Asynchronicity: A USB Audio Primer

well done, Chris.

clay

Thanks for the excellent article Chris. You are doing a great job.

Chris, thanks for the audio primer.

I have read that USB is capable of 24/192 if the manufacturer develops a special driver. Since the Four Musketeers of superior USB capable DACs (Ayre, Wavelength, dCS, and Empirical Audio) are capable designers and programmers, where are the 24/192 DACs from each?

I also remember reading than ethernet is also a very excellent interface for computer audio but where are the ethernet DACs and I don't mean wireless?

Clearly every personal computer built generally has USB and Ethernet connectors compared to the relatively few that have toslink, coaxial digital, or firewire. So is the problem that the DAC manufacturers have not fully embraced computer audio and are keeping one foot in the legacy door of CD transports and other audio products?

A problem with audio on some USB-ports. Well done with the write up, Chris.
You have to ascertain which of your USB-ports is suited for audio, as not all are.

The use of USB Audio Devices on the Left-Hand USB Port Is Not Recommended. This applies to ALL MacBook Pro Models (Core Duo and Core 2 Duo).

• The 15" MacBook Pro models have 1 USB port on the left side, and one USB port on the right side.
• The 17" MacBook Pro models have 2 USB ports on the left side, and one USB port on the right side.

Due to the current USB configuration of the 15" MacBook Pro under OS X, use of USB audio devices is supported on the right-hand USB port only. Use of such devices on the left-hand USB port(s) is not advised because it may cause audio interrupts and/or dropped samples. However, the use of an iLok on the left-hand port has been qualified and is fully supported.

Due to the current USB configuration of the 17" MacBook Pro under OS X, use of USB audio devices is only supported on the right-hand USB port, and the left-hand USB port farthest from the screen. Use of such devices on the left-hand USB port closest to the screen is not advised because it may cause audio interrupts and/or dropped samples. However, the use of an iLok on the left-hand port has been qualified and is fully supported.

These USB port recommendations are specific to USB Audio devices only (such as the FastTrack USB or Audiophile USB). USB Keyboard and Control Surface products do not have a recommended USB port at this time.

Very nice job, Chris. Well done!

I want to single out this statement you make:

A DAC with a so-called poor USB implementation may have the best S/PDIF implementation on the market and vice versa. Thus the sound of a DAC may vary widely based on the input used.

This goes to one of the main bones of contention I have with the TAS article. In addition to its failure to discriminate (or even discuss) the different types of USB implementation, the article seems to lump all DACs with a USB input together in the category USB DAC. The way the ARC and Bryston review is positioned in the magazine makes it seem like these products are the ARC and Bryston entries into the USB DAC sweepstakes. I think that's misleading. These are their DACs, which happen to offer USB inputs. It's too late to put the genie back in the bottle, but we'd be better off if "USB DAC" was used only for products like the Ayre, Wavelength, and UltraFi that were designed specifically for USB.

I know, I'm splitting hairs.

Chris,
Great article.
My dog doesn't like to be wagged by his tail, so I bought the Ayre for him.

Aaron H

Great article, Chris. But I don't remember seeing or hearing any USB DAC's in your "state of the art" demos at the Symposium!

Jeff

Thanks
very clear.
Wow - you got a dCS front end. Well done. I hope you have got a transparent system. You are going to need it ha ha ha...
Best
Andrew

HI Guys - Thanks for the kind remarks thus far.

@ audiozorro - DAC manufacturers have definitely embraced computer audio but many of them are still figuring out how they'll get into the game. Meaning, should they do Firewire, USB, 24/96 or higher etc... Right now the economy has some manufacturers in a holding pattern as well. I'm positive you'll see 24/192 USB eventually. The manufacturers I've talked to are going to do it right instead of rush it out the door just to have a component that says 24/192 on the box. Don't hold your breath for 24/192 USB, it may be a while.

@ Soundproof - Thanks for the USB information. I wrote an article in April 2008 about problems with USB audio via certain USB ports on computers. It's definitely a concern that everyone should take serious, but steps can be taken to reduce and even eliminate the problem in many instances. Here is a link http://www.computeraudiophile.com/content/USB-Port-Not-USB-Port

@ DanRubin - That's a hair worth splitting.

@ jivers - You had to squeeze that in here didn't you. I would have liked to include some state of the art USB implementations in Studio A, but a line has to be drawn somewhere. I don't think the MBL / Magico v3 / Ayre QB-9 system in the lounge was a disappointment :~)

I dipped my toe into the waters of hirez with a little E-MU 0404 USB DAC, which does up to 24/192 at 4% of the cost of the top CASH picks. It doesn't need a Lynx card, but is able to take hirez files directly. The only issue is that it uses Win XP and not Vista. It also won't do 192 or 176 with a Mac. It can play the RefRec HRx files at 176/24 with no problems. I am using Media Monkey as the software. Larry

Hi Larry - There's no free lunch as I'm sure you know. The EMU DAC does support 24/192 and is only $200, but just because it goes to "11" doesn't mean much. I think there is good reason nobody else is supporting 24/192 via USB. While it's technically possible there are some major compromises. This particular DAC has very high noise related to the oscillators and the USB circuitry. Thus, the jitter is about 15x higher than some of the CASH list products that only support 24/96 via USB.

That said, this DAC does have a big following and many people are really enjoying the sound.

Chris, thanks for the explanation. As this was a test for hirez, I've decided to see what the fancier setups are like. I very much liked what I heard at the Computer Audiophile Symposium and today I had Tim Muratani and Mike Romanoski over to my home to scope out setting up a demo of the new Amarra Model 4 DAC and the software package that was demo'ed at the Symposium. It runs on a Mac through firewire, not USB. I am very interested in copying a fair amount of my vinyl onto 24/192 and use the software to remove clicks and pops. It looks like I would be able to generate the RIAA and other EQ curves in the digital domain as part of the A to D conversion process. I have a lot of old records (mostly EMI and Decca classical from the late '50s through the '60's that use non RIAA EQ. I'll let you know how it turns out. So far I have been very impressed by the expertise of Tim and Romo and how to optimize this all in my system. Larry

Hi Larry - Wow, what a small world. I just got off the phone with Tim about thirty minutes ago! I think Romo is a very knowledgeable and great person to work with as well. Please let me know what you think of the demo once you have it running in your system and once you have a chance to convert some vinyl.

Sounds like you have some very fun listening ahead of you.

Just some questions:

* what does it mean Polling Interval 1ms, in both adaptive and asynch mode?

* can the oscillator of the motherboard influence the SQ of adaptive USB dacs?

* can a greatly optimized PC audio reduce the difference between adaptive and asynch USB dacs? I'm thinking to a Linux based machine, with no tasks to do but MPD to play music. I think in this scenario the timing should be improved considerably

* what about the buffer of the TAS1020 chip? Shouldn't it help when dealing with jitter? If data coming from PC is used to feed this buffer, and the DAC clock is used to acquire audio samples from this buffer rather than from USB directly, I see no issues in the adaptive implementation.

Rider,

* what does it mean Polling Interval 1ms, in both adaptive and asynch mode?

You can see the Polling interval is also set on the adaptive. This is when the computer sends the Start of Frame (SOF) and is programable from 1-32ms

* can the oscillator of the motherboard influence the SQ of adaptive USB dacs?

Not really though if it does change the variation of the SOF frame then yes it would effect adaptive. But I have not seen that happen in my experience.

* can a greatly optimized PC audio reduce the difference between adaptive and asynch USB dacs? I'm thinking to a Linux based machine, with no tasks to do but MPD to play music. I think in this scenario the timing should be improved considerably.

Actually not, see it has to do with the variation in timing between the Adaptive device. Optimizing your computer will only effect the variables inside the computer which makes things sound different. It will not effect the adaptive device any.

* what about the buffer of the TAS1020 chip? Shouldn't it help when dealing with jitter? If data coming from PC is used to feed this buffer, and the DAC clock is used to acquire audio samples from this buffer rather than from USB directly, I see no issues in the adaptive implementation.

Actually the way the code is setup in the TAS1020, NO. See if you look at figure 2-1 on page 25 of the TAS1020B you can see that the the SOF goes into a 16 bit timer and this is output via ACGCAPL/ACGCAPH. These values are used to dertermine the difference between the SOF timing and the internal timer and the difference is applied as the Adaptive difference into the Master Clock (via Frequency Synthesizer).

See there are two jitter problems with Adaptive:

1) The inherent change in the Master Clock due to the way Adaptive protocol works.

2) The Frequency Synthesizer it self has significant jitter even if you set it and left it there. More than 100x that of an external fixed oscillator.

~~~~~~~~~

24/192: Gang this protocol requires Class 2 Audio support which is not available in Windows in Linux. When they get closer to implementing it we will have a plan to support it.

Thanks
Gordon

thank you for your prompt response and continuous knowledge sharing.

G'day Gorden
great to see your dacs have found their way down under at long last.

re: "Actually not, see it has to do with the variation in timing between the Adaptive device. Optimizing your computer will only effect the variables inside the computer which makes things sound different. It will not effect the adaptive device any."

A fortiori are you saying that the only thing effecting jitter is the usb implementation?

Cheers
A

A and the rest of the Gang,

Correct... a computer cannot add jitter to the system what so ever. It is only when the PCM data is converted to I2S (or L/R justified) that jitter will occur.

The implementation of this will be critical to the overall jitter of the system.

Thanks
Gordon

I'm not sure I understand this jitter problem.

Firstly, if the PC sends samples from its memory buffer at 1ms intervals (as per USB specification) to the DAC's own buffer, and the DAC, using its crystal oscillator as a clock, then processes those samples, where do the variations in timing arise?

Secondly, and more important, does it even matter? According to the AES, jitter below 20ns is inaudible, and even something as dependable, common and cheap (and therefore inherently non-audiophile) as the M-Audio Transit measures a mere 2ns.

All answers gratefully received.

Sid

Hi Sid - Welcome to Computer Audiophile. Please post your second question about jitter in a new thread on the forum. I don't want to derail the discussion covering a USB Audio Primer with very technical jitter talk. We all know jitter conversations carry on forever and nobody comes out a winner.

In the new thread can you also post the AES document that states jitter below 20ns is inaudible? There are several types of jitter, some matter, some don't. Measuring jitter is very difficult as well. I am really interested in the AES testing methodology in how they determined the 20ns inaudibility.

Thanks a ton Sid!

Oh dear.

I've stumbled into The Computer Audiophile Jitter Wars, with my very first comment. I'm sorry; I should have spent more time in the forums first to get the lay of the land.

So, very briefly, here's the AES paper:

E. Benjamin, E. Gannon, Theoretical and audible effects of jitter on digital audio quality (1998)

And here's a more recent study published in "Acoustic Science & Technology":

K. Ashihara et al., Detection threshold for distortions due to jitter on digital audio (2005)

Phew.

Sid,

Adaptive USB 101:

Every 1ms a frame called Start of Frame is sent. The SOF frame is a packet all by it self. It does not really mean data is coming down, just that it is another start of a time period as set in the enumeration, which in this case is 1ms.

The USB receiver also has a 1ms timer going on. The SOF time stamp is subtracted from the USB receivers time stamp and the difference is applied to the Master Clock output.

Changing the Master Clock on the fly like this adds jitter to the out going I2S (or R/L justified) link.

The computer and or operating system can effect this. But with newer USB controllers the processor need not actually evoke sending the SOF, instead the controller does this on it's own.

Therefore it is less likely that the computer will effect the jitter but really the difference in clock speeds between the two devices will determine how much jitter is added to the I2S stream.

Thanks
Gordon

Good job on the USB primer 101! - even though it sounds a bit like an advertisement for asynchronous USB products - ex. Wavelength - who happens to advertise on your site! -

Another perspective on why manufacturers may be taking the easy way out by using "quick and dirty" adaptive techniques, is that wired computer based audio may be a passing fad! Lets face it. Getting rid of the wire is MUCH more flexible and convenient. Even companies like HSU research is now using wireless for their subwoofers! If you want to run two or more subs, not having wires running all over your listening area is a key feature! The opinion that wired is always better than wireless is as debatable as the superiority of USB over SPDIF! Like you said - its a matter of implementation.

Bada bing bada boom! - Gotta love the research - not much of that in audiophillia land!

Hi Eric - Can I ask what you are "bada-binging" about? Did you read the research papers or did you not do much research like audiophilia land?

It seems like you have something against high end audio and are seeking statements to cling on to that purport to discredit high end audio. There is actually a lot of research that goes on in audiophilia land. Extreme engineering is what brings us some remarkable products. Without research we would still be listening to perfect sound forever, also known as CDs. I also think consumers do as much research as they want. The readers of CA are doing the research right here.

Again, I'd like to keep this article on topic. Talk about USB Audio is what this should be all about. We can certainly take up jitter in a new thread in the forum, or we can continue the plethora of jitter threads already played out.

Hi Eric - Thanks for bringing up the fact that I spoke about Wavelength products and used Gordon Rankin as one source of data about USB Audio. It's critically important to keep this in the forefront. This is why I made it clear in the article who I used as sources of data during my research. Also, it's great that all the readers keep me honest by leaving comments on anything that may seem improper. It is almost impossible to research Asynchronous USB thoroughly without talking to Wavelength Audio.

Again, I get your point 100% and it's always good to discuss it when reading any article.

Take it easy Chris! - I LIKE research, and I LIKE high end audio. You completely misinterpreted my comment!

You like to challenge people who make assertions by asking for the research to support it. In this case I was responding in a positive way with the "bada bing" comment! The writer responded with the research you were requesting - you probably were not expecting that! I did read the research and felt it added to the discussion.

I think you need to take a look at how much of a control freak you are, unless you just want writers that tell you and your little sponsors how great you are!

cheers
Eric

Eric - Your response was exactly what I was expecting. It also says much more about you than it does me. Personal attacks like calling me a control freak are very sophomoric, especially when you include statements like, "... you and your little sponsors ...". This is classic playground terminology that only takes away from any point you were trying to make. I take pride in the fact I am not a control freak and I encourage all comments whether they agree with me or not. In fact I even added a sentence about that in the article. Take a look around the site here, you'll see many comments that disagree with what I say and add much to the discussion. A one-sided site ads nothing and would be a disservice to this wonderful hobby and hobbyists. I am no preacher and the readers of CA are certainly not in the CA choir.

Asking for research is certainly not challenging people. It's actually trying to get more information in front of those who seek it instead of relying on hearsay. I am actually very happy the links to the research were provided and I expected the links to be provided. Your comment about reading the research and feeling that, "it added to the discussion" was close to something I would have said in college if asked a question about research I really did not read.

Saying you responded, "in a positive way with the "bada bing" comment" is a little misleading. I appeared to me as more of a "gotcha" like you thought I was burned by the fact that research links were posted. It also appeared to me like you rubber stamped the fact that links were provided without reading the material and doing more research which you are a big fan of. More research into this topic including counter arguments to the papers provided clearly leads one to conclude there is no consensus on this topic. I asked for the link from the reader because I wanted to make sure he was not just saying something he had heard through the grapevine. We all know forums are full of ideas that live on forever often repeated without any knowledge of the origination of the idea.

Thank you Chris

It helps clarifying what we are talking about when discussing USB DACs.
A part from Wavelength, DCS and Ayre, are you aware of other manufacturers having developped asynchronous USB interfaces, or being in the process of developping it?

Thanks

Hi Fyper - NObody I know has developed an asynchronous USB DAC at this level. The EMU 0404 is technically asynchronous via USB, but it's design and implementation are 180 degrees different than Ayre, Wavelength, and dCS. There is a reason the EMU is only $200 even though it's one of the only async USB DACs in the world.

http://www.bd-design.nl/contents/en-us/d168.html

Which I think existed earlier than you heard about async USB ...
I worked on the software myself in 2006.

I like to have this as an example that it is not (or at least not always) the big audio companies who are innovative, but the power of high qualified individuals is.
You could call this one obsolete already (16/44.1 only), but there will be more ...

Interesting Peter. It appears to be using Bulk mode similar to FireWire. What chip does it use?

Peter,

Bernt is a good friend of mine and yes this would be considered an asynchronous device. But it is not an ISO Asynchronous device and therefore would require drivers.

I would not call 16/44.1 obsolete a number of my customers think the dynamics of the NOS style dacs sound better than current 24/32 bit units.

Thanks
Gordon

Chris ... you comment on using USB Probe - is this a piece of software I can download from somewhere??

Eloise

AD1865.

I don't consider it similar to Firewire. It uses an internal buffer of 64KB (and uses USB 1.1 IIRC).

Good to hear. All I know is that I told Bert to find you at the last RMAF and talk a bit about things. I don't think that really happened, you being quite busy.

It is true that it requires its own drivers, and for that matter its own player software. Not a good thing by itself.

a number of my customers think the dynamics of the NOS style dacs sound better than current 24/32 bit units.

At least I try to keep that up. (but for 24/192) :-)

Peter

Peter,

Yes he did stop by at RMAF and we talked for a couple of minutes but it was mass hysteria at the room.

BTW you mentioned that small companies are not the driving force here. I would have to put WA as one of the driving forces and I am not a large company. Actually if you look at computer audio none of the large companies are really prepared for it. They are all using technology from some other company.

In regards to the 24/32 @192.... got me I think a number of people are ripping their cd collection and thinking it's 16/44.1 why do anything more than that.

Also the AD1865 dac chip is an 18 bit part used by Audio Note also. He could just write a ASIO wrapper for .NET and be compatible with a lot of applications.

Thanks
Gordon

Thanks. Hoping that Chris allows for a little more offtopic :

It seems that I suggested that only the big companies can get these kind of things done, but I actually wanted to say just the opposite (also still having in mind the slaughter of A.J. and T.F. at me stating similar before) : the real progress is made by the small companies and even individuals. There is logic in this I think;

Small companies who give all their heart to the product, and individuals just the same, may work 1000s (yes thousands) of hours to get something done, which would not be allowed by the big companies who just have to pay for that, possibly by hireing expensice contractors. The easy example is myself. I must be somewhere at 6000-7000 hours on the development of XX, and against my normal hourly tariff even my own company (ehh, owned by me) would not have paid for such an investment. Not with a selling price of EUR 72, and not with a selling price of EUR 999. It would require sales amounts of over 10,000 and 750 units respectively, and neither is feaseable (on beforehand, when decisions for investments must be taken).

Another kind of example would be the development of a loudspeaker. So, talking about Bert, I must have spent 500 hours at least on helping him with let sounding a horn not as a horn. Again against my tariff he sure wouldn't have spent that. Helping as a friend does wonders here, and friends are never big companies (unless something would be in it for myself afterall).

So, the real innovative stuff not only really happens with individuals (who just need to be very good at their job, like electrical engineering), but it also is logic it goes like that.
Of course these things happened at the large audio companies in the past, and it just may these days. But economy is not really challenging right now.

Peter

Chris ... you comment on using USB Probe - is this a piece of software I can download from somewhere??

This is where I get it:

http://developer.apple.com/hardwaredrivers/download/usbdebug.html

I noticed BD design lists using any USB cable as a feature. Probably the case though the I imagine the esoteric cable companies will disagree. I also look forward to more ethernet DACs.

But of course it is true that just a random cable can be used, since this way of using USB is just about data (and not about audio !).

Do you care about the USB cable when you, say, download pictures from your digital camera ?

Peter

While the innovation comes from the small companies I would suggest that it won't take long for the big companies to get involved if they see a market. Perhaps 2010 will be an interesting year for USB audio.

Hi Peter - nice to see you about!

Best
Andrew

It will be interesting to see which way the specialist HiFi manufacturers go.

So far we've seen Bryston and Audio Research go the route of basic USB connection which can easily be improved on. Ayre go the route of licensing WaveLength's async USB technology and soon will have Naim launching their new (first) DAC on which any form of dedicated computer interface is conspicuous by it's absence, preferring to support optical or SPDIF either direct to the motherboard or via third-party interface.

Eloise

It's hard to generalize and implementation is everything but if we just look at the DACs on the CASH List the interfaces that have provided sonic superiority are:

Berkeley Audio Design Alpha DAC – AES
Weiss Engineering Minerva – Firewire
Devilsound DAC – only Adaptive USB offered
Bryston BDA-1 – AES
Benchmark Media Systems DAC1 HDR – AES
Wavelength Audio Proton – only Asynchronous USB offered

IMO the excellent digital interface cards and software drivers with the Lynx AES16 and Juli@ give them the slight advantage. The CA reviews of these DACs seem to confirm. What I find surprising are the FW DAC owners that use their DACs such as a Weiss DAC2 or FireFace 800 as input to the Berkeley DAC. What is noticeably absent in the above is toslink, which is almost universally said to be an inferior interface in terms of sonics but which many manufacturers seem compelled to implement for CD transports and legacy systems. And obviously the best USB has to offer is excellent 24/96 but nothing higher.

The question of whether or not you need to do format conversion and where the clock resides in the chain seems very important but often gets confused in these comparisons. A lot of apples and oranges discussions.

A comparison I would pay money to hear is Lynx -> BADA (or Bryston or whatever) compared to Ayre USB DAC compared to Firewire directly to Weiss Minerva/DAC2 compared to Firewire to Metric Halo ULN-2 or 8. No interface conversions, but still apples and oranges in that the DACs are entirely different. One of the things I am curious about is, independent of the DAC, does USB have "a sound"? Does Firewire?

Does the Lynx card output AES natively?

When feeding a BADA or Bryston DAC, is the Lynx calling the shots (clock wise), or is the DAC the master clock.?

Hi Dan - When using a lynx with the Alpha DAC or Bryston BDA-1 there is no master clock. The Lynx and each DAC would user their clocks. This is because there is no word clock in or output on the Alpha or BDA-1. The dCS Paganini I am listening to right now has word clock in and out and the Puccini u-Clock has four word clock outputs. Thus I am clocking the Lynx and the Paganini via the U-Clock (master clock in this configuration) when under 96 kHz.

For completeness, could an explanation of the use of Bulk USB in audio be given? I have in my mind some idea about this being used as the main USB protocol for external disks but I think it is used in audio too just can't remember in what products. Does it simplify USB communication, not needing asynchronous communication by using a big buffer & reclocking the data out?

JKeny,

Consider the amount of time you are investigating this why not just go to USB.org and look it up.

Bulk cannot be used in Audio without a driver. If you use either of the bulk or interrupt endpoints you would have to write a driver.

It actually complicates things using bulk and interrupt because there is no guarantee that the packets will reach the endpoint in a continuous time period.

Only Isosynchronous USB is guaranteed to support Audio and has the highest priority on the buss.

Thanks
Gordon

Just confused by reading various post regarding asyn usb implementation. I think it is implemented in some control feedback to adjust USB to send faster or slower to match with the DAC clock. But what that means is that the incoming data rate is constantly being adjusted and always slightly different from DAC clock. Is it being resampled? What would DAC do to handle this problem. I'm thinking of CD player, the CDP is constantly adjusting the CD spinning rate to have a constant data stream. This process may create a lot of jitter.

Another question, if DAC clock is slower than the speed of player software. With Asyn USB, the USB transmission follow the DAC clock. So we have a player software keep sending data with its own speed, which is faster than the USB transmission rate. What the PC is going to do to handle it?

I think if the player software also slave to the USB transmission speed, it should be fine. But how to make a player software slave to USB transmission speed? Is there such a player.

If we can test whether wavelength DAC is bit perfect, it is going to help understand whether the implementation slave the audio play and USB from end to end. If not, then there would be some resampling by PC which may impact audio quality.

Anybody test whether Wavelength is bit perfect with USB interface?

"Anybody test whether Wavelength is bit perfect with USB interface?"

I'm not sure you are asking the right question here. There is no way to test such a thing because the output of all DACs is analog. Thus, the bits are "gone."

if DAC clock is slower than the speed of player software. With Asyn USB, the USB transmission follow the DAC clock. So we have a player software keep sending data with its own speed, which is faster than the USB transmission rate. What the PC is going to do to handle it?

Either async or isync, the player sees a buffer. When the buffer is half empty (or whatever is decided for) the player receives a message. Next the player becomes active and fills the buffer. This goes on and on.
With the async that needs a driver (see Gordon's post) the buffer is at the DAC side (in de DAC box).

Peter

Smallp,

We will release at RMAF an Asynchronous Streamlength USB to SPDIF converters for those people who want to use their present dacs. This was also the what we would send potential clients who wanted to buy Streamlength code using it either as I2S or SPDIF into their solutions. The reason we did this was to truly verify bit accuracy which it did.

But before that we used my Tektronixs MSO Series Oscilloscope with the I2S fittings which can actually test the I2S link between the TAS1020 and the DAC chip. In our tests we would use a audio file I generated, we would then feed this to the dac and take samples at the DAC chip and also on the USB link using our Total Phase USB2.0 Analyzer to verify first the computer is sending bit accurate data and then at the dac chip.

But remember in Async mode you are not changing the sample rate by small amounts to assure the link is stable you are merely sending flow control data to the PC to assure the link is stable.

Thanks
Gordon

Could Gordon or someone else explain why an asynchronous USB to SPDIF converter would be a good thing? I'm not meaning to be snide, I just don't get how it adds value. Won't you lose all the benefits of asynchronous conversion once you start feeding SPDIF to your DAC?

Dan,

No an SPDIF solution would not be as good as an Async full USB solution if done correctly.

The only reason I am doing this is some people seem to be married to the dac they have now and want some really low jitter solution, so I am giving it to them.

Thanks
Gordon

Excellent and very helpful article!

On the usb cable question posted previously. I originally used a cheap 12 foot usb printer cable with my wavelength brick. I then got a belkin gold 10 foot usb cable at staples and it did sound better. I am not a cable snob. I use zip cord for short speaker runs. I had thought that either the zeros and ones were transmitted or they were not. Any opinion on why the sound is different or am I hearing things?

Thanks for the info Chris,well done as usual,Bob

Thanks, Chris, for a great article.

I am a novice in the area of computer audio. I have been building a computer based system over the last 18 months and have been making gradual improvements over that period. First by upgrading to higher res files on a Mac Pro (I use AIF – storage is cheap), then by adding a USB DAC (PS Audio). Done separately over several months, both improved the sound of the system considerably.

Last month I read the TAS article on USB vs. firewire. Being the inquisitive audiophile that I am, (a “sickness”, we call it at home…) I gave it a try. I bought a Saffire and installed it. In my system, there was a remarkable sound improvement, far greater that the modest cost involved. It was, in my opinion, everything the reviewer said it was. This endorsement may be meaningless to those reading an e-mail, but for me the proof is in my desire to spend time listening…reexamining and exploring our music collection all over again. I began to wonder why USB was seemingly being pushed in the marketplace, so great was the difference.

Then I found your web-site while doing a search on firewire DACs.

I found your site to be very informative, and especially interesting was the forum on the USB interface. I am pleased to read more detailed information than what was available from the TAS article, and it helps me to understand why USB will continue to be used as an audio interface. That being said; for my system, with my current equipment, the firewire interface is miles ahead of USB.

Still, I have to wonder…now how would a new DAC sound…?

TAS is a joke. After reading Bob Harley's opinion about S/PDIF vs USB DACs, I would never read anything by him again.

Hi Chris,

One thing to consider is a simple test to determine the contribution of USB in the equation, at least for certain manufacturers…Assuming you have two identical interfaces with both AES-EBU and USB inputs, no small assumption I admit. With all other factors the same, play a high quality file into one interface, using that interface as a USB–to–analog converter. Make a subjective quality judgement. Then, play the same file out of the first interface's AES out into the second interface, using the second interface as an AES-to-analog converter. Again, make a subjective quality judgement. Because the interfaces should be identical, the only difference would be the data path and data recovery method.

Also, since someone else mentioned a 1394 interface and this is about USB as a design choice, folks may want to read another, albeit biased, view of USB for audio applications:

http://www.seneschal.net/blog/index.php/2009/bus-wars/

Ok there is tons of material on USB options, firewire and various interface converters. But no one seems to know which one is the better option. Infact, the audio media has done a very good job of confusing the audiophile consumers. Chris you are quick to respond to Absolute sound but you do not talk about comparing your experience with firewire or lynx card option are any other...why?

Moreover, if Wavelength audio is a real USB option, compared to the one Absolute sound used. Then why did Wavelength decline to participate in Absolute sounds tests. I would think this would be their opportunity.

I feel its all an act. The media and manufactures want to keep the consumer in the dark so they can make there buck. Sorry I am not upset but stating what i read. I have read a lot of articles, but they all shy away from any concrete decisions. Now i know i will get a response, it all depends on configuration and components used. But I am looking for a straight comparison with USB (with a top USB cable) versus Firewire versus Lynx card . Or Empirical audio versus weiss converter (in the price range) all other components staying the same period.

FYI i am currently using a USB option myself. But i have been trying to find out what is the optimum solution... Thanks!

Ajay,

I think if you read a little more, you will find why I removed myself from the Absolute Train Wreck as I put it.

Neither writers were prepared or even willing to really know anything before they started. In Alan's case who is a millionaire why is he using a $550 outdated laptop with a Media Monkey a program known not to be bit true. Alan's other computers being Windows 2000 which does not even support Asynchronous USB spec and a really outdated G series iMac.

Look these guys have no problem using a $800 power cord but cannot even get in step with what is required to pull off even a satisfactory computer hardware requirement to review these products.

When I heard the plan (what plan??) I pulled the product.

It is very simple to see that anything using an asynchronous is going to be better than some adaptive mode.

There are several ways to do this with either Firewire or USB. The good news is these products are out of the harsh environment of the PC's noise machine.

Both USB and Firewire have Async direct sound in the specification but no OS at this point supports that with Firewire.

So you can write drivers on Firewire or USB and use Block mode and write a matching device driver and have very good results. That is if you pull the rest of the puzzle out. This could then support any sample rate that the hardware supports and the computer can feed.

USB Class 1 Audio System Support (meaning it does not require a driver) is supported now in OSX, XP/Vista/7 and Linux this is good to 24/96 as the buffer size becomes a USB limitation above that.

USB Class 2 Audio System Support is only supported at this time in OSX. This and High Speed USB are required for for products that will excede 24/96 and is capable of 24 channels at 192K/24 bits from my calculations.

Thanks
Gordon

My understanding is that there is no equipment available yet that will allow 24/192 from a USB DAC. 192 KHz is penetrating the market currently and sounds remarkable!

robjob,
Not true - check out the Musiland Monitor range of DACs - 24/192, asynchronous USB at a price you won't believe!

Also the EMU 0404 USB for $200 street price does up to 24/192.

The issues with devices such as the EMu 0404 USB2.0 and the Musiland Monitor range is that they do not support USB Class 1 audio (think that is correct term) and therefore need their own drivers. If this is a problem is something only "you" as a purchaser can decide. The Musiland Monitor is limited to Windows (XP and Vista I think) and the EMu 0404 works on a Mac only at 16/44.1 & 48 digital output and 24/96 analogue output (according to their website, some people have commented that there may be new drivers supporting full rate). Also if later you change OS, then you may find a device that worked fine before no longer works. The devices that Chris meantions (Ayre, Wavelength and dCS) all use the generic Class 1 audio driver which is supported in Windows, Mac and (I think) Linux; and will be supported in new operating systems for as long as USB exists - so for a $200 device unsupported OS is one thing, but at upwards of $1000 for Wavelength Proton to $20,000 (approx) for a dCS Scarlatti Upsampler / USB interface its a completely different issue.

Yes: this is the same for FireWire devices I know.

Eloise

Gordon,

Could you please elaborate on your claim that MM isn't bit true? On the TAS forum too you said that it was "a well known fact that MediaMonkey is not even bit true".

Chris,

In one of your earlier articles you gave advice on how to set up a Windows-based system with MediaMonkey. Did you also find that it wasn't bit true?

Thanks,

Sid

Hi Sid - I've always been able to get bit perfect output using MediaMonkey.

MM Native or Driver

If you are using a DAC without specific driver, specially written for this hardware, then beside the strange Japanese ASIO out to the ASIO4ALL wrapper, you have absolutely no chance to get a bit perfect output.

I haven’t tested it with the special lynx driver to verify if this works in those case, but I can report, that displaying the HDCD sign or the Dolby or DTS sign is no real sign, that the stream is 100 % Bit True.

For example if you set in Vista SP2 the output setting to the correct sample rate you can get with non exclusive WASAPI also a 24 Bit HDCD light going or and Dolby and DTS logos, but this isn’t bit true.

I have a special signal, created out of two Audio Precision signals. The left channel carries a 16 Bit digital DC (constant value at maximum level) and the right channel carries a 24 Bit walking zero signal (all bits are “1” except one bit is “0”, and this “0” Bit is walking synchronous within the 24 Bit range.

So with this signal and a Soft or Hardware that is able to display the bit statistic (like Audio Precision, or Digicheck and RME hardware, or in the program WaveLab or in the Plug-In Ozone from iZotope) you can watch these bits.

So for example in the non exclusive wasapi mode you should think, that it is bit true due to the 24 Bit HDCD sign, but this special signal shows you within seconds, that it isn’t. So with MM and the lynx driver, the HDCD is a good start, but not a save test.

Juergen

Hi Juergen - Maybe I'm missing something here, but it seems like you've gone to great lengths to achieve a non-bit perfect signal. If I read your first paragraph right, a specific driver, the ASIO driver, and ASIO4ALL give a bit perfect signal. This leaves a couple options to mess up the audio stream, just like all other applications. Is non-exclusive mode very popular? I can't see a reason to use non-exclusive mode like you've done in your examples.

How are you testing your HDCD content in the following situation? Are you using a Pacific Microsonics Model Two or an Alpha DAC?

"For example if you set in Vista SP2 the output setting to the correct sample rate you can get with non exclusive WASAPI also a 24 Bit HDCD light going or and Dolby and DTS logos, but this isn’t bit true."

Did you run your special signal through a Model Two D to A, then A to D to get the HDCD bit on the special signal?

"...I have a special signal, created out of two Audio Precision signals...So for example in the non exclusive wasapi mode you should think, that it is bit true due to the 24 Bit HDCD sign, but this special signal shows you within seconds, that it isn’t."

So if I understand this, you created a special signal, added the HDCD code on the 16th and 24th bits of the L or R channel, then run this into a DAC that has an HDCD indicator such as an Alpha DAC or Model Two?

Chris

In the last two months I have written a test procedure for the Audio Precision in order to verify very quickly, whether a application and setting in computer audio is set to Bit True or not. I have recognized that some drivers and or applications behave different for 16 Bit signals than for 24 Bit Signals I have a 16 Bit signal on the left channel and a 24 Bit signal on the right channel.

With the non exclusive mode in vista, this was only an example, not a recommendation, how you can cheated if you are not using a highly sophisticated signal. I am not so deep into the operating system level to explain, why for example a J-Test signal with digital black between the frequency bins a played back correctly in this case, but the DC signal, constant value, not.

So my purpose was “only” with what signal I could get totally sure, that a device is bit true with 16 and with 24 bit signals, and also not swapping channels. So I tested a lot of software and operating system combination to look whether a system is or not bit true.

Right now I am at the High End show in Guangzhou, but if you like, I can send you this special test signal (packed with FLAC for smaller size) for the 4 different sample rates, when I am back home. I am sorry that I will not be at the Rocky Mountain show in two weeks, to have some discussions with you, but I have too much work to be done (BTW Maier Shady knows me very well and also Tim should remember me from “older” days).

With HDCD (and Dolby and DTS) I have not created these signals. I have used “regular” HDCD Reference Recording signals and regular DD and DTS soundtrack signals and played through an Alpha DAC and through surround processors with different software / settings, to see whether this could be a correct indication but noticed, it isn't.

I know you are using Media Monkey and Lynx AES16e PCIe card and using the wave driver. I have an older lynx L22 card in my small recording studio (semipro), but not in my home office, so I haven't tested it with that combination, but at home with some native usb devices and with different EMU soundcards, I couldn't get a bit perfect signal out of MM and also through most other software.

For XP, I get only bit true, if a software offers ASIO out and I am using for this hardware written ASIO drivers or the ASIO4ALL wrapper. Without ASIO out (or KS out) I am getting no bit perfect out on native USB cards, and with no, I really mean no. But I do not know what's in your case with wave drivers for the lynx AES16e card. And for Vista, it is very similar but you have a third way, which is the WASAPI exclusive mode, and that works very fine, if set up correctly.

I hope this clarifies a little bit.

Juergen

Hi Juergen - Thanks for the info. I wish you were going to RMAF. I've actually talked to Maier about you and I would love to chat about some of the stuff you are doing.

We'll have to meet up at another show.

Thanks Juergen

great post! Did you happen to try with Linux and MPD (Music Player Daemon) too?

Chris, I am sorry, that I will not be at this RMAF, but on the same time, there is also a big High End show in Japan, but definitely I will be in the States the next time for the upcoming CES in Las Vegas in January 2010. So if you plan to set up a computer audiophile Seminar in Las Vegas, I will be more than happy to participate this, also as a panelist, if you like. So please contact me, if this would be the case.

Juergen

PS: A funny story here from the show. At one room, they are using the Ayre USB converter and a Mac Mini front end and playing back CD files (44.1 kHz) but on the Ayre display I can read 88, so I ask for that reason. The answer was that 4 is a bad number here in China and 8 is a lucky number, so they prefer to sample rate the data up from 44 to 88 and have a lucky number on the display instead of having a bit true playback and seeing the bad number 44. Isn't this funny?

Juergen,

Yes "4" is from the Gang of Four. You will notice any company will skip the 4th version of a any product. Instead they will either rebrand it or goto 8.

Larry Kay (Fi/TAS) had a house in San Francisco that was #8 on the street and he new he could ask more because of that.

~~~~~~~

I have not had much time to do this but I agree that we should sit down and map out what works and does not work at the Application Layer, Interface and Operating system. The good thing is this is all software and many companies are asking about their bit true.

Maybe we can setup something with an USB/Firewire to SPDIF and PCIe to SPDIF. We can then verify easily what is and what is not bit true.

But another thing I think we should look at is timing. I have been sending HEX oscillator files to my USB to SPDIF converter as a square wave. I think another thing is to look at the duty cycle and see how accurate the timing is.

We can also check bit true at the USB level with most of the instruments I have and verify known files if they are bit true.

Gang,

Most of the "Bit True" tests are averaged. Meaning say we send a known square wave out at 1KHz @ 1vrms. When we look at the scope side of the data it shows all the data so ok it's bit true. This is typically the way most companies test for bit true. But what if we had a known file. Then we pass that to one of these SPDIF converters and verify the entire file bit for bit. I think that would make a lot more sense.

Thanks
Gordon

Juergen, that's the best reason I've heard to justify why the minimum sample rate should be 88.2 kHz instead of 44.1 kHz. Maybe if the large Far East market is successful in ditching 44.1 kHz, which is typically downsampled nowadays, the rest of the world may follow suit and we'll all benefit from higher resolutions and unnecessary downsamples.

Hi Juergen, thanks for all this. One small addition :

For XP, I get only bit true, if a software offers ASIO out and I am using for this hardware written ASIO drivers or the ASIO4ALL wrapper. Without ASIO out (or KS out) I am getting no bit perfect out on native USB cards, and with no, I really mean no.

I know you said "USB cards", but since you're into RME ... RME's MME drivers are bit perfect for XP.
But ... what about the Fireface USB version ? that would be an interesting twirl.
Notice that MME does not exist for Vista.

Peter

@PeterSt: You are right, RME cards are bit true with MME and with ASIO drivers. I have three of them, in my older computers. The drivers work rock solid, this is a big plus for RME, but for the reason that the jitter is slightly too high for me and also the distortion on the analog out, I haven't bought any more in the last 3 years or so.

@Audiozorro: I am sorry that I have to inform you that Brint (I do not know the correct spelling), the international sales manger of Ayre, has set back the Audio Midi Setup of the Mac Mini to 44.1 kHz. Is this now good to have Bit True or should he better stay with 88 in the display to have better luck in China?

@riderforever: I have started with some measurements under Linux Ubuntu 9.04 and under Ubuntu Studio. The 16 Bit where fine, but for 24 Bit I haven't had enough time, so I stopped it here, because even only with XP, Vista, Tiger and Leopard, it takes a lot of time to verify all what is said in different post, if this is right or wrong.

@Gordon: The 24 Bit walking zero signal is nearly perfect to measure Bit correctness, because the source signal is known and you are walking through the complete dynamic range of the digital system. The 16 Bit digital DC signal is perfect to look at what is happening with the bits does give only correct information if your system is 24 Bit (if you have a 16 Bit hardware BB PCM270x etc., they cut of the 17 – 24 Bit) so in this case even native Wave Out or native Direct Sound out looks correct in the 16 Bit world. But with a combined signal, as I described some pots earlier, I could check everything an one time, together with channel swapping. Concerning timing I have made also some other post, where I measured the jitter difference between native Wave Out (higher random jitter), native Direct Sound Out (higher discrete jitter) and “native” ASIO4ALL out or exclusively WASAPI out (as low as the hardware design can get), which is really clearly visible and I can tell you from the graph what native driver mode is used.

So if someone set up a computer with a RME card and the DigiCheck software and use the digital in as a “slave” (not master mode), and play back the DC / Walking Zero signal from the host computer via digital out will be able to see whether the play back chain (software player and setting) is bit perfect or not, or use a dedicated measurement system like Audio Precision or similar.

Juergen

PS @Gordon: I visited Larry Key, a long time ago, at the Fi times, in Sausalito, and the funny thing was, that he lived next door to James Hetfield (Metallica), so was this the lucky number 8?

PS @PeterSt: For Firewire I have only a KRK Ergo and this has only analog outputs, so I am sorry, that I can't make any digital Bit True tests.

Actually 4 is not from the Gang of four. It predates them. 4 in Chinese is the same sound as the word for "death." So 4 is avoided. 8 on the other hand is half of the character for "happiness". Therefore 88 is even better, the full character for happiness. In Hong Kong, people bid for auto license plates with lucky numbers. My father-in-law had AA 717, which was considered lucky, because 7+1=8 and 1+7=8, so he had a surrogate for 88. I suppose 176 would be even better, since it is 88 +88, double happiness.

Larry

Jeez Chris,

Things have got very technical on computeraudiophile in recent months!

Seriously, there is a phD in probably all of these separate issues. Perhaps dozens of phD's actually. Is there a Berkley or Stanford or MIT department of electronic engineering looking at any of this stuff? What we need is academics with a non commercial interest to look at all this. Also throw in some people who have a sound background in audiology, statistical bias and how to design a proper double blind trial. Non commercial universities are the answer Chris.

Aren't we getting a bit off the track...there are many more practicle issues in choice of DAC interface/software/computer chip etc...

...such as how does the human use the computer. Where is the audio gear positioned in the room? Are the DAC's available? How robust is the interface? How easy is it to access the digital files? Does it all crash etc etc........Nobody say's the ipod is the best electronics interface to play computer audio (far from it), but I bet it is the most popular device in the whole world right now for doing just that....

Why don't we leave the electronics to academics, and the practical stuff to the commercial marketplace?...which will soon sort out which interface/software/etc is better for everyone. When the universities produce the papers and they are scientifically reviewed, we will have our answers to the more complicated stuff! Forum members need to publish references from scientific journals if they wish to support their cases. It's getting that complicated.

Just my thoughts Chris. please don't take offence....

From Computer Audiophile: "Hi Eric - Thanks for bringing up the fact that I spoke about Wavelength products and used Gordon Rankin as one source of data about USB Audio. It's critically important to keep this in the forefront. This is why I made it clear in the article who I used as sources of data during my research. Also, it's great that all the readers keep me honest by leaving comments on anything that may seem improper. It is almost impossible to research Asynchronous USB thoroughly without talking to Wavelength Audio.

Again, I get your point 100% and it's always good to discuss it when reading any article."

I was also curious why the only 'expert references' you used for this article are companies that make asynchronous USB products. Perhaps you should have spoke with experts who implement the adaptive mode to balance the information you received to write this article.

All the best,
Elias

I've bee the owner of a Benchmark DAC1 Pre for about a year. I originally got it just as a DAC for CD. I soon discovered it was useful for playing ripped cd's from my XP computer and found that very beguiling. In your piece you have the following statement:

"Another less common adaptive USB implementation is done using a TAS1020 chip. Manufacturers then have a choice of implementing the chip exactly like the PCM270x without additional programming or possibly using the example code provided by TI, or the manufacturer can purchase code from CEntrance, Inc. to use with the TAS1020. Popular devices using the CEntrance code are the Benchmark DAC1 variants, Bel Canto USB Link, and the PS Audio Perfect Wave DAC. Using the TAS1020 and CEntrance code greatly enhances the USB interface and allows native 24/96 playback without the need for additional device drivers or special software."

The Benchmark documentation goes to great lengths to assert that their implimentation completely eliminates jitter. Your comments in the paragraphs following sugguest that this is not, or at least, may not be true, and that any non asyncronous design will have jitter. Could you comment on this please.
Also, is the Benchmark design one of those that "Some listeners report as a Hi-Fi type of sound that is initially impressive, but long term listening may confirm otherwise." I've not noticed long term listening fatique, but I would like your opinion.

paugust

"The Benchmark documentation goes to great lengths to assert that their implimentation completely eliminates jitter. Your comments in the paragraphs following sugguest that this is not, or at least, may not be true, and that any non asyncronous design will have jitter. Could you comment on this please."

Hi paugust - I think even Benchmark would agree that it's impossible to completely eliminate jitter and the use of that phrase is used subjectively. Any non-asynchronous design will have jitter. So will any asynchronous design. Adaptive and asynchronous are two ways to do USB and each implementation varies greatly from manufacturer to manufacturer.

"Also, is the Benchmark design one of those that "Some listeners report as a Hi-Fi type of sound that is initially impressive, but long term listening may confirm otherwise." I've not noticed long term listening fatique, but I would like your opinion."

I'm sure some listeners of the DAC1 agree with that statement and many others do not. If you haven't noticed any fatigue then you're in a good position.

"Any non-asynchronous design will have jitter. So will any asynchronous design."

Yes, but well implemented asynchronous can have lower jitter than async ever can and just makes better sense.

The TAS1020B implementation, as in the Benchmark, does have some severe jitter at the TAS output. This you can measure at the chip output itself, or you can read the CEntrance implementation.

But at the TAS output, the Benchmark uses an ASRC (asynchronous sample rate converter) with a fixed crystal oscillator on the output to re calculate the digital data at a different and fixed sample rate.

With this process, you do really remove the jitter, because you create a totally new digital data stream, but the drawback is that this output stream is far from bit true for ever.

So some people like this method of ASRC because they get very low jitter and are highly immune against input jitter, but they are also some people, that couldn’t live with the sort of clean, tending to sterile sound of ASRCs devices. So it is up to everyone’s preferences.

I hope this helps a little bit for clarification and I am sure, if you want to need more detailed information, the Benchmark team will help you out.

Juergen

Thanks to both Chris and Juergen for your quick replies to my Newbe question. As I look through your CA Academy, Chris, I am begining to undersand more and more, but, in the process come up with more and more questions, as I try to decide on how best to move into a CA system. I hope to get some advice on that topic, but I should probably post such questions on a different part of the forum.

Juergen, in your reply, you state: "With this process, [using a asyncrhonous sample rate converter], you do really remove the jitter, because you create a totally new digital data stream, but the drawback is that this output stream is far from bit true for ever."

Are you saying that any time you convert to a different sample rate, (even if your upconverting), that you loose the status of bit perfect? If so, this makes sense to me, but I would think the converted bit stream would be as good a representation of the music as the original, (again as long as your upconverting). In other words, no information would be lost so why wouldn't the result be just as good? Or, am I missing something? Also, do you have a theory as to why ASRC devices should sound clean and sterile to some while Asynchronous USB converters, presumably, would not?

This is a great forum. Thanks in advance for your reply

Paugust

I will try to make it very simple and easy, to explain the difference between oversampling (integer number) and up sampling (fixed output rate). I know this explanation will be easy, so I hope that not too many will chime in and try to explain in a more detailed way, what was wrong with my very simple post.

A lot of, or nearly every DA converter, does oversample the signal, in order to get rid of the alias frequency and have to use only a very soft analog output filter. With this method, in the first digital filter stage, the original data still remains, and there is “only” added some calculated signals between the original data, depending on the used digital filter. So still the original data are mostly there as an “anchor”.

With an up sampling, (ASRC) to a fixed output rate, you take the original data and calculate a complete new set of data. With this method you loose your “anchor” in the audio band. Sure there are big differences between different ASRCs and the newer once do measure and sound better than the older ones, but the point that you do calculate all data totally new, and have no original anchor points at the output, does makes the difference.

Juergen

PS: All DSP guys, please be patient with my answer given to a “Newbe”. I do not want to write an DSP compendium. Thank you.

Juergen,

Another thing to note with an ARSC is that these devices are usually between the system receiver (be it SPDIF, USB, Firewire whatever) and the DAC chip. Many of these do complex fixed math with only a 24 bit result. Meaning a lot of the low level information is getting thrown out.

With the oversampling filters inside the dac, which most of the math is the same it is a little different as the dac processing can use a wider word to output the data without losing the information in the math.

Most of this math is pretty simple table driven multiplication, summation stuff but you have to remember that it has to be done very quickly so a lot of the time the precision is dumped for faster processing.

Thanks
Gordon

Gordon, are you saying dithered 24-bit DSP will throw away audio information that was originally present in a dithered 16 or 24-bit recording?

Can you name a single A/D or D/A that gives true 24-bit performance? The highest performing (32-bit) chip that I know of will actually only acheive about 21 bits of performance (dynamic range). So, with a properly dithered 24-bit ASRC, I can't understand how "a lot of the low level information is getting thrown out." Even if it was 32-bit ASRC....or 64-bits for that matter, no D/A chip on the planet will have sufficient dynamic range to maintain that low level information. Also, the best dynamic range from an A/D will max at 21-bits...so if the original recording didn't have 24-bit performance, why would a 24-bit ASRC throw out "a lot of the low level information?"

All the best,
Elias

From Computer Audiophile: "I think even Benchmark would agree that it's impossible to completely eliminate jitter and the use of that phrase is used subjectively. "

I'm not sure I understand what you are saying here. Why is it impossible to completely eliminate jitter? At what stage? At the USB receiver? At the DAC chip?

Best,
Elias