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24.0.202.30
In Reply to: RE: Or get the clock out of the PC posted by Jim F. on January 12, 2009 at 08:05:04
> I assume, BTW, that part of the [reason for the] popularity
> [and sound quality] of USB DACs is that they also allow the
> word clock to be local to the DAC, while leaving the USB interface
> to buffer the data from the computer asynchronously.
And so, of course, do network media players like the Slim Devices
Transporter, whether hard-wired with Ethernet cable or
operating over WiFi (but still only going up to 24/96, in the
case of that particular device).
You are still, of course, at the mercy of the goodness
(or otherwise) of the clock in any of these units. A pro A/D
unit **has** to have a decent clock, because the sound quality
of an A/D converter (and the reputation of the whole
company, as judged by picky studio engineers) is **directly**
dependent on the quality of the clock.
However, in all of these cases (where the clock is not at
the mercy of the computer's electrical environment) I would
expect the sound quality (barring any gross anomalies,
dropouts, etc.) to be independent of the player software
running on the PC.
Follow Ups:
Hi Jim,
YOu have a very Textbook argument here. I just don' think it bears out in reality.
It sounds pretty good, and while I have not spent any time
comparing playback software (I use Foobar -- either
0.9.5 or 0.8.3, with their respective ASIO DLLs),
I have no reason to believe that the "brand" of player
software could have any possible effect on this arrangement
(unless, of course, I were getting glitches, which I'm
not. My music is on an external disk drive in a
Firewire enclosure, and I keep it defragmented. It's
also a 1TB drive, with a pretty big buffer -- what
are they, 16 MB these days? And Foobar seems to do
a perfect job of decoding WavPack lossless on the fly.)
However, in all of these cases (where the clock is not at
the mercy of the computer's electrical environment) I would
expect the sound quality (barring any gross anomalies,
dropouts, etc.) to be independent of the player software
running on the PC.
> [W]hat is interesting is that Alan Kafton is one of the big
> proponents of the software being crucial. Yet he has one of
> those pro Firewire dacs with the great clocks (Rme Fireface 800)
> out of the computer, and still hears differences with players.
To wit:
> Posted by alan m. kafton (D) on December 5, 2008 at 11:02:10
>
> "It's the math"
>
> In speaking with various developers, it appears
> (from those discussions) that the algorithms used in
> software change the sound....there is no other plausible explanation.
>
> Vincent and I use professional-level, studio-quality playback
> software. We can readily hear the qualitative differences. My guess
> is that Foobar and others simply are not resolving enough.
> Having heard iTunes and Windows Media Player in comparison,
> well....there's no comparison. iTunes and WMP are left in the dust.
( http://db.audioasylum.com/cgi/m.mpl?forum=pcaudio&n=40814 )
I don't want to start a flame war here, and will respectfully
bow out of the argument at this point.
But I will say that if the job is to get data off the hard
drive and into the sound card without changing it, then there
is no "algorithmic" or "mathematical" possibility for Foobar
to be "less resolving" than, say, Wavelab (provided, as I
say, that they're not changing the data -- if there's some
sort of jiggery-pokery going on behind the scenes that
we're not being told about, a la the Windows KMixer,
then all bets are off).
**If**, however, you're doing any DSP (such as upsampling)
during playback by using a plug-in, then of course there **can**
be mathematical differences involved. (E.g., Foobar's
version of "Secret Rabbit Code" is still using an older
version of libsamplerate -- because Mr. de Castro Lopo
has not had the time or the inclination to update the Foobar
DLL, while cPlay uses libsamplerate 0.1.3, which is measurably,
and some say audibly, superior.)
Apart from introducing noise on the clock (jitter), then,
to paraphrase Mr. Kafton, there is "no plausible explanation"
for why two players, **if they are presenting identical
data to the sound card**, should sound different.
YMMV, of course.
> So what players have you compared??
Well, time is limited for playing with tweaks,
and I'd have to have convinced myself (as I did before going
to all that trouble to clock the system with the Apogee
Rosetta) that there'd be a worth-while result before
mustering the effort. (I have Wavelab, certainly, and
use it during the course of my off-line processing of CDs,
but the results of that processing are stored as lossless
WavPack 24/192 files, which can't be played in Wavelab.
I use Foobar, with no DSP other than the WavPack
decoding. And yes, FWIW, I use -- for old-time's sake,
I guess-- Foobar 0.8.3 with one of Otachan's ASIO DLLs.)
I would, however, suggest you try slaving your sound card to a good
clock, if you can muster up the equipment.
I see, BTW, that Mr. Kafton is also a fan of accurate clocks. ;->
See, e.g.,
http://db.audioasylum.com/cgi/m.mpl?forum=pcaudio&n=36514
http://db.audioasylum.com/cgi/m.mpl?forum=pcaudio&n=29092
Hey Jim,
I am not into flame wars myself. I don't see why we can't discuss this without getting into one, but if you think that is inevitable, well then its not worth our time is it.
I really wish you would give it a try. If you are going to talk theories, shouldn't you do a BIT of listening if only to confirm what you ALREADY know?? Wavelab and foobar both support 24/192 .wavs so just unpack your wavepacks and test a few songs. I am not making this up even though in theory you are right, players should sound the same.
When I did my listening tests I was doing strait 16/44 comparisons. My Hush is too slow to upsample at any kind of quality. So no dsps or upsampling were involved. Just the same song on different players. It was pretty clear that there were differences. An even easier test is to change the asio out you are using, if you can find one for 08 :) And maybe that is what I am hearing...just different asio drivers for diferent players. Could be.
On my part, I will see if I can do some DBTs of the different players and see if I can really hear what I think I hear. I think I am hearing things as they are because who wants Wavelab to sound better?? But I can at least look into it.
Yeah, I saw that ALan uses the black lion. I am thinking about using that with my Lynx if I get the dough. Looks easy enough to do. I suppose we should ask him if he can detect difference with the BL clock.
Recently I started using an external dac. I haven't listened to different players on it yet but I can let you know if that makes a difference.
Here is MAYBE one link that might explain perhaps some of what is going on, but I think you are right about jitter, and while this link doesn't talk about audio jitter, it might explain what could be happening.
> Yeah, I saw that ALan uses the black lion. I am thinking
> about using that with my Lynx if I get the dough. Looks easy
> enough to do. I suppose we should ask him if he can detect
> difference with the BL clock.
To get the full benefit of an external clock like the
Black Lion (assuming you're using an external DAC
and not just the analog output of the Lynx) don't forget that you'd
also need to send the Black Lion's word-clock signal to the DAC. In other
words, **both** the DAC and the Lynx (the latter being the
"transport" in this case) need to be sync'ed to the same clock, and
so your DAC in this case would also need a word-clock input.
If you just send the clock to the Lynx, while that's a
start (it does get the clock out of the computer), the external
DAC would then have to be **recovering** a clock from
the S/PDIF signal generated by the Lynx (and again, coming
out of the computer), with all the downside **that**
entails. You might be better off in that case just getting
a "clock cleaner" (e.g., something like an Apogee Big Ben)
to stick in front of the external DAC.
The use of a clock external to the DAC actually opens up another
can of worms, though. Some people insist that the best
solution is to have the master clock **inside** the DAC (i.e., be
as close to the DAC as possible) for best results
(assuming the clock in the DAC is a good enough one).
So again, we'd be talking about a good DAC with a word-clock
output (which means we're most likely talking about a pro DAC,
unless you're considering springing for a dCS or an Esoteric). OTOH,
Kafton does say that the Black Lion external clock was
an improvement over the master clock inside his RME Fireface 800
(which does, notice, also have a word-clock **input** that
can accept the Black Lion's sync signal. Generally,
any DAC with a word-clock output also has a word-clock
input. ;-> ).
BTW, keep in mind that even if you retain your Lynx as
the sound card (which I think is a good idea -- I think
an internal PCI sound card is just more hassle-free than
an external "sound card" using Firewire or USB. But
then I was burned by a bad experience with an M-Audio
Firewire device a few years ago. The real pro stuff may work
just fine.), you can **still** use some models of Firewire devices like
an RME Fireface 400 or 800 as a stand-alone DAC. Some such
models (the RMEs among them) can be configured by a software
"front panel" while the Firewire port is connected to the
computer, and then the chosen configuration can be saved
and the Firewire **disconnected**. Then the DAC would be
getting its audio via S/PDIF (or ADAT), just like an
"ordinary" DAC. Think of it has having a box with
the front-panel buttons on the computer. It would be a **bit**
of a nuisance, but if you don't change the settings a
lot (and provided they don't "decide" to change on
their own!). . .
> I don't see why we can't discuss this without getting
> into [a flame war], but if you think that is inevitable. . .
I didn't mean that it's **inevitable**, I just meant that I'm
not going to be insistent about a position if it's going
to step on anybody's toes. Particularly Alan Kafton's,
who is after all a dealer, and who apparently hangs around with
Vincent R. Sanders (of VRS Audio Solutions).
People can be touchy around here, after all! ;->
> I really wish you would give it a try. If you are going
> to talk theories, shouldn't you do a BIT of listening if
> only to confirm what you ALREADY know??
I guess I'm just too lazy to knock myself out over this
one. On the other hand, if somebody around here (or on
Hydrogen Audio, or wherever) ever posts an article that
says something like "Ah HA! So it turns out after all that every
version of Foobar prior to 1.1.0 [or whatever] has been
subtly altering the data by introducing a rounding error
when converting its input to the 32-bit internal DSP data
bus and then back again for the sound card! No **wonder**
they don't sound the same!", then you can believe I'll
upgrade my player software in a flash! ;-> Somebody else
just needs to do the work (and provide a plausible
explanation), that's all. :-0
> Who wants Wavelab to sound better. . .?
No, I'm certainly not going to use Wavelab as my day-to-day
player, even if it does sound better.
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