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DAC USB conundrum

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Posted on October 23, 2009 at 11:01:27
dcolver
DACs from dCS and Ayre are getting favourable press, among other reasons because they have USB inputs from which they clock the incoming signal proactively, rather than taking whatever the connected computer gives them. The coverage would suggest that this is the One True Way, and must assuredly give better results than crappy old S/PDIF.

One of the few DACs that gets even more ecstatic reviews than the two just mentioned is the Berkeley Alpha. It hasn't got a USB connector at all, but just the vanilla connectors of old.

How can I reconcile these positions? If asynchronous USB is indispensible, then mustn't any DAC without it be inferior?

RE: DAC USB conundrum, posted on October 27, 2009 at 11:04:48
dcolver
My original question wasn't so much aimed at the technicalities of USB, but at understanding why the Berkeley Alpha is so admired when it hasn't got the things that recent articles seem to rate as important innovations.
- One is clock-controlled USB, as discussed
- Another I now add is an apodising filter, which the Ayre has and so has Meridian, which also sounds like a Good Thing when described in the magazines.

The answer I was fishing for was something along the lines of
- the Berkeley reclocks everything internally so it's immune to the vagaries of itss inputs
- the Berkeley has an xyz filter with [list of magic properties].

Since the Ayre has limited distribution in the UK, and the Berkeley has none, one can't audition these things at a dealer. I will resort to ordering one of each of them so I can make the comparison at my leisure, since my long beloved Levinson 39 has expired. Luckily these things are inexpensive compared with prices for equivalent equipment a decade ago.

RE: DAC USB conundrum, posted on November 3, 2009 at 18:14:34
cfmsp
Audiophile

Posts: 353
Joined: October 21, 2006

The Berkeley Alpha is a very fine sounding DAC, by all accounts. But, perhaps you are forgetting the Analog in D-A-C. The Berkeley shines in this department.

The Alpha and similar AES/EBU & Coax S/PDIF DACs are a 'legacy' of the Transport to DAC era of digital audio. They do not connect to most modern computers WITHOUT the additional expense & complexity of a sound card or converter (e.g. Firewire to AES/EBU). The Berkeley recommended input is a $700-800 Lynx card - just to get the signal out of the computer (which probably cost less than the card itself).

In my mind, these legacy DACs are a dying breed, given the theoretical superiority of Async Firewire and Async USB. The Berkely is hanging on still due to it being a world class example of AES/EBU DAC.

I listen to a Metric Halo ULN-2, a Firewire DAC with a very high quality clock used to control the flow of data asynchronously. Metric Halo have introduced this year a world class Firewire DAC (as part of a pro audio device that also provides ADC, mic pres, etc.) called the ULN-8.

From accounts of those who have compared the ULN-8 and the Alpha, the ULN-8 comes out on top, and by a significant margin. It costs $6k, versus the Alpha's $5k, but when you add the cost of the Lynx card, the costs are nearly identical.

Supposedly the ULN-8 even bests the legendary Pacific Microsonics Model Two, the now discontinued pro DAC on which the lesser preforming (at lesser cost) ALpha is based.

If you are considering the $6k investment in an Alpha, you owe it to yourself to audition the ULN-8 if possible.

best of luck in your search,
clay



I know this thread stated out with USB..., posted on October 26, 2009 at 08:36:59
clio09
Audiophile

Posts: 1028
Location: Las Vegas, NV
Joined: January 29, 2006
but in the process we've had some discussion on firewire and ethernet. While I find this all very interesting and am learning a lot, the question I have is how does wireless fit into all this. After all, it eliminates a cable in the process of allowing computer generated audio.

Someone contributing to this thread mentioned the following in an article:

"Networked audio (Ethernet), both wired and Wi-Fi is a unique case. Because the data is transmitted in packets with flow-control and re-try for errors with buffering at the end-point device, it is not as much of a real-time transfer as USB, S/PDIF or FireWire. Networking also avoids the use of the audio stack of the computer audio system since it treats all data essentially the same. Because of the packet-transfer protocol of Ethernet and data buffering at the end-point, the jitter of the clock in the computer is a non-issue. The only clock that is important is the one in the end-point device. Examples of end-point devices are: Squeezebox, Duet and Sonos. This would seem to be the ideal situation, which it certainly is. The only problem that can occur is overloading the network with traffic or Wi-Fi interference, which may cause occasional dropouts. The problem for audiophiles is that the majority of these devices were designed with high-volume manufacturing and low-cost as requirements, with performance taking a lower priority. As a result, the jitter from these devices is higher than it could be. It should be the lowest of all the audio source devices available."

A good summary in my opinion of the positives and negatives, but the last sentence seems to say it all as far as best possible format.

I know a few people using modded Transporters and a unit called the Zardoz Ultimo who are quite happy, as some here are with USB anf firewire, but I have to wonder why we don't see more audiophile grade wireless solutions. I realize R&D is a big part of the equation, but that is the case with anything in high-end audio, even Gordon's asynch USB work.

Network Audio without Computer and Asynchronous DACs, posted on October 31, 2009 at 04:34:16
cawson@onetel.com
Audiophile

Posts: 380
Joined: September 27, 2004
Yes. you're right, transferring raw files by network has theoretical advantages over USB or Firewire, but I see no reason to do this wirelessly if you want maximum performance - I prefer to hard wire with standard Cat 5 or Cat 6 Ethernet cable.

As you rightly say, existing receiving processors (Sonos Squeezbox, etc) are simply not good enough. You mention Zardoz Ultimo, but I can find nothing about this, athough it seems to be wireless - no thanks!

There's also the PS Audio PerfectWave DAC with Network Bridge. This seems to be a far more elegant and potentially higher quality solution than any of the existing processors or any of new USB DACs.

I'm sure PS Audio won't mind if I quote a few paragraphs from their latest Newsletter.

"Asynchronous USB
……. As to the question which is better, the answer’s pretty clear: asynchronous. Asynchronous clocks are what’s used in the PerfectWave transport and as long as you implement it properly, it’s clearly a better way to go when streaming digital audio data. No question about it.

So, our customers are asking me why we don’t switch over to asynchronous USB for the PerfectWave DAC and the answer’s really simple: there’s really no reason. USB is hopelessly limited to lower sample rate audio and it must always be tethered to a computer.

I don’t know about you, but for my money these two hurdles are simply too much to get in the way. It’s kind of like asking whether coax or balanced is better on S/PDIF - the answer is balanced, but who cares? I want to listen to real high-resolution audio, which for me starts at 176.4kHz and I don’t want a computer anywhere near my listening area. Certainly 96kHz is better than 44kHz but it’s a short-term answer.

The best solution is just around the corner and it’s called the PS Network Bridge. The Bridge will connect your computer or storage via your network and stream any sample or bit rate completely asynchronous; delivered to the PWD via I2S without any jitter or problems. You don’t even need a computer actually, all you’ll need is an online storage device (like a NAS) which can be placed anywhere in the house you like. It’s that simple and elegant and it can even be wireless if you want."

OK, they are biased, but is there anything wrong in their argument that Network transmission of audio files is far superior than UBS, however well it's implemented?

Any comments from the asynchronous USB proponents would be appreciated, as I think this is the way to go unless I’m convinced otherwise.

Peter

Full article below - I have no connection with PS Audio, or anyone else for that matter.

Why USB and Firewire are so much better off than SPDIF, posted on October 26, 2009 at 05:08:46
Gordon Rankin
Manufacturer

Posts: 2247
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Gang,

USB and Firewire are specifications which were derived by a large diverse group of engineers. They looked at the applications, the needs and drew up a huge specification for which many things work by it.

SPDIF was adopted as a protocol years after it was invented by Philips.

Let's recall how SPDIF was developed as I remember meeting the Quality Control Engineer who came up with it. The Quality Control division of Philips was asked how they could test a CD player in what is called closed box. Meaning basically that it was a finished product. The QC department came up with a test disk and a jig with IR command base and a three cable connection for analog R & L and a new connector which was a digital connection with what was later termed S/PDIF. How Sony got first billing on the name I do not know. Anyways the tester would put in the three cables the test disk and hit go and the IR commands would issue tracks and the Digital and Analog were evaluated and the unit would be tested.

Only years later did the IEC adopt this as a protocol.

Just think what would have happened if say 20 or so companies would have gotten together and weighed in on SPDIF.

It probably would be an Asynchronous based system of some kind because that is were the performance can best be derived.

So gang we are basically stuck with SPDIF as it cannot change now. But the cool thing about Firewire and USB is that it is changing for the better everyday and there are specifications to back up all kinds of connectivity and that is good for any industry.

Thanks
Gordon
J. Gordon Rankin

Point taken. Thanks, GR ~nt, posted on October 27, 2009 at 06:37:42
Feanor
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Posts: 2931
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Contributor
  Since:
March 12, 2004
nt
___
My Stereo Configuration
Feanor's Classical Music Survey, 250 Compositions

RE: DAC USB conundrum, posted on October 25, 2009 at 11:10:02
audioengr
Manufacturer

Posts: 4088
Location: Oregon
Joined: April 12, 2001
If async is implemented well and uses a good clock, it is a great solution, as is async Firewire. It has the potential to produce a stream with jitter as low as a free-running reclocker. The current reality seems to be something less however. It is also not a panacea for pops and clicks and stalling, as many though it would be. Steps usually have to be taken to avoid these. Usually requires a fast CPU, newer OS, low-latency disk access, big memory and a new USB2.0 port to work without issues.

It is possible to implement an Adaptive solution that is just as good, cheaper and more flexible. The async solution has the potential to be a bit better, but at much higher cost because it is more complicated and requires two good clocks, not one. With all audio designs, it is the design, implementation and parts selection that makes it sound good.

The problem with most async designs is the clock quality and the design/implementation. Many are still sensitive to the USB cable that you use, which does not make any sense to me. Some are even powered from the USB cable. Some dont work with Snow Leopard yet.

I have had good luck with the Fireface400 async Firewire interface. I have had a lot of problems with the Tascam US-144 async USB interface. I have had no problems with the CEntrance Adaptive USB interface, even with older computers. The good news it that unlike some async interfaces, the Tascam seems to be pretty much immune to incoming jitter, so you can use a really cheap USB cable. Dont expect the music to be playing in the morning if you start it at night though. It usually crashes both iTunes and Amarra eventually. The USB cable quality and length can also affect stability. I have a very good USB cable that is 16-feet long, the Polestar. Sounds great, but will not work at all with the Tascam. It seems to work fairly reliably with a 3-foot USB cable, but not overnight.

What matters here is the sound quality and the stability, not the technology that gets you there.

I'm not saying that all USB or Firewire interfaces are equal. Far from it. Both the technology and the implementation are critical. Check into these and get feedbacks from happy customers before making a decision. Plug-and-play USB chips that use Adaptive mode usually have the worst jitter. Sometimes the best solution is to get the DAC that you like the sound of first and then add the interface later, either a reclocker or USB/Firewire outboard interface. Contrary to what some will tell you, adding the interface externally can be just as good as having it internal. Again, depends on the implementation.

Steve N.

RE: DAC USB conundrum, posted on October 25, 2009 at 14:40:06
Dynaudio_Rules
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Posts: 5594
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I have to be honest....I used to be a big time skeptic for clocking. That is until I heard it with my own ears.



Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

RE: DAC USB conundrum, posted on October 25, 2009 at 23:29:12
fmak
Audiophile

Posts: 4099
Joined: June 1, 2002
I agree with audioengr and your observations. What is the issue are usb async makers ignoring what has been happening in non PC audio for years and reinventing another kind of wheel.

10-15 years ago, Meridian made the 518 processors (relocker plus mastering features and this simply strips the incoming clock and polishes up the incoming spdif stream. It sounded very good, got excellent reviews and was used by many pros. The Digital Lens is another example and is still used by many consumers. The problem is that they only do <= 48k.

I have tried at least half a dozen 'super low jitter' clocks plus boxes that relock, and I can say that none of them sound the same. The issue is:

What is a low jitter clock and how does it interact with the power supply? Makers claim ridiculous nos like 2 pS jitter; others claim low phase noise and 1/f nos but none of them can actually nail the basis of what they say to sound quality. Not surprising since SQ is a subjective 'index'.

As I have posted before, it is a case of optimising the entire signal path to preserve signal quality ie impedance matching, cabling, connectors, clocking, wave symmetry and lack of oscillations etc., that matters in SQ terms.

The dCS 972/954 is the only box I have measured that have excellent wave integrity within and out of the boxes (sqaure waves are just that with very small leading edge overshoots. Most PC digital outputs are just well, rubbish.

DSD on sdif 2 also seems to preserve wave integrity much better than PCM, and sounds very good with DSD output players.

RE: DAC USB conundrum, posted on October 25, 2009 at 22:23:55
Dawnrazor
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Posts: 7499
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Hey Dyn,

No expert here on clocking but i think Steve is talking about reclocking and you are talking about using an external clock.

Arent they a bit different?


RE: DAC USB conundrum, posted on October 26, 2009 at 11:29:18
audioengr
Manufacturer

Posts: 4088
Location: Oregon
Joined: April 12, 2001
It is possible to have an async interface driving a memory-based reclocker or just a one-stage reclocker. Typically you have one or the other. The term reclocker has been used for all types of clocking, so it can be misleading. In this case, its really just a synchronous-logic clocking stage. All clocks are synchronized.

Reclockers can also be used for asynchronous streams, where the stream is free-running and the reclocker changes its frequency to track it. This has historically been done with analog or digital PLL's, but there are more effective ways to do this now for audio.

Steve N.

Firewire Async not supported by any os, posted on October 25, 2009 at 12:03:41
Gordon Rankin
Manufacturer

Posts: 2247
Joined: June 9, 2000
Steve,

There is no implementation of Firewire async as no operating system supports it.

Instead what companies do as was described by John below is that they implement BULK mode and then write drivers to make it look like an audio device. I am not saying this is wrong... many PRO companies do this and it can be done with excellent results.

I am just saying the OS's do not support this natively.

Thanks
Gordon
J. Gordon Rankin

Firewire Async, posted on October 25, 2009 at 07:35:43
Panelhead
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Posts: 430
Location: Houston
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Guess I assumed all good quality FW dacs used Async. Since the Apogee dacs all have clocks and drivers, are they all FW Async?
I ask because I have an Apogee Duet. It is an amazing dac, I am yet to use the mic pres, and ADC capabilities. I would love to compare a 24/96 USB Async to it.
Like has been posted, it is all in the details.

George

RE: DAC USB conundrum, posted on October 23, 2009 at 11:55:38
fmak
Audiophile

Posts: 4099
Joined: June 1, 2002
What you say about the usb unput for the dCS is not true. That input is limited to 96k and is as good as the other inputs, not better. The Ayre, of course is 96k usb only.

Hirez at >96k sounds a lot better to me. Approaches analog. 96k is not there.

It is my understanding, based on my experience, posted on October 23, 2009 at 11:49:32
Sordidman
Dealer

Posts: 9245
Location: San Francisco
Joined: May 14, 2001
that the USB Audio CODEC comes with high levels of jitter and needs to be re-clocked.

So, a device like Empirical Audio's conversion to SPDIF or I2s in the Off-Ramp 3; or an internal, asynchronous DAC like Ayre and Wavelength, are much more better than straight on in USB. I've used my APL, and the Off Ramp 3 and am happy with the performance of both. I would love to compare an Ayre or Wavelength to what I have.

Cheers,



Surrendered to self preservation,
From others who care for themselves.
A blindness that touches perfection,
But hurts just like anything else.

Marketing, posted on October 23, 2009 at 11:23:33
Dynaudio_Rules
Audiophile

Posts: 5594
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Nothing but marketing BS.

I2S is better than USB and so is Firewire. But I have yet to hear either of the three better the performance of a XLR digital connection.

However the placebo affect goes a long way in the audio world and people tend to buy based on momentum and world of mouth...eg Benchmark DAC's, and the current USB MusicStreamer are examples.

The MusicStreamer uses asynch just like Ayre and the other high cost DAC's, but apparently they can sell the same technology for a fraction of the cost. I would not doubt it if one of the mainstream audio companies were behind the Musicstreamer buying the hardware in bulk and selling it in a basic product to recover some costs on volume [low profit margin]

Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

Some thoughts on async etc., posted on October 24, 2009 at 15:23:20
John Swenson
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Posts: 2013
Location: No. California
Joined: October 13, 2002
I'm going to take a stab at this and hopefully do it in a nice even handed way.

Lets tackle the USB/I2S point first.

There is a conceptual difference here, USB is a computer bus, it is a way to get data in and out of a computer, so is firewire, I2S is not. Processors do not have an I2S "bus" to talk to the outside world. I2S is an interface designed to go between chips on a board. It CAN be used to go between boxes with appropriate drivers and connectors, but you still need some way to connect to the computer, something has to get the data out of the computer and generate the timing signals going over the I2S interface.

Currently there are three methods in use for getting the data out of the computer, PCI card, USB and firewire. A DAC with an I2S or S/PDIF interface (in any of its physical transport flavors) must still get its data and timing from something connected to the computer through one of these methods. A DAC connected to an I2S interface generated by a box connected to a USB bus is not going to be any better than a DAC that had that USB interface builtin.

From theoretical grounds the best way to get the computer data to a DAC chip is to have low jitter oscillators right next to the DAC chip generating the timing for the chip and that clock also being used to control the rate at which data comes from the computer so it matches the rate at which data goes into the DAC chips.

In traditional unidirectional S/PDIF, AES/EBU and I2S this cannot happen since there is no way for the DAC to control the rate of data from the computer. The DAC has to somehow deal with the data rate coming out of whatever generated that timing. All the many varied methods that have been invented to deal with this are essentially band aids used to deal with the non-ideal topology.

So how DO you do it right?

A PCI card can get data off the PCI bus at whatever speed it wants so it seems an ideal way to do things right. But that means putting the whole ball of wax on a PCI card, sitting inside of a computer, usually powered by the computer's power supply. There are a lot of engineering hurdles to overcome to get really good sound with this method.

The PCI card can also be used just as a digital interface, but then you need some method for getting the data to the DAC box. If you use I2S, S/PDIF etc you still have the problem of the PCI card generating the timing not the DAC. To do it right you would need to have some method for getting timing data from the DAC to the PCI card. This method works great and is not that hard to implement, but for some reason has hardly ever been implemented commercially. (I have done this myself and it DOES work great)

Both USB and firewire have both adaptive and asynchronous modes. In the adaptive modes the DAC has to adapt to the data rate coming over the bus, in async the DAC controls the timing. The async mode is exactly what we want, the low jitter clock can be right next to the DAC chips and no band aids are needed. (BTW firewire is no different in this respect than USB, most implementations use adaptive mode and a few use async. The assumption some people have that just because firewire is being used it is automatically async is not true)

Async USB is not just marketing BS, if implemented correctly with very low jitter local clocks its one of the best methods for achieving extremely low jitter at the DAC chips.

The problem is there are no off the shelf USB audio async chips available. There a few that theoretically can do async mode but they have to be programmed to do so. The manufacturers do NOT provide the programming to do this. This is what Gordon did, he spent almost two years writing the code to implement async mode on one of these chips. It is NOT easy!! Several years ago I tried this myself and gave up after 4 months of beating my head against a brick wall, Gordon stuck it out and was successful.

This effort he put in represents a huge investment on his part. As with any major R&D effort the payback for this has to be implemented by a slice of the price of the product sold to customers. The price of the actual hardware used to implement the async mode is quite low, its the intellectual property of the firmware running on that hardware where the value lies.

Then there is the rest of the DAC. One of these boxes is not just an async interface, its a high end DAC using high quality components. Most of the cost goes to other parts of the box, the DAC chips, I/V conversion, analog out stage and power supplies. I'm going to make a guess here that the reason for spending all the effort on the async interface was not to have some new marketing buzzword, but because he had figured out how to do a very good job on the DAC chips, analog stage, power supply etc and the limitations of the computer interface of existing solutions was the major bottleneck in achiveing exceptional sound. Going with async USB was one of the few methods to do it right and get jitter low enough to not hinder the rest of the design.

This is primarily why you see audio async only on expensive boxes, its hard to do and only people who are doing the rest of the equation very well are willing to pay the price to implement it properly.

There IS another way to do async USB, that is to throw away compatibility with the official audio async standard. It still takes significant firmware coding, but not nearly as bad as the official async mode. BUT you have to write custom drivers on the computer since you are in effect inventing your own custom USB protocol. If you go this route you have to be willing to write drivers for all the major platforms AND maintain them, provide upgrades when new OS's come out and provide support to end users. This is a lot of effort and most high end companies simply do not have the resources to implement this. The EMU 0404USB and that little new Musiland thingy have gone this route.

The HRT MusicStreamers (at least the ones anybody has seen) do NOT use an async interface, they use the adaptive mode like everybody else.

The one that is interesting is the Musiland thingy. It actually implements an async mode, but NOT the official one, so it requires a custom driver. They took an asynchronous bulk data mode implemented by the USB chip they use and wrote a custom driver to implement an audio interface for it. But then they messed up the rest of it by using a frequency synthesizer to generate the audio clocks rather than using very low jitter oscillators! They seem to have enough programmer resources to write the drivers, we shall see how well they do with multi-platform and long term support. They seem to be putting in the required effort for driver writing but are selling a very low priced product. We shall see whether this is economically viable.

The other thing I want to touch on is the AES/EBU digital interface (the XLR one). Any such interface needs to be able to pass high frequency digital signals, to do this right takes a broadband RF connector with a consistent impedance over a broad frequency range. The XLR connector is a LOUSY RF connector. There are many designs out there that are MUCH better RF connectors than the XLR and any of them would have been a better choice. As far as I can tell the only reason the XLR was chosen is that studios have miles of mic cables with XLR connectors. Because mic cables are lousy digital interconnects they chose to use a very high voltage level, 3-5 Volts so they would have some chance of having some signal left at the other end. (almost all other high speed digital interconnect systems use somewhere between 250mV and 500mV). Just think what it means when you put 5V onto a 110 ohm system, thats almost 50mA of current the driver has to provide, thats a huge amount of current switching very rapidly. That rapidly switching high current supplied to the driver chip is going to induce significant noise on the power supply traces and ground plane on the board. There is a high probability that noise is going to increase the jitter of the output signal.

Its one saving grace is that its balanced, if the input receiver is done right this can make it not quite as bad as it should be given the lousy composition of the rest of the system.

I'm not trying to say that the AES/EBU is always going to sound terrible, some companies have spent a lot of engineering talent trying to overcome its inherent liabilities. Its just that this interface could have been SOOOO much better if it had been designed on technical grounds rather than giving in to studios that wanted to use mic cables as digital interconnects.

I still think the best way to implement this whole getting audio data out of a computer is with a dedicated ethernet chip that can be timed by an external clock. Boxes such as the squeezebox are a step in the right direction, but still rely on computers themselves with all the issues of processor load etc. Its actually possible to implement something like the netjack protocol on a fairly simple custom chip that is actually doing far less processing than a USB receiver chip. My goal is to one of these days make such a chip.

There is more I want to share but this has gotten long enough so I'll stop now.

John S.

John thanks for the clarification, some more insight, posted on October 25, 2009 at 12:00:28
Gordon Rankin
Manufacturer

Posts: 2247
Joined: June 9, 2000
John,

Thanks for filling in what I often forget to!

Actually the Musicland and EMU are pretty different. The Musicland uses the Cypress dummy device driver and .NET wrapper to implement the Device interface. See Cypress developed a dummy device driver that will work with any of it's 2.0 controllers. They also give you the .NET code to implement what every interface you want. Then put the wrapper around the .NET to make it look like a sound card.

The one problem with using Bulk mode is that there is no guarantee of delivery because it has such a low priority. Whereas Isosynchronous has the highest priority and therefore best avenue for audio.

The EMU 0404 was developed by Markus Medau of Ploytec. Basically it uses his Windows driver and MAC driver but really it is a USB Class 2 device hidden and associated with some standard interfaces so if you don't use the device driver it will show up as a standard 24/96 device. Markus also did the Tascam piece and some other async units. We talk often about stuff like this.

I totally agree with your comments on the SPDIF XLR. I didn't say it would not work either, I just think there are better ways and connectors to do it on.

Thanks
Gordon
J. Gordon Rankin

RE: Bulk Mode, posted on October 26, 2009 at 09:37:13
fmak
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Posts: 4099
Joined: June 1, 2002
The Musiland has the most robust asio driver I have used in a usb device and not open to intereference. The 0404 is well, Creative; with a driver that corrupts quite easilly.

One Question, posted on October 25, 2009 at 09:33:15
Dynaudio_Rules
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Posts: 5594
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Jitter is not an "Audio Only" phenomena right....I mean there are some very large companies studying jitter and how it affects all sorts of data transmissions in every area.

Is it possible that USB asynch is the end-all-be-all in zero jitter data transmission?

If asynch USB is all that it claims to be I suspect we will be seeing it used in military, science, large scale data transmissions and communication not just consumer audio. The former markets being much larger than the latter.

Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

Not the only one, posted on October 25, 2009 at 12:05:35
Gordon Rankin
Manufacturer

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Dyna,

Async USB is not the end all... there are other ways to implement an ASYNC system in USB, Firewire maybe SATA who knows.

ASYNC USB just works well because the OS supports the drivers out of the box.

Thanks
Gordon
J. Gordon Rankin

Ok, posted on October 25, 2009 at 14:35:05
Dynaudio_Rules
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Posts: 5594
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Well this was a good discussion, I learned a lot.



Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

RE: One Question, posted on October 25, 2009 at 11:26:01
audioengr
Manufacturer

Posts: 4088
Location: Oregon
Joined: April 12, 2001
Jitter is not a "audio only" phenomena, but the difference is that for data transmission, all you care about is data integrity. This is a lot easier, unless you are transmitting for miles or at multi-GHz. For audio, it is real-time, so you care about modulation of the D/A. Even at relatively low frequency, jitter has a large effect in audio.

Fair Enough...nt, posted on October 25, 2009 at 14:32:34
Dynaudio_Rules
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Posts: 5594
Location: Georgia
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.

Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

RE: One Question, posted on October 25, 2009 at 10:55:42
rick_m
Audiophile

Posts: 2529
Location: Oregon
Joined: August 11, 2005
Maybe more questions?

I (blush) haven't bothered to look at the Specs, that would be your basic laziness showing up, but based upon what I've read on this very forum the main thing async USB does is allow you to avoid a PLL and it's attendant VCO or VCXO at the receiver end. Apparently you can remotely command the host to send the data at two (or more?) rates so that very little activity is required at the DAC to manage the buffer outside of keeping your eye on the FIFO pointer and toggling the rate if it's getting near it's limits. The input clock, I guess, still has to be derived from the data stream but in the perfect case will be completely isolated from the buffered data stream.

There are still numerous potential ways in which the source and DAC can interact and produce jitter but from what I've read here (and experienced in other contexts), keeping the error loop in a VCO clean is harder than keeping a politician honest so eliminating it is a major topological improvement.

However at the least you will still have the intrinsic jitter of the DAC's clock and probably various other terms such as ground bounce from the input clocking, noise on the power supply, and feedback between the analog output buffer and the D/A clock slicer.

Nothing's perfect, count on it. A better solution would be to have an "audiophile" computer which would receive a clock from the external DAC and then generate all it's internal clocks with PLL's using that as the master reference. If the DAC clock went away it would resort to it's own crystal. I'm sure Hp and Apple are hotly working on it as I type :-).

Rick

RE: Some thoughts on async etc.-a little hype, posted on October 24, 2009 at 22:24:14
fmak
Audiophile

Posts: 4099
Joined: June 1, 2002
In one respect, the Async usb thing is marketing hype, and it is that it is new, unque and superior to other interfaces. The theoretical stuff is fine, but is it bourne out in practice?

The hype is:

1. it is new and unique (but Firewire Async interfaces have been there for yonks and people like RME and the pro audio companies have been using it for sometime.

2. it is superior to aes3 in theory and practice (it is not from experience all the reviews of high end systems)

3. it does not have hirez restrictions and that 24/96 is 'satifactory' for high end/high quality reproduction (it is not, from 5 years of listening to hirez material using both high end, home made, and pro quality dacs, processors and computers)

4. as an aside, usb processors do not seem to like sharing usb ports with HDDs and other things and so is more restrictive in operation.

Try it for yourself in a well sorted system, cut off the Vbus supply to a usb HDD or dac, and you hear a marked hf and sound stage improvement (provided that you supply the device externally and properly). It is the same but perhaps less so with Firewire. This is indicative of the poor quality of supply rails in motherboards having a bearing on reproduction quality and no amount of theorising can avoid that. Improve the supply to/in a pci sound card and it is the same story.

For me, bang goes hype.

RE: Some thoughts on async etc.-a little hype, posted on October 26, 2009 at 00:01:41
Old Listener
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Posts: 693
Location: SF Bay area
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You are on your hi-res hobby horse again riding through fantasy land.

> 1. it is new and unique ...
> 3. it does not have hirez restrictions and that 24/96 is 'satifactory'

Both Gordon and Charles have been quite clear in their technical explanations. Gordon has been clear that the EMU 0404 USB also operates in an async mode with their own drivers. His advance was to implement firmware that allows his DACs to work in async mode with the drivers.

Gordon didn't claim to have invented async mode; that was in the USB spec.

They have been quite forthright in saying that the current products are limited to 96 KHz. That allows the consumer to make his own choices.

> 2. it is superior to aes3 in theory and practice
> (it is not from experience all the reviews of high end systems)

Many, many people have pointed out technical issues with SPDIF transmission. the pluses and minuses of using a master clock or sending the DACs own clock signal back to the soundcard have been discussed many times by far mor epeople than just Gordon.

I don't recall an single mention of aes3 in a post by Gordon or Charles. By the way, do you have a list of motherbaords and soundcards that support aes3 transmission and DACs that support aes3 reception?

reviews with measurements are appearing so consumers can read them and see how actual async mode DACs perform.

> 4. as an aside, usb processors do not seem to like sharing usb ports
> with HDDs and other things and so is more restrictive in operation.

This has nothing to do with whether discussions of async mode USB DACs have included hype. A prospective customer for a DAC needs to sort out audiophile urban legend and find the facts and apply them to his own situation.

Give your poor, tired hobby horse some rest.

Bill

RE: Some thoughts on async etc.-a little hype, posted on October 26, 2009 at 04:46:50
fmak
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Posts: 4099
Joined: June 1, 2002
You clearly do not know about aes3.

If I were Gordon/Charles, I would be annoyed by your irrelevant assertions on the issues in discussion. Are you there surrogate?

Seems to be another of your complexes?

RE: Some thoughts on async etc.-a little hype, posted on October 26, 2009 at 09:08:24
Old Listener
Audiophile

Posts: 693
Location: SF Bay area
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> You clearly do not know about aes3.

You are right, I made a mistake. I thought you meant something beyond aes/ebu. Nothing special or wonderful about plain old AES/EBU.

The technical issues around spdif or the 110 ohm AES/EBU version have been discussed many times by many people. (Even by you.) Gordon or Charles or John did invent the issues with impedence matching on the connenctor, signal reflection in the cable or the difficulty of extracting a very low jitter clock signal from the incoming stream.

> If I were Gordon/Charles, I would be annoyed by your irrelevant
> assertions on the issues in discussion. Are you there surrogate?

I'm not a shill for anybody. I get really tired of your mis- characterizations of what other people say.

I'd like to see more good technical information about all approaches to PC audio.

USB audio is not your cup of tea. That doesn't mean that any discussion of its advantages or the technical issues should not be discussed on this forum. An honest technical discussion of an approach you don't endorse is not automatically hype.

> Seems to be another of your complexes?

Call it what you like. Your inventing descriptions of my posts doesn't make them real.

I'm just reacting to your stream of mean-spirited, attacking posts.

For a long time, your forum posts were quite unpleasant and not very accurate. You seemed to cool off for awhile. Lately you are back to attacking everything and everybody you don't agree with.

Bill

RE: Some thoughts on async etc.-a little hype, posted on October 26, 2009 at 11:23:39
fmak
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Posts: 4099
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And you think that Gordon is polite, accurate and dispassionate?

It's a complex.

You don't hear much here about Firewire having async and adaptive flavors, posted on October 24, 2009 at 19:30:18
keith_d
Audiophile

Posts: 931
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I suppose we're next going to have a long thread on whether or not the relatively cheeeep Apogee Firewire DACs are no better than comparably priced adaptive USB DACs, right?

Or maybe even whether or not the expensive Weiss Firewire DACs truly keep jitter low as they claim....

RE: You don't hear much here about Firewire having async and adaptive flavors, posted on October 24, 2009 at 23:05:14
John Swenson
Audiophile

Posts: 2013
Location: No. California
Joined: October 13, 2002
I've had three firewire devices over the years (I'm not listing them here) and every one of them has had abysmal jitter, the cheap little 2706 USB DACs I made had much lower jitter! So firewire is not an automatic panacea.

As with just about everything else its all up to the implementation. There is even one very highly praised firwire box which DOES use the async mode, but then uses a very crappy local clock (very similar to what the Musiland does with USB). I just don't get companies that do that.

John S.

RE: Just strip the clock and regenerate, posted on October 25, 2009 at 04:30:41
fmak
Audiophile

Posts: 4099
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Whay's the hassle; why do we nesd to buy into another format and another set of problems?

This is where the hype lies; commercial interest.

RE: Just strip the clock and regenerate, posted on October 25, 2009 at 15:57:19
John Swenson
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Posts: 2013
Location: No. California
Joined: October 13, 2002
But how do you regenerate a clock and achieve jitter as low as a good low jitter fixed frequency oscillator?

John S.

RE: Just strip the clock and regenerate, posted on October 25, 2009 at 23:41:27
fmak
Audiophile

Posts: 4099
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There are many excellent examples going back 20 years. These have been ignored in the search for new PC solutions (commercial hype?)

We start with the Digital Lens and Meridian 518. We have had many solutions since by high/mid end companies as Charles indicated.

Please read my response to audioengr at the top of the thread.

There is also no such thing as a 'best' low jitter clock. Even some of the ones with very low claimed phase noise have very poor waveform outputs (shape, symmetry and band limitations) and don't sound very 'spectacular'. They can cost as much as a reasonable dac as well, relying on hype to market them within a boxed solution.

Ironically. many of the people who make 'better and better' clocked solutions appear to listen to rbcd or at best 96k material.

How can one claim to make best 'audio' solution PC product using sub standard and not very good sounding music software ie RBCD with massive phase anomalies and high low level distortions?

RE: Just strip the clock and regenerate, posted on October 26, 2009 at 11:38:53
audioengr
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Posts: 4088
Location: Oregon
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"There are many excellent examples going back 20 years. These have been ignored in the search for new PC solutions (commercial hype?)"

I think you are a bit out of touch. These have not been ignored, they have been improved upon.

Steve N.

RE: Just strip the clock and regenerate-Clarification, posted on October 26, 2009 at 11:51:07
fmak
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Ignored by some posters in this Board (not others). Of course there has been progress over 20 years.

Fred a fixed clock will always kick the crap out of variable one., posted on October 26, 2009 at 05:16:01
Gordon Rankin
Manufacturer

Posts: 2247
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Fred,

Any clock that is variable is in a sense adding jitter. Not to mention that is so much easier to make a fixed oscillator with low jitter as compared to a moving one.

Take a look at some of the great discrete VCXO's and they are lucky to get below 10ps of jitter which would equate to at least 158ps of jitter on the Word Clock.

Heck you can get fixed clocks for less than a $1 that have really good jitter below 10hz that are at least 10x better than any VCXO.

If you talk frequency synthesizers then they are even worse... but remember a lot of the best VCXO's don't have the pull range required for audio.

Thanks
Gordon
J. Gordon Rankin

RE: Fred a fixed clock will always kick the crap out of variable one., posted on October 26, 2009 at 07:50:31
fmak
Audiophile

Posts: 4099
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When did I say VCXO? Some have multiple XOs.

Why have ultra low jitter spec and then use a tube and trandormer output with high distortion?

Excellent post! Thank you and one quibble., posted on October 24, 2009 at 19:25:47
Charles Hansen
Manufacturer

Posts: 4320
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For people that don't know, John Swenson is a super-knowledgeable designer that has been the "hired gun" behind many well respected high-end products. He is conversant with both digital and analog (very few engineers are) and he knows what he is talking about. Plus he has no "skin in the game", so his statements are completely unbiased.

The one place where I would disagree would be with John's post is that Ethernet is the best solution.

It can be a great solution, and provide jitter levels as low as either async USB or async FireWire. So that part is great. The problem is with software.

As John pointed out, async FireWire requires custom device drivers for all operating systems and all hardware platforms. But Ethernet is even a bigger can of worms. It requires a complete custom music player for all operating systems and all operating platforms. To date only two companies have tried. Linn has gotten mixed reviews. Squeezebox has done much, much better (and by making it open source, ensuring that it will run on any OS) but even that application lacks a CD ripper.

In contrast, an async USB DAC can use any music play software. Just open up the Control Panel (or the Mac equivalent) and send the output to the USB DAC.

John, if I am missing something, please feel free to set me straight.

RE: Excellent post! Thank you and one quibble., posted on October 24, 2009 at 22:57:35
John Swenson
Audiophile

Posts: 2013
Location: No. California
Joined: October 13, 2002
Hi Charles,
unfortunately nothing I have designed has actually made it to production yet. Hopefully the Bottlehead DAC will be the first.

On the ethernet front the system I am envisioning uses the netjack protocol which uses UDP, which is a fairly simple IP protocol which can be implemented fairly easily.

You run a program on your windows PC, mac, or linux box which is a JACK server that sends its output to the network interface using the netjack protocol. The Jack server creates a virtual soundcard on the machine which looks like an ASIO driver on a windows machine and an ALSA driver on a linux box. I'm not exactly sure what it looks like on a MAC since I don't have any macs. The advantage of this is you can use any software on the computer that you want, any GUI, file types, database, whatever and its completely isolated from the actual audio hardware. You can do whatever you want on the computer, fast, slow, hard drives, SSD, big screens little screens, small memory, big memory, all kinds of remotes and it doesn't make any difference to the sound!!! (as long as your software doesn't muck with the bits).

This system can be run today using the free JACK software, but the audio end (the master, it talks to the actual audio hardware) has to run under some OS (I prefer low power dedicated linux boxes). BUT the protocol is simple enough that there is no reason it cannot be implemented directly in hardware on an FPGA. I'm not even talking an embeded processor on the FPGA, I'm talking direct gates implementing the protocol where everything runs in parallel, no realtime kernel needed, no low latency interrupts. Internet plugs in one side and I2S comes out the other with a clock line going in which comes from the low jitter oscillators. (there will be some control signals such as the sample rate etc). I call this the netjack chip. Its inexpensive enough you can put it in DACs, self powered speakers, whatever. (headphones?)

This is actually quite doable with todays technology, there already exists an FPGA core which implements the UDP protocol as gates, it does evertything I want except it costs $9000 to license the core, which is what is keeping me from doing this right now. The netjack protocol itself is quite simple to implement directly on an FPGA.

The one problem with this is you DO have to run the netjack slave software on the computer running the player, fortunately the JACK group is writing that! Once you get it running you don't have to do anything with it, it just sits there in the background and routes audio streams around. Right now the netjack protocol is undergoing some significant changes so its probably not a great idea to make a product out of this just yet.

I have been running the software version of this for a while now and it works very well, it really is true that this completely decouples what is happening on the "playing computer" from the sound hardware. Currently you have to choose a sample when you invoke the master, but this is a limitation of the jack server being used. With the netjack chip there would be no such restriction.

John S.

RE: my netjack experiences (Ryelands, are you listening?), posted on October 27, 2009 at 09:26:40
aljordan
Audiophile

Posts: 819
Location: Southern Maine
Joined: November 4, 2003
Hi,

I have been listening to a Netjack based system over the past month. I had actually first tried Netjack about six months ago but I did not know of an easy way to remotely choose music at the time so I let it go.

Why would someone want to use Netjack? Similar to Squeezecenter, it allows a person to keep a very small, headless device (no keyboard, mouse, nor monitor) in the listening room, without having to have the music library stored on large local hard drives. There are other means of doing this, such as Squeezecenter streaming to a Squeezebox, Squeezecenter streaming to a PC running Squeezeslave, or Music Player Daemon accessing a library on networked remote shares. Squeezecenter / Squeezebox devices are limited to a max 96k sampling rate. Netjack and Music Player Daemon are limited by the sampling rate of your DAC. Music Player Daemon is only stable on Linux. Netjack works on Linux, Windows, and I think on a MAC. Another benefit, possibly only applicable to a low processing power player like mine, is that I keep my library in FLAC format, and the Netjacked server will take care of decompressing FLAC back to WAV before it is sent across the network to the player. Squeezecenter server also has the ability to decompress FLAC on the server before it is sent across the network to the player, and doing so has a positive effect on the Squeezebox clients, but the server has to be specifically configured to do this.

My Netjack player is implemented as a Fit-PC Slim with a small SSD running a stripped down version of Linux. The Fit-PC feeds either an asynch Wavelength USB DAC or an asynch E-mu USB DAC. I will eventually try running a Netjack player on my networked CMP-like box feeding either the above USB DACs or a Lynx 2B sound card. I've implemented the Netjack server on both Linux and Windows.

What I've gleaned so far is that a player running on a server feeding Netjack makes a positive difference compared to MPD running on the player accessing a file-system over a mapped network drive. Given that the Fit-PC Slim is a "processing-power-challenged" device, its possible that using the Netjack transport makes more of difference in this case than it would on a more powerful computer. I haven't yet tested it on a more powerful computer. While my tests are subjective listening tests, I hear more extended frequency extremes and more depth in the soundstage using the Netjack setup.

Please note, at least in the case of feeding my USB DACs, that I hear more differences between differing computer hardware solutions on the player end compared to the various software tweaks I've played with. In other words, my Fit-PC Slim with an aftermarket power supply running normal media player software sounds better feeding my USB DACs than a standard quiet PC or AC powered laptop running all the software tweaks I can throw at it. In this regards, I think John Swenson's idea of an FPGA based player would have benefits.

So, if you are hardware-happy, and like to be experimental on the software side, I encourage you to give Netjack a try. The architecture here is that the player computer only needs to run the Netjack daemon (it can be Windows, Linux and maybe a MAC). The server also runs the Netjack daemon, and it can be Windows or Linux or maybe a MAC). On the server, you can use any media player that supports a Jack output on Linux, or any media player that supports an ASIO output on Windows. Do you like to use Foobar, Winamp, JRiver Media Center, CPlay, Music Player Daemon, XMMS, Aqualung, or Alsaplayer? They will all work. If you have a USB DAC, you can get that big noisy monster PC and monitor out of the listening room and replace it with an itsy bitsy teenie weenie PC.

Ryelands, I'd be particularly interested if you would try this on your networked CMP solution. I haven't played around much with different players on the server side, but in theory they should sound pretty much the same. Imagine, using a responsive media player on the server with a good interface and great metadata tagging capabilities like J-River, instead of that archaic, boring CPlayListEditor crud that is available for CPlay. I don't think it would be terribly difficult for you to set up, and I'd be glad to help.

Alan

Question, posted on October 29, 2009 at 00:49:00
Charles Hansen
Manufacturer

Posts: 4320
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>> My Netjack player is implemented as a Fit-PC Slim with a small SSD running a stripped down version of Linux. <<

So it sounds like you are using JRiver with the ASIO output feeding Jack on your server. How do you control JRiver from your Netjack player? Do you have a monitor and a mouse?

Thanks!

RE: Question, posted on October 29, 2009 at 05:33:38
aljordan
Audiophile

Posts: 819
Location: Southern Maine
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Hi Charles,

I did try J River on the server feeding the Netjack player and it works, but this is not normally how I run things. The person would have to control JRiver at the server, or use one of the remote control apps for J River.

My player in the listening room does not have a keyboard, mouse, nor monitor. I normally use Music Player Daemon on the server, and control it with a laptop, an iPhone, or an iTouch.

Alan

thanks for the excellent posts., posted on October 25, 2009 at 23:19:18
Old Listener
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Posts: 693
Location: SF Bay area
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Thanks to Gordon, John and Charles for excellent posts in this thread. This forum (and the PC Audio forum) has more value when you all contribute detailed explanations of PC audio technology.

John, a comment on the netjack approach. For success in the market, I think you will need a Window audio driver so that players that use the Waveout, DirectSound or WASAPI (exclusive or not) interfaces can send audio data to your device. In the Mac world, you might get by with a Rogue Amoeba Airfoil-like re-director or maybe you will need a full OSX device driver.

---
My own choice for an Ethernet based architecture for my own use would be somewhat different from the architectures Charles and John described. The dedicated playback device would receive commands over an Ethernet connection to play files and use network file I/O to retrieve the corresponding audio data. (MPD, Sonos and the Tremote functionality in J. River Media Center 14 use such an approach. None of these examples is integrated with a purely local, very low jitter clock for the DAC function and a world class DAC implementation.) I understand that if SRC was used, it would have to be in the dedicated playback device. That is not a concern for me since I am skeptical of SRC miracle claims.

---
As Charles said, async mode USB already exists in available products and it works with the player of your choice on the operating system of your choice.

I agree with Charles that there are a few closed systems using Ethernet connections but no standards are in place yet.

Bill







RE: thanks for the excellent posts., posted on October 26, 2009 at 13:56:54
John Swenson
Audiophile

Posts: 2013
Location: No. California
Joined: October 13, 2002
Hi Bill,
my problem with the way say MPD does things is that you are doing file IO at the DAC end. This is much more complex than decoding the netjack stream requiring a processor and maybe an OS. Once you do that you're back into all those questions about processor speed, memory speed etc etc.

If you are willing to give up a virtual soundcard interface and go with a dedicated player its a piece of cake to take MPD and add a netjack output. MPD would then run on the server with the disks and be remotely controlled by any of the MPD clients. Conceptually very similar to the squeezebox system, but with a much simpler streaming protocol so it can be completely decoded in a small cheap chip included directly in the DAC box.

John S.

RE: thanks for the excellent posts., posted on October 26, 2009 at 17:29:25
Old Listener
Audiophile

Posts: 693
Location: SF Bay area
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> my problem with the way say MPD does things is that you are doing file
> IO at the DAC end

Different goals, different choices. I understand what you have said and see why you made the choices you did.

I like the file I/O approach because it requires no new protocols and no new support for existing protocols by another organization. If I'm going to do s/w development, I want to control my fate.

I think you could make an analogy to the Squeezebox and Sonos product lines. I can be impressed with what Sean Adams accomplished but I think the Sonos architecture makes more business sense. The Sonos folks defined a finite development project, completed it and have been selling the result since then.

> If you are willing to give up a virtual soundcard interface and go with
> a dedicated player its a piece of cake to take MPD and add a netjack
> output. MPD would then run on the server...

I am not especially enamored with MPD but since it exists, it would be attractive for reading audio files and playing them on the playback device. I'd much rather use J. River MC as the UI component (or an equivalent that I write) and pipe the output to something like MPD on the playback device.

---
If you build a good implementation of the FPGA and get it into a well
done DAC, I'll be interested. If the Netjack folks get the PC and
Mac end finished and to a robust state, I'll be thinking of a purchase.

Right now, I am quite attracted to the async mode USB DAC approach. It exists and it makes sense to me. The idea of isolating the computer side from the DAC appeals to me more than tweaking the PC side to minimize the interaction. I just don't spend several thousand dollars very easily.

Bill

RE: Excellent post! Thank you and one quibble., posted on October 25, 2009 at 22:40:39
Charles Hansen
Manufacturer

Posts: 4320
Joined: August 1, 2001
John, you have go slow with me -- I'm just a dumb analog guy.

I'm not sure what you're proposing here. It sounds like you want to install a complex computer as a server and then put a simple computer as a receiver.

Enabling the complex music server to use any music player software will require something called JACK that will let the software talk to the Ethernet port. Then the simple player only needs a $10 FPGA with a $9000 software core to decode the Ethernet.

So far, it sounds a lot more complex than USB. With USB, the drivers in the server are in the OS already. Then the receiver only needs a $3 USB chip (and a licensing fee to Wavelength, or program it yourself).

The bit advantages of Ethernet are longer cable runs (1000' versus 15') and built-in galvanic isolation with the Ethernet transformer. We have to add about $20 worth of opto-isolators the way we do it. There are other ways that are probably cheaper, that's just the way we're used to doing it. Or you could just skip it altogether.

I dunno, I think I'd give Gordon a call before I spent $9000 on a software core. But maybe I'm still missing something.

RE: Excellent post! Thank you and one quibble., posted on October 26, 2009 at 00:33:23
John Swenson
Audiophile

Posts: 2013
Location: No. California
Joined: October 13, 2002
Here's my whole line of reasoning on this:
I don't like having a computer in or anywhere near my audio rack. They generate EMI, vibrations, noise etc. Of course you can build fanless no moving part computers and shield heck out of them, but then to get one that can store a terabyte or more of files AND have no moving parts and not emit not much of any EMI is getting tough and expensive. Throw on top some one that wants fancy UI for interacting with the player software with album art, extensive searching etc and you need an even more powerful computer if you don't want any interactions with the music at all.

Why bother with all that? Put the computer in another room, now it can be a regular off the shelf inexpensive computer. In the listening room you have quite a few choices for remotely controlling the software on the big box.

The ethernet connection lets you hook the DAC up to any computer anywhere in the house and use whatever software you want to run on that computer.

If you try and do that with USB you run into length limitations. True you can use an optical USB link, but the "inexepensive" ones don't do high speed, which precludes any 24/192 implementations. A USB link that can do a good job of high speed audio to a computer 100 feet away is NOT cheap.

The part of the system in the audio hardware (the DAC) doesn't need a computer, it fits entirely in an FPGA in the same way that the digital filter you use doesn't require a computer, its implemented with gates in the FPGA. There exits an FPGA core that implements all the necessary IP protocols, but the company charges $9000 for it. (thats a one time fee, not per device, once you pay that fee you can make as many FPGAs using that core as you want).

In order to make this work you DO have to install a program on the computer running the player software. The software works as a virtual soundcard that your chosen player software talks to. Its sort of like using asio4all, only it exists on windows, mac and linux.

The whole concept is to give the user the flexibility to build a system where the pieces are optimized for what they do, a file sever that can just be a file server, it doesn't have to be silent etc, a user interface that can be just a user interface, it doesn't have to store gazillions of bytes of data, and an audio hardware interface that can handle high data rates, allows the DAC to be in control of timing, generates the bare minimum amount of EMI, noise on groundplanes etc, AND does this in such a way that the sound quality is not effected by whats going on in the computer(s) implementing the rest of the system other than the bits being transferred by the player software.

In addition the bandwidth is high enough its possible to implement say a 6 channel DAC for those that want to do triamping for their speakers and run the crossovers in software. The horsepower for the DSP can run on an inexpensive computer in the other room. It can even run on a different computer than the file server or player computer if you want to set it up that way.

I'm not saying its the ONLY way to do some of these things, it implements a flexible system that has a lot of advantages. If you use USB you have to have a computer either in the rack or near it. My contention is that said computer will have a higher chance of affecting the sound quality of the system than the netjack chip I'm proposing.

I don't think its a good idea to implement this right now. The protocols are still being worked out. This system is part of an existing open software project primarily aimed at musicians and studios that need to move audio data around, but I think it has some very good aspects for audiophiles. And even in a full software implementation with a computer at both ends it sounds darned good.

After things settle down a bit I think this will be a very good way of building a distributed digital audio system that sounds extremely good.

John S.

John S.

All right, you've got my attention., posted on October 26, 2009 at 22:48:18
Charles Hansen
Manufacturer

Posts: 4320
Joined: August 1, 2001
Just like some folks like integrated amps and others like separates, I can see that a solution like you are proposing could make a lot of sense. We would possibly be interested in offering something like that.

Let me know if there is something we can do to help (we are *really* bad at writing PC software). Or if you figure it out, we may want to license it from you.

Like you said, it is not ready for prime time yet. But in a few months or years, that might be a great solution for a lot of people. The really tricky part is going to come when you are trying to remotely control the computer in the other room, however.

The original Squeezbox had a two-line display that was barely adequate. Now they are migrating to things like the Controller and Touch. I don't like the idea of introducing a Wi-Fi based component in my listening room -- I think it defeats the point of moving the electrically noisy computer to another room. I have an idea for a way around that. Unfortunately, the Squeezebox devices seems to be limited to 96/24 and if there is an upgrade path, I can't see it. It's a shame because it should be trivial to dump 192 down Ethernet.

You've got my e-mail. Please keep me posted.

John more thought, posted on October 26, 2009 at 05:23:09
Gordon Rankin
Manufacturer

Posts: 2247
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John,

First there are a ton of new optical USB 2.0 products. You can goto Lcom and take a look. There are also some that work off of 10BT cable and are isolated etc...

But really if you are worried about a computer and EMI and stuff then what do you think an Ethernet server is? Heck most of them probably would not even pass FCC testing A or B. Something every commercial computer would have to qualify for.

I think for Ethernet to really become an audio standard (and it can) is that it needs some easy way to be linked to any computer and or application. While JACK does that, it's not something that holds up under fire. What we need is some prototype setup that developers can have a virtual device driver interface that will allow the Ethernet dac look like a selectable audio output.

Thanks
Gordon
J. Gordon Rankin

I believe his entire premise is to get the thing out of the audio system's room, posted on October 26, 2009 at 06:35:09
keith_d
Audiophile

Posts: 931
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>But really if you are worried about a computer and EMI and stuff then what do you think an Ethernet server is? Heck most of them probably would not even pass FCC testing A or B. Something every commercial computer would have to qualify for.

Thus he has addressed and is addressing the above point.

Wow John, I'm speechless, posted on October 24, 2009 at 16:30:33
Dynaudio_Rules
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Posts: 5594
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I think I will read your post a couple of more times and let it sink in...



Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

RE: Some thoughts on async etc., posted on October 24, 2009 at 15:52:49
cfraser
Audiophile

Posts: 2061
Location: Pickering, Ontario
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Nice.

I think I understand the sort of idea you have, but real Ethernet is probably the last interface/method you'd want to use for anything where timing is critical. Of course you could have a custom/dedicated "ethernet" circuit etc., but this is kind of like custom "USB" that isn't really USB and just uses the same cables/connectors. Time to move on to connection/interface methods/connectors actually designed for the job from the ground up? For a change?

RE: Some thoughts on async etc., posted on November 3, 2009 at 08:04:34
regal
For me I would rather have a traditional USB transport like the TerralinkX with a good clock and quality external power supply than a asyn $150 hiface which is powered by the computers switching powersupply and has immature drivers. I'll wait for the next gen of asynch USB transports. I think Jitter is over obsessed about anyway, but I do applaud the work to reduce it. Maybe in a few years an affordable minute jitter computer transport will be available until then stick with reliable bit-perfect solutions.

BTW the MusicStreamer is Adaptive, read the review, posted on October 23, 2009 at 15:52:34
Gordon Rankin
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Posts: 2247
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Dyna,

Get the facts straight... the MusicStreamer is Adaptive, read the review.

It uses the PCM2706 it's a 16 bit interface.... the new pro thingy they aren't even sure what it is going to use as they say it does both... Yea I would like to know what computer that would work with.

Thanks
Gordon
J. Gordon Rankin

RE: BTW the MusicStreamer is Adaptive, read the review, posted on October 25, 2009 at 11:04:52
sed
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Posts: 14
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Their description says the Pro will have drivers. My guess is the software allows you to switch between the two modes. Gaming industry drivers usually do this to support different levels(costs) of external devices.

not possible, posted on October 26, 2009 at 05:27:00
Gordon Rankin
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SED,

First if it has drivers then it would not need to be either async or adaptive it could have it's own set of extensions and work in an async like interpretation all it's own.

BUT you cannot enumerate to both, only one.... like this:

Endpoint 0x01 - Isochronous Output
Address: 0x01 (OUT)
Attributes: 0x05 (Isochronous asynchronous data endpoint)
Max Packet Size: 588
Polling Interval: 1 ms


~~~~~~~~

Change the Attribute to Adaptive and then the computer changes the protocol to adaptive.

Thanks
Gordon
J. Gordon Rankin

Totally Wrong, posted on October 23, 2009 at 14:00:22
Gordon Rankin
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Posts: 2247
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Dyna,

First off I2S is an internal protocol. People linking up I2S externally do not have have a good system. It was never intended to do more than a few inches... why would cables make it better.

Firewire jitter is off the charts.

XLR, tell me how an XLR can be 110 ohms? SPDIF has tons of jitter as it is part of the protocol.

If you have an ASYNC system then the quality of the clock is a direct relationship to the quality of the system. Therefore you can do a ton better with a well executed Asynchronous USB system compared to any Firewire or SPDIF...

WHY??? Because your not fixing jitter. Any SPDIF, Adapative USB or Firewire system you look at is touting some jitter new fix it thing. With Asynchronous USB you don't have to fix what isn't there. This keeps the intrinsic jitter on a new level and that is why it works so well.

Thanks
Gordon
J. Gordon Rankin

Inappropriate, posted on October 25, 2009 at 11:55:00
audioengr
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Posts: 4088
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"People linking up I2S externally do not have have a good system."

Seems like a breach of rules here, or at least a breach of etiquette.

This is like saying that Async USB is a poor way to implement USB, or that using a TAS1020 as an async interface device is a poor choice. I would never think of posting such drivel, and yet you post this. Maybe you dont understand how to implement a high-performance I2S interface externally, but I've got news for you, others do.

I can point you to reviews going back ten years comparing S/PDIF to well implemented I2S interfaces. Why do you think PSAudio recently designed I2S into their system, or MSB?

"you can do a ton better with a well executed Asynchronous USB system compared to any Firewire"

Also, there is simply no difference either between the SQ of a well-implemented async Firewire interface and a well-implemented async USB interface. They can be virtually identical.

Your post is not only a criticism of other manufacturers products and methods, but blatant advertising of your own. Not unlike so many untasteful political ads.

Can we just stick to the facts here? Do I need to notify the moderators?

Steve N.

RE: Inappropriate, posted on October 28, 2009 at 08:27:26
cawson@onetel.com
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Posts: 380
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"Your post is not only a criticism of other manufacturers products and methods, but blatant advertising of your own. Not unlike so many untasteful political ads.

Can we just stick to the facts here? Do I need to notify the moderators?"

Here, here! Gordon is up to his usual tricks - the way HE does it is right - every other way is rubbish!

DAC designers have always used various methods and the introduction of newer transport / computer / DAC connections (USB, Firewire, E-SATA, I2S, etc) makes no difference to the fact that there is not (and probably never will be) a BEST way. There are DIFFERENT ways. Who cares about the technology used, it's how it sounds that matters. Well implemented transport to DAC connections dating back 15 years still stand up very well to the trendy connections of today with all the "wool-over-the-eyes" techno-babble from Gordon and his chums. He is a mine of information and no doubt a good designer - but the same can be said of many DAC designers that just get on with the job, rather than spend half their time blowing their own trumpets!

I'll bet you a pound to a penny that in 5 years time everything will have changed again and that Gordon's successors will be rubbishing his designs in the way he is currently rubbishing his predecessors' and (inexcusably) his contemporises' designs.

Peter

Can we just stick to the facts here?, posted on October 25, 2009 at 22:31:00
Charles Hansen
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>> Can we just stick to the facts here? <<

Well, you're an engineer -- your moniker says so.

It is a fact that there will always be more jitter at the receiving end of a cable than the sending end.

Some facts, posted on October 26, 2009 at 11:56:37
audioengr
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Posts: 4088
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Jitter is a function of the receiver at the end of the cable detecting the signal transition, not the cable itself.

There are at least four phenomena that can add to jitter as the receiver detects the signal transition on a cable:

1) Cable bandwidth - the transition or "edge" is slowed and then the receiver does not detect the transition as accurately in time. This is due to the internal reference noise, external power supply noise and thermal effects.

2) Skin effect - If the conductor diameters are large compared to the frequency of the energy in the signal, the currents will not be uniform in the conductor and this can cause signal distortion that is pattern-dependent. Some edges may be pushed later or earlier in time due to this distortion and its dependence on the signal pattern/spectra.

3) Dielectric absorption - this is a charging effect of the insulators in the cable. If poor dielectrics are used, the charge that is "soaked" into the dielectric will be held and not released quickly. The amount of charge that is stored and released is a function of the signal pattern. This means that the driver must sometimes drive more energy into the cable in order to make the same transition. If the cable is still storing charge from the last transition pattern, then it may require less or more energy from the driver to make the next transition as it did to make the previous transition. The driver never has 0 ohms output impedance and usually has some feedback paths internally, so this can push the edges in time.

4) Impedance mismatch. Reflections on the cable can occur due to mismatches of the terminator, the cable, the connectors and associated wiring. These reflections can bounce from the receiving end back to the source end and then back to the receiving end just as the receiver is trying to detect the edge. This can definitely push the edge in timing.

The effects of all of these phenomenon can be minimized by proper cable design and materials selection.

Steve N.

RE: Some facts, posted on October 26, 2009 at 22:56:33
Charles Hansen
Manufacturer

Posts: 4320
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>> The effects of all of these phenomenon can be minimized by proper cable design and materials selection. <<

Exactly.

The key word here is "minimized". Not "eliminated".

And don't forget about reflections due to connectors, isolation transformers, receiver circuits, characteristic trace impedance, et cetera, et cetera.

All of this can be side-stepped completely if the master audio clock is put *right next* to the DAC chip and then run the clock back upstream to the transport. Linn did this 20 years ago with the Karik/Numerik.

I saw Demian Martin at RMAF and he was talking about implementing an I2S interface with the clock running upstream (as it should for the lowest possible jitter).

Pros and Cons, posted on October 27, 2009 at 12:01:01
audioengr
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"All of this can be side-stepped completely if the master audio clock is put *right next* to the DAC chip and then run the clock back upstream to the transport. Linn did this 20 years ago with the Karik/Numerik."

Not side-stepped completely. There are still transmission-line effects on the circuit board, as well as some new factors that you have introduced, namely:

1) Power supply is usually shared between the clock circuit and the D/A chip

2) Crosstalk between the I2S signal lines - this does not happen on a good cable

3) Shared ground-plane under or near the signals

I'm not saying that the "close" I2S connection cannot be executed well because it can. There are just other pitfalls that must be addressed that usually are not.

The cabled I2S connection can also be excellent, and has its advantages.

One advantage is that it creates another separate that can be upgraded as the clock technology allows.

The clock can also be battery powered, which is an improvement that is more difficult to add to an "integrated" version.

Here is an example:

My own DAC has a built-in USB interface with direct I2S connection to the D/A chip. If I attach an external USB converter with its own power supply and drive it with 1m I2S cable to the same DAC, it sounds a bit better than the internal USB interface. If the external USB converter has battery supply, it sounds even better than that. Both the internal and external USB converters use exactly the same USB module.

Steve N.

RE: Some facts, posted on October 27, 2009 at 08:37:34
fmak
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Posts: 4099
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Guido Tent of Tent Clock fame has been producing a kit for years.

Fact is, the cable is still important.

RE: Can we just stick to the facts here?-Fact, posted on October 26, 2009 at 04:49:36
fmak
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Posts: 4099
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So what, audioengr is quite rightly pointing out Gordon's incorrect and inappropriate remarks in pushing his own (and yours) products.

RE: Can we just stick to the facts here?-Fact, posted on October 27, 2009 at 11:05:00
Eric B
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Posts: 41
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Go back and read Gordon's post. He talks about I2S implementation, Firwire jitter, XLR, SPDIF, ASYNC, and USB. IMHO Gordon's post did not contain incorrect and inappropriate remarks. He was arguing his technical philiosophy. I'm sure his products reflect his philosophy, but he was not directly pushing his own & Ayre's products. There are a lot worse examples of AA industry inmates touting their products than this.

RE: Totally Right, posted on October 23, 2009 at 15:12:57
Dynaudio_Rules
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Posts: 5594
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First off I2S is an internal protocol. People linking up I2S externally do not have have a good system. It was never intended to do more than a few inches... why would cables make it better.

Come on now, how could you say that? Its common knowledge even by YOU that IS2 is a preferred method of transfer and cables can be acquired up to 2 meters [see PS Audio] but of course you know that. IS2 has the least amount of jitter in fact some say its jitter free.

Firewire jitter is off the charts.

Could this be why so many Professional companies use Firewire? Companies and recording studios that spend millions of dollars on systems opting for Firewire products. Could it be that Weiss and others have it all wrong?

XLR, tell me how an XLR can be 110 ohms? SPDIF has tons of jitter as it is part of the protocol.

So you are saying that, for example Mogami, AudioQeust and other are doing false advertising when they claim to have a 110 ohm XLR cable?

If you have an ASYNC system then the quality of the clock is a direct relationship to the quality of the system. Therefore you can do a ton better with a well executed Asynchronous USB system compared to any Firewire or SPDIF...

Many preachers come to believe what they preach after awhile no matter how ridiculous it sounds.

WHY??? Because your not fixing jitter. Any SPDIF, Adapative USB or Firewire system you look at is touting some jitter new fix it thing. With Asynchronous USB you don't have to fix what isn't there. This keeps the intrinsic jitter on a new level and that is why it works so well.

So you are saying that the Antelope Audio Isochrone 10M "Atomic Clock" which is not asynch USB btw has jitter problems? Come on now....


Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

RE: Totally Right, no wrong, posted on October 23, 2009 at 15:32:55
Gordon Rankin
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Posts: 2247
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Dyna,

Your all wrong...

I2S was never intended for cabling. But why would you know that??? are you an engineer?

Everything has jitter and anyone who tells you otherwise is just doing the marketing game.

Impedance on a connector is very easy to calculate. That's why an RCA can never be truly 75 ohms it has to do with diameter over diameter. XLR doesn't have a chance in hell of being 110. Those same companies you mention sell 75 ohm RCA digital cables and claim as such. But everyone knows only a BNC can be truly 75 ohms.

~~~~~~

Look at all the Adaptive USB, Firewire etc... they all have jitter reduction. Why because they have tons to get rid of...

What does a clock have to do with an entire system? You could have a perfect clock and a poor system and it would still be a poor system.

Thanks
Gordon
J. Gordon Rankin

RE: Totally Right, no wrong, posted on October 23, 2009 at 15:43:18
Dynaudio_Rules
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No I am a Digital Build CAD Analyst, but I work with Engineers every single day. Have been doing so for the past 23 years. During that time, I have yet to meet an Engineer or Designer [such as myself] that is the last word on any subject, including the subject of their trade.

Neither IS2 nor USB was intended for audio cabling, but both obviously have been adapted for such use.

Either way, IMO asynch is nothing more than a gimmick based on a good idea. Asynch alone just like the perfect clock alone will not make a bad system sound good. And there are plenty systems without either that sound great.



Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

I2S is not valid for digital audio? HUH that's news., posted on October 23, 2009 at 15:56:49
Gordon Rankin
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Dyna,

Common what are you smoking...

Async is for real and that why everyone is talking about it.

Heck that is why companies who don't have are claiming they do. But know that we have given the tools to the magazines they have been telling people what is Adaptive and what isn't.

Hey look at it another way... If Adaptive is so good then why do all these companies who have SPDIF and USB in a single box and then the magazines test their product and the USB port has like 2x the jitter that the SPDIF does???

Jitter elimination is like a filter it can only remove so much crap and then the rest flies into the dac.

Thanks
Gordon
J. Gordon Rankin

RE:Input Formats, posted on October 23, 2009 at 22:08:01
fmak
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Posts: 4099
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Assync usb is just another method of dealing with jitter transmission from a PC. It is no better or worse than other jitter elimination schemes but is much cheaper from some companies such as M2Tech in Italy and Musiland from China; and they do 24/192. It is not true that all aes/spdif outputs have lots of jitter either.

See the Stereophile review of the dCS Scarletti which compares input format performance on a properly engineered dac, both sonically and measurementwise.

AES/EBU is 110 Ohm; the problem is degradation of the wasveforms due to reflections. This can be easilly seen on a good scope with proper terminations.

That is why proper relocking of PC digitals outputs are mandatory.

Fred come on, posted on October 24, 2009 at 11:49:02
Gordon Rankin
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Fred,

First there is no jitter in the data transmission from the computer to the dac in any interface other than SPDIF.

Ok so we have that established... Adaptive be it the Firewire or the USB method adds jitter because it has to change the Master Clock to assure the buffer to the Codec port (I2S, LJ, RJ, DSP whatever) has enough data to make sure it does not under or overrun causing pops and cracks.

Async mode is basically two types;

1) Simulated DAC interface using a device driver to make either USB or Firewire look like a dac and assure the buffers are supplied so there are no over underruns.

2) System device drivers for Asynchronous USB (Firewire does have this mode but no OS supports it) were all the OS's have support for it.

Ok so with Adaptive interfaces the jitter is very high usually from 1800ps to 4300ps. Any device using this technology better have some jitter reduction. BUT!!!!! as I said before these act like filters and cannot remove everything.

Short story a company at RMAF asked me to stop by their suite because they were interested in my technology. So I stopped by with Dustin Forman who designed the ESS Sabre32 dac chip. They also use the Sabre dac and was telling Dustin that they had less than 2ps of jitter. I said well the PCM2706 typically has about 3300-3800ps of jitter in I2S output and even more using the SPDIF output. Dustin immediately said there is no way we can remove that much jitter.

Anyways... back to the subject. Ok so with any Async method you can use a really low jitter master clock and derive all the Codec interface signals (again I2S, L/RJ DSP whatever). Heck you can even blow Dyna's horn and use a freaken atomic clock for that matter. At that point the I2S is derived and the intrinsic jitter (that created by the conversion of USB parallel data to serial) can be kept to a very low minimum. Therefore there is no reason to have jitter elimination as there was such a minimal amount from the beginning.

Fred... also your an engineer... how they making the XLR 110 ohms?

Thanks
Gordon
J. Gordon Rankin

RE: Fred come on, posted on October 24, 2009 at 13:09:40
fmak
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Posts: 4099
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What is the relevance of what you posted in relation to what I said about interfaces? I don't give a sausage about adaptive and Assync usb because I am perfectly happy with high quality >96k equipment. Just read and look at the measurements and reviews say, in Stereophile and HiFi News of good gear. USB audio doesn't even rate really high up.

AES/EBU impedance? Ask your friend Jocko. He said it measured 110R but fell down on reflections (on the TDR).

Gordon, you need to digest what people say and then respond, not shoot off left right and then centre (and sometimes get it all wrong).

OK, there are more subtle ways of promoting your wares.

I have to say, posted on October 24, 2009 at 12:05:31
Dynaudio_Rules
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You make a very good argument. No doubt your case is based in fact, however good sound is not as black and white as numbers on paper.

Have a Great Weekend.



Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

RE: I have to say, posted on October 24, 2009 at 14:11:09
Charles Hansen
Manufacturer

Posts: 4320
Joined: August 1, 2001
>> No doubt your case is based in fact, however good sound is not as black and white as numbers on paper. <<

Unfortunately, too true.

If we could just measure things and get them to sound good, then everybody would be making great sounding equipment. But some gear sounds better than others, as we all know.

A good example of this from long ago was the Levinson 30/31 transport + DAC. For five years or more, all of the magazines agreed that it was the best sounding digital to be had. And they also agreed that the best sounding connection was the AES/EBU input.

Now, it's not very difficult to make a conductor that is 110 ohms, as required by the AES/EBU standard. But the specified connector, the XLR, is not and cannot ever be 110 ohms. So theoretically a good BNC-terminated cable used on the S/PDIF connection should have sounded better. Presumably there was some other factor involved that allowed for higher performance on the AES/EBU input. Or perhaps all of the reviewers were smoking the same drugs.

RE: I have to say, posted on October 24, 2009 at 15:13:00
Dynaudio_Rules
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Posts: 5594
Location: Georgia
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>>Or perhaps all of the reviewers were smoking the same drugs. <<

Well you know....birds of a feather...

Honestly, I take the ramblings of reviewers with a grain of salt. They give me a good general idea, but its hit-and-miss as far as me liking the same equipment they give good reviews too.

When I was in Michigan I was fortunate enough to live near an audio shop that had some really nice gear...much of the same things reviewed for mega-buck systems. And the owner was super friendly, never pushy trying to sell something. You could actually go there with music in hand and listen to your hearts content while having a nice conversation with him and other customers. Hours later or perhaps days or weeks people would end up buying with confidence, happy with their purchase. Anyway I digress...

Oh how I miss living near a good audio shop...

Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

RE: I have to say, posted on October 24, 2009 at 19:29:58
Charles Hansen
Manufacturer

Posts: 4320
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Yes, it's always best to listen for yourself, preferably in your own system. A good dealer is indispensable. I don't know what part of Georgia you're in, but there are a couple of good dealers in the Atlanta area. Plus Jim Smith (author of "Get Better Sound" and previously the Avant Garde importer) is down there somewhere as well.

RE: I have to say, posted on October 24, 2009 at 12:30:11
Mercman
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Posts: 2001
Joined: October 20, 2002
I have stayed out of this discussion up to now. But I would like to respond to DR's last comment.

The reason I use Gordon's DAC is that he seems to have a great reverence for the "art" component of audio. He is a musician and knows what real music sounds like. You are right about good sound not being black and white like numbers on a paper. But he does build the most musical sounding DACs I have yet to hear. His stuff is not for everyone, nor am I saying that it is the best. His stuff is just a great deal of fun to listen to. I've had the Crimson for 3 years now. That is a record for me with digital gear.

And yes, there is room for many approaches to the design of fine audio equipment.

RE: I have to say, posted on October 24, 2009 at 13:19:53
Dynaudio_Rules
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Posts: 5594
Location: Georgia
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Fun is whats its all about....I was pulling his leg with the 'gimmick' comment. I really do believe he knows what he is talking about, and his approach is novel to say the least. I think you understand the jist of my posts.....good fidelity being somewhere in the gray area of synergy, its hard to quantify. Which is why we all spend so much time, effort and money on this crazy hobby..:-) If he can make products that can capture the emotion of good music then my hat is off.

Have a great weekend...




Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

"Heck that is why companies who don't have are claiming they do.", posted on October 23, 2009 at 19:09:05
Plinko
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Posts: 1359
Location: N East, USA
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Who are these companies?

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 24, 2009 at 12:17:46
Charles Hansen
Manufacturer

Posts: 4320
Joined: August 1, 2001
It's bad form (and probably against the Forum rules) for Gordon to name names.

But I know of one company making a DAC with an adaptive USB input. Their sales manager was telling all their dealers that it was asynchronous, but stopped after a few months. Below Mercman has a post about a manufacturer that is claiming to have both adaptive and asynchronous in the same device.

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 25, 2009 at 11:11:11
Plinko
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Posts: 1359
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No offense intended but I guess that means that it is in good form for manufacturers talk about other manufacturers without naming names?

The comment was offered...I'd simply like to know who these manufacturers are so that I can avoid them.

Btw, the Wavelength and Ayre DACs look like great products! Been looking at DACs for a while now and I'm not convinced that asynch USB is the ultimate panacea that it's marketing makes it out to be. Of course competitors offer their own panaceas. All part of the game, I suppose.

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 25, 2009 at 22:27:08
Charles Hansen
Manufacturer

Posts: 4320
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The one company whose sales manager was making inaccurate claims was only doing so to his dealers and has since stopped. So there's no point in naming names there. In the other case, the manufacturer has already been identified in this thread so there is no reason for me or Gordon or anyone else to get involved.

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 23, 2009 at 19:16:51
Dynaudio_Rules
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Posts: 5594
Location: Georgia
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Like companies selling 110ohm cables that are really 75ohm cables.



Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 27, 2009 at 05:30:16
aljordan
Audiophile

Posts: 819
Location: Southern Maine
Joined: November 4, 2003
Hi,

Last year I had a discussion with a radio engineer regarding the proper impedance of various connectors. He mentioned that the D-Sub connector that is used to connect the digital IO on my Lynx card (and many other sound cards) cannot possibly maintain the correct 110 ohm connector impedance. He also mentioned that only the BNC connector will have the proper 75 ohm impedance for spdif. I don't know what the end effect of mismatched impedance is, but from the RF perspective, he thought it was important enough.

Alan

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 27, 2009 at 08:42:33
fmak
Audiophile

Posts: 4099
Joined: June 1, 2002
The HD26 sub connector on the Lynx not only does not have the correct termination impedance, but there is substantial cross talk as well. The nest of tracks they have ahead of the connector pins to enable shorting plugs to be used adds to this mess.

This is why my aes 16 does not sound at all transparent compared to my RME9632. It doesn't sound bad after I have made my own cables and added capacitance to the 12V supply, but is is just not up there. I plan a major mod when I have the energy.

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 27, 2009 at 12:04:28
audioengr
Manufacturer

Posts: 4088
Location: Oregon
Joined: April 12, 2001
Why dont you just synchronously reclock the AES16? It has Word-Clock input.

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 27, 2009 at 12:26:51
fmak
Audiophile

Posts: 4099
Joined: June 1, 2002
It still falls short. I believe that there are 3 issues.

The Power Supply around the card which has poor 78M regulators even for the clock.

The poor terminations as discussed

and

hf ringing on the output aes stream.

The card also sounds different between using 2 wire and single wire connection which may be due to the output terminations.

RE: "Heck that is why companies who don't have are claiming they do.", posted on October 27, 2009 at 13:40:13
audioengr
Manufacturer

Posts: 4088
Location: Oregon
Joined: April 12, 2001
None of these things matter if you establish a new master clock and reclock the stream. Cable from the PCI card, PCI power, jitter are all dont-cares.

Steve N.

RE: Everything matters; that is why PC audio is difficult, posted on October 27, 2009 at 22:47:00
fmak
Audiophile

Posts: 4099
Joined: June 1, 2002
Everything matters. You think that master clock cables and terminations don't matter? Huh!

For more than 2 years you were plugging Foobar 0.8.3 for 'best' playback when other players were much better.

RE: Everything matters; that is why PC audio is difficult, posted on October 28, 2009 at 11:22:46
audioengr
Manufacturer

Posts: 4088
Location: Oregon
Joined: April 12, 2001
As far as jitter goes, the master clock cables from the reclocker to the device dont matter. The clock signals need to have good signal integrity, so terminations and impedance matching matters, but the jitter does not matter.

If it does matter, then there is something wrong in the design.

Steve N.

Mainstream audio companies?, posted on October 23, 2009 at 12:25:45
Kal Rubinson
Reviewer

Posts: 7158
Joined: June 5, 2002
The MusicStreamers are pretty nice. Review in current Stereophile.

BTW, the people behind it are affiliated with Muse Electronics and Classic Records.

Kal

RE: Mainstream audio companies?, posted on October 23, 2009 at 16:01:58
Jim Austin
Reviewer

Posts: 491
Location: Northern New England
Joined: November 8, 2007
Kal, the Website seems to imply that only the pro model is async. I haven't received my 'phile yet; can you clarify?

Thanks,
Jim
http://www.jazz-etc.com

RE: Mainstream audio companies?, posted on October 23, 2009 at 16:13:47
Mercman
Audiophile

Posts: 2001
Joined: October 20, 2002
The literature on the web site says the Pro is Adaptive and Asynchronous. I can't wait to get one. It can do everything for everybody!

Thanks Kal.

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