least significant 8 bits, of which I "state" that you won't hear the difference anyway, as long as you don't play very loud. Theoretically you will be able to hear it once you digitally attennuate LESS than 48dB (like 42, 36, ...).
Note that in this case there's actually no difference between attenuating digitally or analoguely, and it is just about putting the volume so loud that the data of the MISSING lower 8 bits becomes audible (as harmonic distortion). This will lead to the same audible distortion as cutting way the lower bits by means of (too much) digital attanuation as described elsewhere.
My reference to "digitally attennuate LESS than 48dB" is useful because with digital attennuation you can try out the level as set by analogue means above which you should not turn the analogue volume. So as long as you stay under the digitally checked -48dB level, you must be okay.

Is this a bit clear ?
:wacko:

Maybe it will be a bit clearer if you explain (a bit :) at least) why cutting of 8 bits (from 24 to 16) will be effecting distortion when you go  to less than 48 dB attenuation.
How do you get from 8 bit to 48dB.
And  by the way, 48dB  is a lot. At least in my system it is useless re soundlevel.

How much distortion do you get when one  goes to 0dB digital and use just analogue volume with a 16 bit Dac , with 96 hZ files that is?

I'll try (a bit :));

(don't try to understand ... I'm sure I can't myself when I read it back. :sorry:)

Per bit you loose 6dB. You loose 8 bits when from 24 is cut to 16, and 8 x 6 = 48. Wow.

Loosing 8 bits is similar to attenuating 48dB when you started off with 24 bits. So 16 are left at this level of attenuating.
Now think the other way around :

With no lower 8 bits there from the beginning, all is equal to before situation when you start at -48dB going upwards towards -0dB. One difference : no bits are added. Now, the louder the music becomes, the more chance you have to hear the missing bits. At -0dB you will hear them at full power. Mind you, with soft music that is anyway, but with loud most possibly too, because it will cause plain (and loud) distortion.

AGAIN think the other way around : You start with -0dB with a 16bit file but 24 bit playback means. When you attenuate 48dB still not any bit will be lost. Attenuate more, and you start loosing bits. Please read the post of Calibrator and his receiver showing the bits in use, and you'll understand the principle better.

All depends on how loud -48dB already is. Is it way loud already, then no problem for the playback of 24 bit files cut to 16. The cut bits are down under in the noise. If it is not loud at all, and you need -12dB for an acceptable level, you will have shifted 6 MISSING bits into the audible level. But mind you, these 6 missing bits leaves still two more (18) than a 16 bit file contains. This however, is not compareable, because in the 16 bit file all necessary data is "compressed" in the 16 bits, whereas in the 24 bit file all is spread (AND WHICH IS ORIGINAL) over 24 bits.

Again, all is dependend on the SPL your system produces.
In my case, where -24dB is loud enough to never want to go louder (not so far anyway), I can attenuate a 24 bit file to -114dB and I still hear very faint music but *not* distorted. at -120dB I can't hear it anymore. So, no "room" for distortion. -114dB means a cut of 19 bits i.e. 5 bits left.
With the 16 bit file (played at the 16 bit DAC !) I hear destortion at -48dB i.e. 10 bits left. Why is this different ? One reason is that -48dB is way much louder than -114dB so I have the opportunity to hear the distortion better. Another reason would be that in the 5 bits left of the 24 bit data, still the resolution is in there for the part left (which is diffrent from micro data / changes which resides in bits which have been shifted out long gone). I'm not even 100% sure about the latter, and I just try to find an explanation for what I hear myself.

I agree with Peter: In practical listening, we have no need for more than 16bit PLAYBACK performance, because teoretically 96dB S/N is more than enough to handle practically all excisting recordings.

On the other side, I think we can't have a high enough sampling frequency. Not because we can hear much above 20kHz (which is the limit for most red book DAC's) but because we can hear the sideeffect of the very steep lowpas filters at 22kHz. Pushing the sampling frequency up to 96k is certainly much better, soundwise.

I spoke with a renowned music producer who does all his recording and editing with the latest SOTA digital equipment (DSD/DXD systems at 32bit). He said that his classical recordings on SACDs has maximum obtainable dynamics [considering noisefloor of microphones etc], copied directly from his mastertapes without any compression. "Wow", you think, and start envying those with SACD players. BUT, then he continued and said: His CDs are only 3dB worse than his SACDs! And that these 3dB occurs on the SACD only for a fem milliseconds a handful of times during (let's say) a 5 minute song. So for 99,999% of the duration of the song, the dynamics of the CD equals the dynamics of the SACD. In other words 16bit is enough for PLAYBACK. (-Of course, he needs more than 16bits when mixing and editing. Also he recomends SACD for its multichannel capabilities etc, but that's another story).

He also confirmed that there is majore sound improvement when increasing the samplerate from 44k til 96k, but above 96k the SQ improvement is less noticable.

It should also be noted than very few, recordings are made so purist and dynamic as described above.

In reply 10 I said the following: All April Music D/A´s and Eximus CD10 can play 24/96 through the usb input.
Unforunately this is not correct, and I apologize for that.
This incorrect information is due to a communication error in an earlier discussion with April Music about this topic.

As Edward mentioned earlier, when playing 24/96 with foobar the signal are downsampeled before it is sent out the usb port.


-There is no USB chip at this time which supports 96kHz sampling.
-The USB chip can not receive more than 48kBps (this is USB specification).
-Sound card of PC/notebook does not export more than 48kBps through USB
 or SPDIF whatsoever.

All April Music DAC´s support up to 24/192 format, the output from either
transport or PC should be upsampled before reaching the DAC.
Once again, unfortunately, no sound card can shoot out 24/96 (they are
advertising but the actual output is limited) unless they are upsampled.
April Music use upsampling and specially designed jitter reduction to lift the
sonic quality up to a new level.


Best Terje

On behalf of everybody, thank you Terje.

All it needs is someone to make a higher grade USB driver. Maybe someone can call (IIRC) usb-audio.com ...

That's not true. As I mentioned before, my DAC uses the TUSB3200c which can accept 24/96 with my M-Audio driver. Also, Benchmark, Empirical Audio, and Wavelength all use the TAS1020b. They all use a "driverless" configuration which means they do not supply a custom driver - you simply use the native Vista USB Audio driver (just like the Eximus and Stello use). This chip accepts 24/96. The difference is, they have custom programmed the firmware onboard that chip.

As for the USB spec - I'm not sure about that. USB 1.1 operates at 12Mbps - so I don't know about the numbers you came up with. 16/44.1 audio is ~1.4 Mbps and 24/96 is ~4.6Mbps.

Ok. Everything is again as unclear as can be. To me it is.

Possibly you are telling that you are not pushing "audio" as such through USB, but just data ? Or rephrased, do you play through an USB Audio DAC device, or just through an USB device ?

:dntknw:

I don't understand what you mean here.

And who are you asking?

You. :)

There are quite some contradictions in this thread now.
If USB specs tell that 48/16 is max, I don't see how a reprogrammed receiver chip can do anything about that. UNLESS you're not talking audio, but "data", which just is possibe too (at playing audio, might you know it or not ... my question is about that anyway).

:sleeping:

Who said anything about USB spec being limited to 48/16? That is false.

Terje also said:
and...

He said 48 kilobytes. (Which translates to 384 kilobits/sec) That would barely cover MP3s!!
Just look at the properties of a 16/44 song or 24/96. You'll see that they are 1411kbps and 4608kbps.

Where the heck does he come up with the idea that no card can shoot out 24/96??

As I've mentioned, I've been interested in picking up a new dac.  I have been looking at Empirical Audio offerings.  I sent him the email below.

"
Audio Asylum Email from Dave Robinson:

Hi
I have a question about your Spoiler usb-dac (or any of your other dacs).

I've only been listening to my 16/44.1 wav files.  But I'm curious about 24
or 32/96 files and how that is handled by vista > usb out > usb-dac.

You may have read that I'm enamored with XXHighend, and he has a built in test
that checks what rates are supported by the dac and which aren't.  At the
moment my Stello DA100 usb-dac only supports the following USING USB...(it
supports higher using coax input).

Start Audio Device Analysis Exclusive Mode support
040FFD38

   Supported : 2|16|44100|176400|4
Not supported : 2|24|44100|264600|6
Not supported : 2|32|44100|352800|8
   Supported : 2|16|48000|192000|4
Not supported : 2|24|48000|288000|6
Not supported : 2|32|48000|384000|8
Not supported : 2|16|88200|352800|4
Not supported : 2|24|88200|529200|6
Not supported : 2|32|88200|705600|8
Not supported : 2|16|96000|384000|4
Not supported : 2|24|96000|576000|6
Not supported : 2|32|96000|768000|8
Not supported : 2|16|176400|705600|4
Not supported : 2|24|176400|1058400|6
Not supported : 2|32|176400|1411200|8
Not supported : 2|16|192000|768000|4
Not supported : 2|24|192000|1152000|6
Not supported : 2|32|192000|1536000|8

What would your Spoiler Dac support through your USB input?
Does this require special drivers or do you use the standard MS drivers?
Are the drivers built into the dac? hehe am I making any sense?  At the
moment all my rips/collection is 16/44.1, but I am curious about the future
(Linn stuff and others).
"
Empirical Audio sent back


"
24/96 files are handled correctly by Vista providing you set 96kHz in the audio options and have a 96kHz native file or
one that is upsampled by Foobar and SRC to 96kHz.  All of my USB converters do only 24-bit.  Even native CD's are
changed to 24/44.1, making them bit-perfect with Vista.

24/96 using foobar 0.8.3 and SRC is definitely superior to 44.1 when using my USB converters.

Steve N.
http://www.empiricalaudio.com/
"

Somehow I don't think this is what I want to hear.  Dang... all I want to know is if I have a 24/96 file (or other hirez file) that it won't be resampled upsampled nothing-sampled... just played back... I mean if the dac can accept that bit/sample rate by a usb port then it doesn't matter if it's sent by foobar or XXHE... if they insist foobar does it I have a feeling it's all about the change/resample.   have to go to bed... but tomorrow, should I look into how to change "set 96kHz in the audio options" and run the test again on XXHE?(no no, i know it won't work with the stello)  I was hoping to pick up his Spoiler...but maybe I'll have to try his loan program first and see how it handles the XXHE test.

No Dave, what Steve is referring to only applies to "shared mode". Changing your "shared mode" options will have no affect on Peter's "exclusive mode" test (nor XXHighEnd).

Ok. It was late and I read Terje's post too fast. I read 48 kBps for 48 Hz, I read USB ship for USB, didn't understand the last half of hist post (soundcard output), and I guess I wanted to read what my expectations were. WERE.
Also, it is kind of confusing for me with a TwinDAC+ that does 96/18 over SPDIF and 44.1/16 over USB.

Obviously the speed of USB (even 1.1) is fast enough to do it, that's why I asked for "data", or maybe I should say asynchronous.
Also, I thought I read somewhere that indeed usbaudio.sys just doesn't allow for more. Well, this *is* all over the place, but for other reasons than natively not being able to.

I read a bit this morning, and if one thing is clear, it's that more people do not - than do understand these matters.
Also, by now it is fair to say that people must be very careful what they actually listen to, while Foobar just proves you get fooled without notice. This is not so difficult to see : when my DAC does 18bits at most, how can Foobar just play the 24 bits file ? How do people test it ? (and I used the 16 bit connection).
If your DAC actually allows for 24 bits, you *still* won't be knowing what e.g. Foobar makes of it ! Foobar, Vista, USB receiver chips ...

A response like that of Steve N. above says it all. He is asked a question by someone who knows what the culprits can be, is fed with the right stuff to test(ify), and the answer ... well ... shows that this DAC representative doesn't know anything of what is going on. Not to be harsh to someone like him, but the message is : you won't be knowing what you're listening to when you have to deal with these people who *do* make DACs but *do not* know about them. Well, to be honest, about the impact within the systems (OS) they are applied.

Mind you, it was less than one year ago that people like Gordon R. just couldn't tell himself what USB actually allowed and it may have been the Elias G. intervention that made others believe it was actually possible (96/24). Never mind if I twist some names or sequences, I only want to say that the TAS1020b wasn't doing the job either, without mutual exchange of knowledge on how to deal with the buffers properly. If I'd apply that chip but forget about the dedicated buffering I'd have 48/16 only.

Now what ?

I can tell you, I wouldn't care less. That is, about 96/24 files which -so far- hardly is provided in my music anyway. More importantly though, I wonder whether I'd really hear the difference. Pedal's post above ( http://www.phasure.com/index.php?topic=387.msg2746#msg2746 ) says some things maybe too, or anyway I recognize all of it. One thing though I wouldn't agree upon much :


... which is related to the technical impact of 24 bits vs. 16 and the derived functionalities from that. I refer to digital volume now which for me is just "under spec" if I had 16 bits only (and which is related to gain vs. sensitivity of speakers) and which is on par with 18 bits. But that's me and my system, and people would really need 24 bits to do it right when having more gain and less sensitivity.

On another note, aren't a few of those 24 bit supporting DACs non-oversampling ? If so, how would this happen while no ladder (or similar) DACs exist with 24 bits ? :dntknw:

Terje, it's wrong that soundcards can't deliver more than 48Khz on S/PDif. My XFi Elite Pro can deliver 96Khz out (and does) and so does a lot more cards that I know of. Of course, cheap on-board solutions have very weak software solutions often not giving the user the possibility to change the frequency of the bitstream manually and to above 48Khz.

More expensive soundcard solutions have this option.

As an IT Technician, my point of view is that USB should not be used for audio at all...with it's inherent flaws, noise etc. Many implemented USB solutions also have a transmission roof of 48Khz. The speed of USB is not really suited to any more...

If you're wanting to use e.g. 96Khz, use Firewire or S/PDif IMO.

Sorry for not writing this properly.
It should have been 48khz not 48kBps. This is usb receiver chip spec.

Dear Peter,

See DAC test below. Not much to brag about here...
My Eximus DAC is manufactured by APRIL, same company as maker of Stello DAC. Probably they all perform equally regarding digital input capabilities.

What does this means SQ-wise when using XXHighEnd? -Are we missing any sound quality [WHEN PLAYING REGULAR WAV FILES RIPPED FROM CDs - which is probably 99% what all of us are doing till now] because of this? Please advice. Several XX users have Eximus/Stello DAC's, so I believe this question is important.

Hi pedal,

(btw, is there any importance in the picture underneath your text ? I see a read cross only ...)

The answer is not all that simple (for understanding) I'm afraid;

Let me first say that if you do not want to use the Digital Volume from XX for your own reasons, for 44.1/16 playback it doesn't matter at all.
The point is in the if though ...

I determined that applying the digital volume is better than the analogue volume. Never mind why for now, but I reasoned it out en so far it looks ok. Additionally this - by accident - allows you to get rid of the preamp. And, as this is a major smooth-operator :) holding back transients, creating noise and all, it slowly becomes a matter of "needing" the Digital Volume to just have a better sound quality.

Btw, that a preamp holds back and therefore better is to be avoided is notthing new, but -so far- we couldn't really see how to do it right with the ditgital volume and really too few bits available for it. Now, what is applied in consistency in XX is that with 16bits input and 16 bits output all is - or just is not sufficiently legit.
Btw, keep in mind that with 16 bits in (say, the CD) and 16 bits out (your DAC as it sadly seems) everybody on this planet will say STOP !
And when nothing "special" is applied at digitally attenuating, they are very much right ...

And mind you, all who apply official digital attenuating (like in the pro world (which includes RME and the like) will do so by means of internal 40 bit or more float processing, and XX does nothing of the kind ...
With this I only want to say that we're way off the usual tracks here, and my own theories including the practice as of now, tell me it just can be done. Ok, just, or just not;

As I said before, the just / just not is highly related to the gain you apply and the sensitivity of the loudspeakers;
At first, when I couldn't exploit my own 18 bits of the TwinDAC+ (XX could not do that yet) I determined that as long as I kept above 48dB attenuation (so, towards 42, 36) all was okay. But mind you, this was absolute listening without the possibility to compare the 16 bits DAC (useage) to the 18 bit DAC. I thought I could hear anomalies at 42dB, but I took them for granted NET. Together with that I would never play that soft, and it's merely -30dB or -24dB. At -30dB too, I again I thought I couldd hear the anomalies, and again I took them for granted, and again NET.

Please keep in mind that there is a "balance" between more attenuation and the SQ getting worse on one side, and the less attenuation and less distortion (that's what it plainly comes to), or IOW the louder you play the better you can hear distortion obviously, BUT the distortion by itself gets less. At 0dB there's no distortion left on these matters ...

With the repeated "NET" I refer to the overall improvement, and *if* I can hear distortion somewhere a. I'm not even sure it is "that" distortion I hear and b. because of a. and the overall better result, I don't care.

Mind you, this is all with the 16 bit DAC.
With the 18 bit DAC though, the distortion (if audible at all at the given level) officially comes forward 12dB "later". Thus, when with the 16bit DAC audible distortion is there at -42dB, with 18 bits the very same distortion would be there at -56dB. Now here the mentioned balance comes in again, because when you normally play at -30dB, there is no way there's audible sound at all at -56dB. Try it (not with your ear in the speaker, but at normal listening distance).
To make a long story short, I am sure that I hear the difference between the 16bit (used) DAC and the 18 bit DAC both at -30dB. Also, when you A-B this, it becomes more apparent what it actually is you hear with the 16 bit DAC and which you took for granted. It's a kind of roughness that I can *not* really explain from missing bits as such, but of which I know by experience that they are needed to fill out "holes" at certain frequency levels (or better : from certain instruments). This is similar to how I eliminated "roughness" by just adding gain to the highest frequencies in my loudspeaker filter (which lineairly goes up from 5K to 20K by 16dB !).

With 18 bits I'm satisfied at -30dB, but maybe with 20 bits I'm more satisfied ...
Suppose that 18 bits would be the "standard" for legitimally play back at -30dB, then from that follows (it's just math) that with 16 bits all is as legit at -18dB.
When you are not able to play at -18dB (because it's too loud) the relatively simple solution is to adjust the gain of your amp. NOT by means of attenuating the existing gain !

The latter comes down to using the digital attenuation in the first place. So when you just don't use that, there's really no problem with 16 bits files that a more than 16 bit DAC can solve. Not that I know of.
Doubling or Upsampling to 88.2 KHz was created for better SQ (mainly Doubling !) but if it works out like that ... YMMV. For me it does not, because it needs a larger buffer in the soundcard (from 48 to 96 samples), and -again- NET it is not better because of that.

Please keep in mind, all of the above is related to 44.1/16 files only.
Once it would be a standard to have 96/24 etc. files, the "problems" are obvious, because you just won't be able to play them.

I'm not sure if you'd need any advise on bringing back your DAC to April Music, but if you spent that amount of money with the idea it could do 96/24 ... well ... The proof that you could be mislead is sufficiently available in this topic. :yes:. Along with that, the person involved not really to blame for it as well.
For the future : when the specs of a DAC do not specifically mention input samplerates and bits, better assume 48/16 as max. For that matter the CD10 does mention 96/24 for input explicitly. Whether *this* then is true (over USB !) I don't know.

I hope this is something for an answer pedal.
Peter

Hi Pedal,
I think whether we are missing any SQ when playing back 16/44.1 wav files is up in the air.  If I remember correctly, Peter doesn't care for upsampling/doubling of 16/44.1 files ... digital volume,,,???, I think he feels you don't miss anything if not used aggressively, (mines set at 0 though; I use my analog vol). ***edit - see Peter's words above***
I'm guessing the usb on our units can only take in 16/44.1 & 16/48.  Other people talk about how they like the upsampling/doubling, there units can take that signal in, ours can't...idk.
But,
Coax input should accept more rates for April products?  I think that's what I understand?
So...
What about using something like Empirical Audio's Offramp I2S (doesn't the eximus have I2S input?; Empirical adds this input to benchmark,,, I wonder if he could do it for the Stello....), or the Offramp Turbo 2 (for spdif out) http://www.empiricalaudio.com/ These units take in usb and output I2S or spdif respectively.
This from his site too, "All will pass 16/44.1kHz, 24/96kHz, MP3 and iTunes unmodified files"... I wonder if it will pass more than that now?(site could be old?)
or
UD-10 USB Audio Converter http://www.trendsaudio.com/EN/Product/USB_Audio_desc.htm
Perhaps if we use the coax inputs, our dacs will do more.  Maybe just need to find a usb to coax (I2S?) converter that passes hirez signals...untouched hehe
I'm not so interested in upsampling/doubling as I am playing back hirez files when they become available.

This is what is on the product page of my Stello usb dac

"As a professional D/A converter, the DA100 can handle any input data up to 96 kHz/ 24 Bit as well as the normal CD standard 44.1 kHz/ 16 Bit." ... "True 24Bit Delta-Sigma DAC"  Would have been nice if they had mentioned that the usb input was incapable of that....

Can I understand something:

If you have a 24bit/96 wav file, but your dac doesn't support 24... that you can set XXHE to 32 bit dac and it plays the file... yea?

also

I went to the Linn site and they have a lot of flac files at 24/88 would a dac that does 24/96 play these? no, right?  I mean if it did the files are being changed and not played back at the native rate...

Here's my goal... I wouldn't mind trying upsampling doubling twisting bits & pieces with XXHE ... but first and formost I want it and the dac to play back files perfectly in their native format... what ever XXHE (transport) sends to the dac I want the dac to play that without manipulation...just trying to find an audiophillic well made dac that  accepts ... well..how about these for starters (or the most important ones):

Start Audio Device Analysis Exclusive Mode support
040FFD38

   Supported : 2|16|44100|176400|4
Not supported : 2|24|44100|264600|6
Not supported : 2|32|44100|352800|8
   Supported : 2|16|48000|192000|4
Not supported : 2|24|48000|288000|6
Not supported : 2|32|48000|384000|8
Not supported : 2|16|88200|352800|4
Not supported : 2|24|88200|529200|6
Not supported : 2|32|88200|705600|8
Not supported : 2|16|96000|384000|4
Not supported : 2|24|96000|576000|6
Not supported : 2|32|96000|768000|8
Not supported : 2|16|176400|705600|4
Not supported : 2|24|176400|1058400|6
Not supported : 2|32|176400|1411200|8
Not supported : 2|16|192000|768000|4
Not supported : 2|24|192000|1152000|6
Not supported : 2|32|192000|1536000|8

Seems to me that within the immediate futures we're at least going to have some decent choices of hirez files...


Should I just get a Fireface (they should come out with an I2S output ???) I have one firewire into my lapto for the HDDs.  If I get a card with another firewire input ... would these two things get along and not collide?

Hi Dave,

Maybe I don't understand all of your post, but


No. You can just leave it at 44.1/16 and it plays the file (and you won't hear the difference). Mind you, the DAC needs 96 KHz for this. In your case you won't be able to play the file at all (why ? because XX doesn't downsample from 96 KHz to 44.1 or 48 Khz -> it could, but it doesn't seem the right thing to do).

No, just an error.

Question: IF I had a DAC who accepts more than 16bit, then, would the use of digital volume in XX introduce a SQ improvement? Or to ask in another way, is there any “secret” trick hidden in the XXs volume feature which betters the SQ?

-Or is the SQ improvement due to the omission of the preamp (shorter signal path etc)?

Please specify.

-------------------

Apart from this, I must state that the Eximus is build to a very high standard, and it is the best sounding DAC I have heard. The SQ of a DAC is based on several factors, and although Eximus doesn’t accept hi-rez through it’s USB input, it excels in other areas. For my ears, it sets a new standard for neutrality and resolution. It will remain in my system for a long time to come. And who knows, maybe there will be a future upgrade choice for it’s digital input?

BTW: Eximus offers 4x upsampling internally, before the D/A converting takes place. It’s a worthwhile SQ improvement on all quality recordings.

----------------

After a short initial listening test today, I feel that vU2 is another improvement compared to T-5. U2 is simply more transparent. I hear more recorded depth and more acoustic space surrounding each instrument. The bass is tighter. I can play several dB louder (always a good sign) without listening stress.

This is when playing through my external preamp as usual (XX volume = 0). My system needs between 20 and 30dB attenuation, depending on song material. (Some CD’s are recorded louder than others). So, no way to relay solely on the XX volume due to Eximus’ 16bit restriction. But I could eventually do -12dB digitally with XX and the rest -8 to -18dB on my active preamp. Are there any theoretical reasons (SQ) for me to try this out?

The latter for sure, but without that still.
Explanation is (too) difficult though, and I'd rather awaiten my own approval which -this time- takes more than a few weeks of listening.
It's merely a reasoning why things are wrong otherwise (using analogue volume) than any tricks being applied, although the means the digital volume is applied is all the way legit in the first place (you could call that a trick as such).


Yes, these theoretical reasons definitely exist, but I'd rather that you (or anyone) first testify yourself that the results are as I intend (i.e. "WOW"), than creating placebo's all over that won't help anyone including me (or the other way around :)).
Please keep in mind, this is a "way off track" theory, that made me start doing it anyway, and since it worked out for me (placebo !!) I so far think I'm right.
As you know, audio is  difficult, and the last thing I'd want is everybody just wanting to hear the same as e.g. I do. There's too much copying on the Internet already, and I want not only myself to stay far from that, but "you" just the same. No way to learn otherwise !
So far, all bits and pieces brought by some of you, are in 100% line of my own perceivement, and they are all but one on the positive edge. The exception, so far, is the lower bass end (as was brought by SeVeReD), and exactly that I recognize too.

I know, a great deal of these things you want to know because you really can't try it currently, but at this moment I really don't want to be all mighty while things are so much different that I keep on being in doubt myself what's wrong and what's right. You could say that *I* need the confidence from you all to be 100% sure. Only at that stage I will, say, "order" you.
I hope you can understand this (as well as the good cause).

Peter