09.U-0a setup with external USB DAC

[Calibrator]

Thanks Russ.

XXHE resets back to 32bits in that setting.

"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 

sleepy Russ

[SeVeReD]

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

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.

[PeterSt]

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

Dave, you're hoaxed ...

Then, that "a" player allows for playback of, say, 96/24 files, does not tell that it utilizes the 24 bits. Mind you, you wouldn't be able to discern if you don't know what to look for. When the lower 8 bits (LSB's) are cut during play back, you must play way loud to get that into the picture. And the remainder 16 MSB's ... same resolution.

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

[PeterSt]

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 ?

[Leo]

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?

[PeterSt]

I'll try (a bit );

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

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.

[pedal]

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.

[Terje]

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

[PeterSt]

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 ...

[edward]

-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 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.

[PeterSt]

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 ?

[edward]

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 ?

I don't understand what you mean here.

And who are you asking?

[PeterSt]

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).

[edward]

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).

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

Terje also said:

-Sound card of PC/notebook does not export more than 48kBps through USB
 or SPDIF whatsoever.

and...

no sound card can shoot out 24/96

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??

[SeVeReD]

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.