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What is the hierarchy of components/connections/software having an impact on sound quality-
the music file type?
the connection from the server/hard drive to the DAC? the DAC itself?
Is it possible to get equal/better than CD sound from Itunes?
Is it possible to get equal/better than CD soound with a wireless connection between a streamer (Apple TV) and the hard drive?
Just getting into it, and seeing it is the way to go, but I need a map...
Follow Ups:
I don't think anyone has addressed your question around the music file type. You want to make sure your music is in a lossless format. Lossless compression means the file size is smaller but none of the information has been lost. Lossy compression, like MP3 or AAC, means some information was thrown out to get a smaller file size.
I usually recommend using an optical connection from a computer, all thing being equal, because it avoids any potential for non-signal noise to go between the computer and the DAC. Many DACs, mine included, have a design that tries to minimize any noise that might travel over an electrical connection (e.g. coaxial or USB) but there's always a chance of really high noise. We measured some particularly high amplitude high frequency noise from a computer sound card during our initial R&D.
The downside of an optical connection is they usually have higher jitter than a very well designed coaxial output circuit. However, there are also a million ways to design a coaxial S/PDIF circuit so it's also easier to end up with variation in sound between transports that way.
I personally use iTunes on my Macs. I know some people prefer other playback software like Amarra or Pure Music. Be aware that if you choose to use one of those, there is a way to enable filters which will alter the sound on purpose. Most recently I am using Dirac Live Room Correction Suite software in my reference setup. This is to address issues with the gear and room though, not to compensate for something lacking with the original music data itself.
Streaming via Apple TV (I do this also) limits yourself to 16-bit/44.1kHz which is CD-quality but in itself should not result in worse sound. The Apple TV would use an optical connection which is subject to the issues I mentioned above.
You may be interested in Rogue Amoeba's Airfoil software if you are planning to use streaming and Apple devices.
nt
I did address format here:
http://www.audioasylum.com/forums/pcaudio/messages/10/106596.html
I recommend to stay away from lossless and lossy formats. They can a do compromise the SQ in real-time situations. Has nothing to do with static data compares.
Use .wav for PC and AIFF for Mac.
Steve N.
One is that the best treatment for jitter is no treatment and, more generally, that the best overall treatment is to not upsample or put digital or analog filters in the way of any part of the CD's signal before or after converting it to analog.
One doesn't have to agree with this approach, but it's worth listening to equipment that uses it and judging for yourself.
"You don't need to be a Weatherman to know which way the wind blows"
I dont agree on the jitter, but I do agree that Digital Filters do more damage than good. The less DF, the better it is. Better to do filtering with minimal-pole filters in the analog domain, and not to much of that either. Most systems seem to filter-out the ultrasonics nicely without any help from the DAC. It doesnt take much filtering to ruin it IME.
Steve N.
"One is that the best treatment for jitter is no treatment and, more generally, that the best overall treatment is to not upsample or put digital or analog filters in the way of any part of the CD's signal before or after converting it to analog."
If there is no filtering (digital or analog) in the DAC on is just sending high frequency c*** into one's amplifiers, speakers and possibly one's ears. What comes out will in no way be an accurate reflection of what was recorded. The presence or absence of filtering has nothing to do with jitter, it will still sound like c*** on a system with perfect clocks.
These solutions are common when technicians who have been promoted to "engineer" attempt to design systems without benefit of a proper engineering education, e.g. who lack the necessary background in electrical engineering and mathematics. There is little point in arguing with these people, IME, because they usually have a chip on their shoulder. These people would be better advised to recognize what they don't know and make the effort (it will be a large one for most people) to learn what they missed by an inadequate education. (In the large company where I used to work we had a program for technicians wishing to become engineers. We made sure that these people had learned the material they would have gotten had they been fortunate to have gone to engineering school.)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
electrical engineering graduates used to be some of the worst in denial of 2nd order effects in electronics.
"electrical engineering graduates used to be some of the worst in denial of 2nd order effects in electronics."
True, but it isn't unique to that group and experience eventually informs them. Of course sometimes it informs them that they would be happier in marketing...
Rick
Actually, in the design of 'intelligent' products in which software and hardware play equally important parts, it has been said in UK that mechanical engineers are more ready to learn electronics than electronics enineers to learn mechanics.
The breadth of an engineering discipline may shape adaptability and ability to assimulate.
Some General Engineering degree courses in the UK are held in high regard.
Which is best?
It really does appear that those educated via Audio forums, at least think, they are the better engineers.
No doubt there is some good information on the forums, esp with regards to tweaks. However something can be said about a good foundation in 'Classical Education', its what you do with it afterwards [thinking outside the box etc] that counts.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Try an engineering career followed by 10 years of modding other companies audio products. This gives you the best of all experience IMO. You also get to investigate all kinds of high-end audio components, including caps, transformers and wire.
The forums give you the opportunity to learn new things from other engineers and even non-engineers. No Engineer is an expert at every aspect of the design. Information is everything and noone can have their finger on the pulse of everything that is happening is a fast-changing technical world like this.
IMO, medical doctors need forums like this, so they can be immediately imformed of new breakthroughs.
Would that be an Engineering 'career' or an Engineering "Degree" or both?
I have an Engineering 'career' of 20+ years in Product Engineering, but a Degree in CAD Design...and another in Business. Working in Product Engineering gave me a unique insight of every developemnt phase, from concept to final production and everything between. The Business Degree and spending 7 or so years in Advanced Product design gave me experience in the cost/investment side. I take non-professionals opinions with a grain of salt...but at the same time I know I 'surely' don't know everything.
As for the Dr. forums to inform people...it will never happen.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Both. I have EE degree. I actually started in product engineering in 1976. Changed to design in 1977. Designed digital and managed digital design groups at various computer companies until about 2000, when I quit and did Empirical Audio full-time. I did a wide range of things, including design, product engineering, design management, lab management and program management.Now I do all of the product design, manual CAD layout, BOM generation, ECO control and prototype assembly, including fine-pitch SMD. I personally system-test every unit sold.
Its like doing your Taxes yourself. If you do your own taxes, then you know what to do to take advantage of the taxlaw. If someone else does your taxes, you dont have a clue.
Steve N.
Edits: 04/07/12
What does this mean? Non-engineers also use such descriptions.
"electrical engineering graduates used to be some of the worst in denial of 2nd order effects in electronics."
One of the purposes of a job interview is to weed out candidates who don't know what they don't know and aren't willing to admit it. Because of the breadth of material covered in the EE curricula, the underlying physics and mathematics are usually watered down compared to what would be taught in the physics or math departments, but it's still better than what one would get in a vocational school. Anyone who wants to learn this stuff can do so for free today, all the material is available on the web.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Try Khan academy. Its amazing what he teaches there. You could learn practically everything you need to pass the PE tests just by doing his classes.
Regardless of what you do upstream of the actual D-A chip in a DAC, it will be minimized/have less of an effect unless you have a DAC with a high quality analog output stage (discrete Class A, preferrably with its own power supply). But even that will make little difference if what's downstream of the DAC -- volume control, amplification, speakers -- isn't up to snuff.
I tend to agree but add capacitorless implementation as well.
Active preamps. They tend to homogenize and ruin the sound of most good audio systems.
Agree. Some of the Stereophile Class A preamps are really Class C by way of sonics.
The key issue and absolute number one priority is to find a DAC which is immune against or better rejects transport related distortions.It shouldn't matter what's feeding and how you're feeding your signal.
The key to success is getting the DAC input stage or a great reclocker in between your system and DAC or Full Digital Amp.
There shouldn't be any need to buy expensive playback software, invest hundreds of $ and hours (each of us) in never ending PC/Transport HW tuning just to cover the flaws of connected DAC input stages.
Even professional people with more than 13 years of modification experience seem to set their priorities wrong.
Unfortunately these people (and many more) still sell those flawed input stages and communicate their "own" priorities to guide you around those flaws instead of getting them properly addressed and resolved.All those issues are not new btw. We're talking about those issues since 5-6 years at least.
However. The good news. I see light at the end of the tunnel. ;)
Enjoy.
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::: Squeezebox Touch Toolbox 3.0 and more ::: by soundcheck
Edits: 04/05/12 04/05/12 04/05/12
What DAC have you found that both sounds amazing and rejects all incoming jitter?
Does it really sound the same with all sources and digital cables?
Those speakers that you own look really, really, cool though.
I would love the opportunity to hear them.
"In this land right now, some are insane and they're in charge. To hell with poverty, we'll get drunk on cheap wine."
nt
Cut-Throat
One more:Did anybody ever read about a DAC review, where the
immunity respectively rejection on incoming distortions has been evaluated.E.g. Does an audiophile player application such as Cplay makes a difference YES/NO. How much of a difference does it make?
I don't think so. I've never seen such as thing.
Is it done done on purpose??
Fact is, the manufacturers of High-End gear recommend to improve your source to make their products look good or look/sound better.
It's usually much easier and cheaper for them to recommend something like that instead of improving their electronics.The reviewers and later customers usually follow that advise, without questioning it.
######################################################################The key problem by tweaking the transport side is:
Not any different source/transport setup will produce the same sound quality on the same DAC.
And don't believe that you'll ever be finished with an awful and expensive transport tweaking!!! It'll go on forever.
######################################################################
The funny thing is that lower level products gain much more by using a
well done Transport (HW and SW), then high-end & high-priced products.It could easily be that a 250$ DAC product + well done transport or reclocker sounds better than a 1k+ DAC hooked up to a basic transport and would still be much cheaper than the 1k+ device.
Those high-end & high-price manufacturers recommend to use best transport setups to avoid such a disaster.
They need to keep at least a little distance to those low cost solutions.
The difference between low and high quality gear would usually be much more evident if both sides were fed by low quality transports.But obviously the reality is a different one.
So. Better watch out, when reading the next glorifying review of a new DAC device. What they refer to is typcally a combination of DAC & Transport. You never really know how well the DAC itself performs.To me the key criteria for choosing my next device is how immune it is on incoming distortions and how well it handles/refreshes that information.
And Beware: Some manufacturers even claim immunity. From my SBT project feedback I learned that even those devices that are supposedly immune respond to changes on the transport side.
My goal is to end up hooking up any device ( PC/SB Touch/iPad/iPhone/Android device/ you name it) to my DAC/Full Digital Amp ending up with the same sound quality. (Assumption: bit transparent sample transfers).Enjoy.
Cheers
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::: Squeezebox Touch Toolbox 3.0 and more ::: by soundcheck
Edits: 04/05/12 04/05/12 04/05/12
If you were an engineer with experience designing DACs, you would know that its not that simple. If you put layers of PLLs in the DAC, then you have the jitter of those PLLs. If you put a resampler in there, then you have those audio artifacts.
The best solution IMO is to leave the D/A alone and feed it the lowest possible jitter source. That source may be externally generated or internally from an async USB interface or network interface, both using fixed frequency clocks, not PLLs.
What you evidently dont get is that it's the fixed frequency clock with low jitter that is the key to the best possible digital audio result. Its not about some DAC rejection circuitry.
Steve N.
nt
Anyone in the Anglo Saxon world can called himself/herslf an engineer, though not a professional (PE) or Chartered engineer.
In many parts of the world, it is illegal to falsely call oneself an engineer by profession. Penalty can be a fine or imprisonment.
What measurements are you looking for? comparing to what?
As in Stereophile measurements on your products. Jitter was no better than other products.
The RMS jitter in the review may have been about the same as JA's reference, but then how did the SQ of it beat the competition?There is more to this than just an RMS measurement. The spectra is more important. This about the fallacy of these measurements: if there are spikes in the spectra, these can make a P-P measurement look bad, but a RMS measurement look good. On the other hand if there are a lot of shorter spikes, the RMS measurement can look bad and the P-P measurement look good. This proves that you need the spectral plot to know what is really happening.
And BTW, if you look at the latest recommended components, JA's original reference is rated "A" and mine is rated "A+". Guess which one he uses.
Edits: 04/07/12
"you need the spectral plot to know what is really happening."
Steve, how does the jitter spectrum correlate with perception? What's important? Are there especially bad frequency ranges to have jitter in or is it some other aspect like the correlation. Maybe discrete bad, spread out good or the other way around. I don't have any way to measure it but knowing more about it might be helpful.
Thanks, Rick
Correlation of jitter spectra to sound quality is a complex subject at best. The one thing that I have noticed is if the skirts adjacent to the fundamental clock frequency are minimal and the average jitter level is low within a few kHz of the fundamental, this seems to sound better.
Just what I was looking for...
Rick
are excellent in taking bits out of a review in promotion of what you make. According to JA, it sounds no better than others.
Then why is it the ONLY USB converter to get A+ rating?BTW, JA's unit now has all of the performance upgrades, so he will be doing a follow-up, including new measurements I hope. However it's still an Off-Ramp 4, not the latest OR5.
Edits: 04/08/12
"If you were an engineer with experience designing DACs, you would know that its not that simple"
I am an engineer. And I can tell you, I extremely well understand your challenges. I got myself pretty deep into interface design.
You IMO ask a lot of money for your products compared to the competition.
I'd therefore expect a sophisticated solution.
If you still integrate prebuild stuff like hiface ( wavelength, centrance stuff wouldn't be any different in this regard) or whatever interface
you'll have difficulties to get around the limitations, which come with those designs. The same is valid for SPDIF interface chips.
Just building a nice power supply around it and doing some extra filtering might not be sufficient.
"The best solution IMO is to leave the D/A alone and feed it the lowest possible jitter source..."
Your simplistic logic makes me scream.
First of all you need to put a very thick and high wall between you and the source feeding you.
Some weeks ago we had pretty much the same discussion. And it was you mentioning something like "source related distortions are all about common mode noise. And you got it finally under control now"
???? What about it ?????
"What you evidently don't get is that it's the fixed frequency clock with low jitter..."
Believe me I got a deeper technical understanding about all this than you'd believe.
Meanwhile there are quite some designs with fixed clocks out there.
Even those need a well done source isolation.
Asynchronous USB transfer modes obviously improved over simplistic PCM2707 isynchronous implementations. However. There is still plenty of impact of source related distortions left. I've been running an isynchronous device with a reclocker behind it recently.
I can tell you, you wouldn't hear a difference to pure asynchronous ( even bulk transfer ) designs.
There are always more then one way to achieve the same thing.
And some ways are dead-ends. That you'll need to figure a out.
Otherwise you'll get stuck.
BTW: If running asynchronous transfers it doesn't mean that you get 100%
rid of source related digital jitter. But that you know.
Asynchronous USB is a nice marketing term.
In another post in this thread you talk about Sabre DAC chip coming with a great jitter rejection. I'd say you're reading too many marketing papers.
My (DIY) experience with Sabre DACs is that these are extremely sensitive on incoming distortions. There are products out there using multiple stages of jitter/noise rejection to make that Sabre Chip sound good.
And I'm sure. If you get your interface done. Intrinsic jitter effects, output stage effects, clock effects inside your DAC will play just a minor role in comparison on the overall result.
So. As I said.
From my point of view the absolute number one priority for all DAC manufacturers must be the goal to get the interfaces immune on source
related distortions. It shouldn't matter which source I use.
I find it funny to see that you as an engineer asks the masses ( simple consumers) to fix your more than challenging issues ( it's not only you - your competition gives similar advise ) by tweaking their sources.
I can tell you. That's pretty annoying for many people.
Exactly this is one reason why computer based audio feels like gambling for best sound by many people. And that includes me - as an engineer .
Cheers
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::: Squeezebox Touch Toolbox 3.0 and more ::: by soundcheck
It is encouraging to hear from an engineer with your point of view, as I would love to have a reference-level DAC that is immune to changes in the transport, cables, source-level jitter, etc. Do you have a prototype ready or close to being ready?
Cheers,
Bill
"If you still integrate prebuild stuff like hiface ( wavelength, centrance stuff wouldn't be any different in this regard) or whatever interface
you'll have difficulties to get around the limitations, which come with those designs. The same is valid for SPDIF interface chips."
I only used the M2Tech OEM interface board for a short time, to get to market quickly. It fed my Pace-Car reclocker, so the jitter from the module was not important. I probably sold only 3-4 of these. Now I have second generation of my own async module designs.
"Some weeks ago we had pretty much the same discussion. And it was you mentioning something like "source related distortions are all about common mode noise. And you got it finally under control now?"
This is only using USB interface, and yes I have a solution that I am productizing.
"If running asynchronous transfers it doesn't mean that you get 100%
rid of source related digital jitter."
I dont believe that. My experience is that it does eliminate any source jitter. There are other effects, such as RFI and CM noise that can still get through however. When one reclocks multiple times and uses a fixed clock on a separate power supply, there is no other conclusion. There is no other explanation.
"Asynchronous USB transfer modes obviously improved over simplistic PCM2707 isynchronous implementations. However. There is still plenty of impact of source related distortions left. I've been running an isynchronous device with a reclocker behind it recently.
I can tell you, you wouldn't hear a difference to pure asynchronous ( even bulk transfer ) designs."
Yes I would, because I have. My older Off-Ramp 3 generation used Adaptive mode and I could feed it into my Pace-Car reclocker. Async USB interface is still a LOT better IME. However, If I went back and redesigned the Pace-car, which I dont sell anymore, and put the latest in regulator and clock technology in it, then I probably would not hear any difference in the two. The price difference is huge however, probably 2X.
"I find it funny to see that you as an engineer asks the masses ( simple consumers) to fix your more than challenging issues ( it's not only you - your competition gives similar advise ) by tweaking their sources."
This is like saying the vendor for Autocad is responsible for bugs in the Windows OS. I dont think so. I rely on lots of computer solutions to provide source playback streams for my products. Some are better than others, just like Mac OS can be more stable than Windows XP for Autocad. Pure Music is more stable than Amarra, but does not sound as good IMO. What I can do is provide the best guidance for my customers to avoid problems and achieve the best SQ. My customers rarely complain of these computer difficulties, because they generally dont have them.
"You IMO ask a lot of money for your products compared to the competition.
I'd therefore expect a sophisticated solution."
I'll email a link to you with internal photos so you can decide. We feel we are unique in this business. We are not anything like the competition.
Steve N.
After performing my experiments with the Synergistic Research Tranquility Base to reduce "noise" from my computer, I understand what you are talking about in real world applications.
The difference in sound quality with the Base is significant.
Not being an engineer, I'll just have to wait patiently for you fellows to solve these issues.
"What they refer to is typcally a combination of DAC & Transport. You never really know how well the DAC itself performs."
Glossing over reality are you?
Your scope is too puny, what about their sensitivity to digital and analog cables and their power supply rejection and do they have longitudinal current isolation and their EMC issues in general, both emissions and susceptibility. Oh, and of course shock and Vib.
Sorting out poorly specified DAC's by listening to them sandwiched between likewise iffy sources, amplification, power sources and listeners is not all that predictive of how well it will work in my system.
But I agree with your point that the source-DAC interface is especially important in this case but it's tough to trot out an ordered list of issues since home audio systems have essentially useless specifications and controls making each unique. The situation is asinine from an engineering standpoint but seems to work at every level so I guess it's a success. True it has bred a lot of superstition at the user level but that's probably a good thing: It adds interest and zing to what would be a ploddingly predictable appliance otherwise.
Regards, Rick
Reviewers could improve the situation with DACs if they used the worst possible transports in their reviews, rather than the best. If this became widespread, DAC manufacturers would be forced to build input and clock circuits that work well. One way of creating a "bad" transport would be to introduce a buggered up "reclocker" between a good transport and the DAC under test. This special reclocker would introduce controlled amounts of different types of jitter.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
IME, there are no technologies that deliver "input and clock circuits that work well", at least nothing that can compete with a good fixed oscillator driving an async USB or network interface. Sabre is probably one of the best, and it still benefits from a low-jitter source.
Even if the DAC did perfect rejection, you would be tied to the jitter of the internal clock in that DAC. What if they picked a poor one? Nothing you can do, short of modding it.
I would rather see lots of DACs that can benefit over time from newer clocking technologies as they evolve, whether by upgrades or external converters. Lots of folks with older and even recent DACs that are getting more mileage due to newer technologies. If you like the sound of your DAC, isnt it more compelling to make it sound even better rather than going through the pain of selecting a new one?
Steve N.
There are two separate issues. Getting rid of the dependence on the transport is the first part. Getting the DAC to sound good is the second part. Of course, that requires a low jitter clock, etc..
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
''Getting rid of the dependence on the transport is the first part.''
Don't think that anyone has done this.
"'Getting rid of the dependence on the transport is the first part.'
Don't think that anyone has done this."
That's why computer audio is such a mess. People are trying to fix a problem, but they are focusing on the most complex part of the system, probably because it suits their particular skills or budget. They are doing the equivalent of searching for car keys under the street light, rather than where they were dropped.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
I got beat up when I made a comment about how bad computers are as front-ends.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
That's because all of your "solutions" are computers. Unless you are drawing a distinction between the dedicated server products and a PC/Mac.
In the Apple world, it is becoming clearer to me that the Laptops sound better than the Mini, iMac, or MacPro since they have less electrical noise.
Given the potential treatment of a Mac Mini or Laptop with a Tanquility Basik or Base, excellent results can be achieved.
Also, software improvements are coming in the near future for Pure Music / Audirvana Plus that get around the loss of Integer playback in OSX Lion.
> > > > That's because all of your "solutions" are computers.Oh yeah, a SB-Touch has a "Linux" OS. Haha
But, even if I use a Windows machine I can still say computers suck. Just because I use something with an OS does not mean I think its ideal.
Fact IS computers are horrible A/C polluters and are not ideal for Digital front ends. Putting crystals on top or footers, bases or using software that sweetens the music still does not change that fact.
Laptop or Desktop, does not matter...neither have "audio-grade" power supplies.
Honestly I am simply shocked that you use a computer as a front-end with a system like yours. It is obviously the weak link. But, your choice of Dac means you 'must' use a computer...
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Edits: 04/06/12
"Fact IS computers are horrible A/C polluters"
How right you are. I run my MacBook Pro off the battery.
''How right you are. I run my MacBook Pro off the battery. ''
MacBook converts the battery voltage using switching elements to get the various rails needed for its operation.
Yes I know that Uncle Fred. But I'm not polluting the AC directly.
Honestly, haven't you ever considered using something like a Linn or the Weiss front-end instead of a Mac?
In a system like your having a "Non-audio" component as the core is a travesty, battery or not expensive base or not.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
I have a beautiful Basis turntable :)
Seriously, as good as the Mac/PM sounds and as good as cMP/cPlay and Linux/MPD, SBT, etc. they are all 'similar' as far as resolution and information retrival goes....ie none are truely "head and shoulders" better than the other. They all sound slightly different or slightly better in different systems but none are on the level of something like a Linn, Weiss or even the Consonance D-Linear7 for that matter. I had a chance to audition the Linear-7 last year and it was defineately better than the SBT and my MacMini. People even say the Bryston is way better than the MacMini...not by a little either.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
You lot are the ones who 'beat' up inmates who don't agree with you, especially the vendors.
Some of the statements about what makes MACs sound best have little basis in fact, such as notebooks have 'less' electrical interference than other models. No such generalised statments can be made other than perhaps specific attributes in specific versions of hardware.
Most PC audio inmates go the opposite way to you guys, who prefer fanned systems, large RAID hdds with blowers in them, and inferior built-in switching power supplies, run with wonder software that somehow ameliorate hardware deficiencies. Some are even happy with Max Mini optical outputs and promote them over electrical usb/spdif transfer.
"You lot are the ones who 'beat' up inmates who don't agree with you, especially the vendors."
Sometimes I think I'm reading a foreign language when I try to decipher your posts.
"Some of the statements about what makes MACs sound best have little basis in fact, such as notebooks have 'less' electrical interference than other models. No such generalised statments can be made other than perhaps specific attributes in specific versions of hardware."
My comments are based on what I've listened to in my sytem. Here is something interesting from a review of the Meitner MA-1 DAC that Chris Connaker wrote:
"Early on in the review period, before I inquired about the intricate technical details of the MA-1, I used my Mac Pro workstation for playback through its USB interface. Immediately I notice something wrong with the sound. Every track, well recorded or not, sounded dull and the higher frequencies seemed completely cut off. The music was unappealing and could not hold my attention long enough to finish an entire track. I switched between all sample rates, playback applications, USB ports and USB cables in an unsuccessful effort to determine the cause of this subpar sound. I new what the MA-1 was capable of as I'd been listening through the Aurender S10 server via AES and S/PDIF (RCA) for weeks. I'd been thrilled with the sound up to this point. After too much dissatisfaction with the sound quality I switched to my C.A.P.S. v2.0 server with an SOtM tX-USB internal PCI to USB converter and SOtM SATA filter. The SOtM SATA filter has individual 12v, 5v, and 3,3v RF noise filters in addition to ripple noise filters. The SOtM tX-USB PCI to USB card in the C.A.P.S. v2.0 server has its own power line noise filter, individual ultra low noise regulators to power up to two attached USB devices, onboard ultra low jitter clock, onboard PCI host controller, and separate power connector. The tX-USB has an easily accessed manual switch that enables/disables sending power over the USB cable to the DAC. The MA-1 does require USB bus power for the USB input to function. As quickly as I noticed something wrong with the previous configuration I noticed how right this setup sounded with incredible details and no digital edge. Running the Meitner Audio MA-1 via USB from the C.A.P.S. v2.0 server was every bit as good as the Aurender S10 via AES if not slightly better in the bass regions. Attack and transients were simply stunning using the Meitner recommend ASIO driver and J River Media Center. Comparing this async USB setup to the Aurender's S/PDIF (RCA) output was no contest as the C.A.P.S. v2.0 server surpassed it in sound quality. Lacking a BNC output may be an Achilles heel for the Aurender S10 if an electrical S/PDIF connection is required. Switching to the C.A.P.S. v2.0 server provided a solution, but I was not entirely sure I new the cause of the problem. I had a hunch it was due to lack of galvanic isolation on the USB input. A lack of such isolation would provide the USB connected computer a direct electrical connection to the DAC's sensitive internal components. I didn't truly know if Meitner had isolated the USB input as I hadn't asked about all the technical details at this point. I followed up with the Meitner Audio team. I was told the MA-1 USB input is not isolated and this was very likely the cause of the sound quality issue I heard when using my Mac Pro workstation with its noisy power supply, spinning drives, video card, and generally noisy internal environment. The Meitner team is very learned in computer technology. We discussed the Mac Pro and how much better many of the newer computers may be when paired with the MA-1. This is because many companies are using laptop type motherboards and power supplies whether the computer is a laptop or desktop. in fact the C.A.P.S. v2.0 server is much closer to a laptop than desktop when considering the internal components. My subsequent results when using a MacBook Pro laptop fit snugly with this explanation. Using a MacBook Pro with Amarra 2.3 and iTunes the sound quality was pretty close to the C.A.P.S. v2.0 server and Aurender S10."
"The music was unappealing and could not hold my attention long enough to finish an entire track."
This strikes me as nonsense. Either it's just typical BS audio writing where the most minute differences are exaggerated, the author has a severe case of audioneurvosa, the Meitner DAC was malfunctioning, or some other part of the system was especially quirky. I rule out the possibility that the design of the Meitner DAC could be so terrible as to cause such gross deformity in the sound, because of the good reputation of the name.
This type of writing is one of the major problems with high end audio. It drives away many serious and first-rate engineers who might otherwise be inclined to apply their talents toward producing better sound. It's not just that people write this BS, it's that audiophiles ingest it and propagate it.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
''My comments are based on what I've listened to in my sytem.''which doesn't make what you say 'facts'. I am happy to accept that you heard what you did, but this does not make the attempts to rationalise what you hear as facts.
Computer Audiophile? Light reading and sometimes amusing.
Edits: 04/06/12
Over there you either agree with Chris or else.
Same for Jim at Jriver and Lavry at his site.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
''Over there you either agree with Chris or else.
Same for Jim at Jriver and Lavry at his site.''
= amusement
I'm not the computer Guru here. All of my comments are based on observations. Are you doing anything different?
fine.
Generlisations you make, NO.
I think you have gone off the deep end Fred. I thought this was an audiophile forum, but now I see it is a scientific forum moderated by Fred.OK, I won't make anymore generalisations, but I will contine to generaliZe.
Edits: 04/06/12
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
The kind of music were expensive sound systems does not matter.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
I assume that posts made here are rational, not speculative based on preconceptons.
It would appear that your comments are speculative since you have never used a Mac. Since these computers are so popular, it's not hard to find friends to bring them over for testing. Comment on what you have experience with. If my comments don't fit YOUR preconceptions, do some testing and prove me wrong.
Some of us knows what makes sense and what seems like nonsense.
''Getting rid of the dependence on the transport is the first part.''
> > > Don't think that anyone has done this.
~~~~~~~~~~~~~~~~~~~~~
How can someone do this??? Not depend on a Transport.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
That's what I'm saying.
It's all about setting the right priorities.
If you're done with the first part, the 2nd part will turn out to be much easier because much less complex.
Cheers
--------------------------------------------------------------------------------------------
::: Squeezebox Touch Toolbox 3.0 and more ::: by soundcheck
> > > This special reclocker would introduce controlled amounts of different types of jitter.
So I guess that Dac that sounds the best under such circumstances would be the best....at least in the Jitter rejection department.
But, I think the JISCO thingy and Antelope Audio both Attenuate jitter in a controlled way as to make the downstream Dac easily filter it out. [I think] So maybe 'controlled' amounts of jitter would actually help some Dacs.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
I'm not sure I follow your logic that added jitter could make things sound better. In theory, there is a cancelling jitter stream that exactly counterbalances the jitter. But I should think it would be easier to balance a sharpened pencil on its point throughout a mild earthquake, a task that is also theoretically possible. :-)
I have no problem packaging a good reclocker and a sensitive DAC together and calling the combination a "good" DAC. If you don't like to see all the extra wires and boxes you can add a power strip and put all the stuff inside a cardboard box, which you can label "good DAC" with a magic marker.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
> > > Re: added jitter could make things sound better.
Attenuating jitter - JISCO deivce
In a digital transmission we will always have jitter because we never have infinite precision. The goal is to attenuate the jitter to a value, that is so small, that it wont bother us.
Antelope Audio's Approach:
and how adding strategic jitter does, similar to dither, break up modulation patterns to increase the linearity of D/A conversion chips.
http://www.youtube.com/watch?v=-65gN44G9hU
http://www.youtube.com/watch?v=qyb9rZgPnGo&feature=related
It seems that the JISCO alters the jitter while the Antelope adds jitter.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
The jitter Igor is discussing applies to analog circuits in a delta-sigma modulator that's part of an analog to digital converter.
In a digital to analog converter the conversion from more bits to fewer bits at a master clock rate is done by a delta-sigma modulator implemented in digital logic. The conversion could be done off-line, e.g. by software going file to file as with Korg Audiogate or Weiss Saracon. The time critical use of a clock appears further downstream in the DAC at the very point the digital signals get turned into analog. At this point there is no opportunity for jitter to interact with the operation of the digital modulator. (There may be need for random noise added to the modulator to improve it's performance, if so it can be done by using a pseudo-random number generator.)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Okay, gotcha...
My Dac uses a DSP module which seems to do the job nicely. Although I still try to feed it a low-jitter diet.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
"What is the hierarchy of components/connections/software having an impact on sound quality?
the music file type?
the connection from the server/hard drive to the DAC? the DAC itself?"
Here is the order of importance based on 13 years of modding and DAC design:
1) without a doubt, #1 is digital source jitter
2) following jitter, it's more jitter, from the digital cable to the DAC - better use a good one and 1.5m whether its USB, Firewire or S/PDIF coax.
3) the playback software - itunes and WMP are junk. Try Amarra or Pure Music on Mac or Jplay mini on PC.
4) the playback format - Use non-compressed formats only, .wav or AIFF. Rip with dbpoweramp on PC or XLD on Mac.
5) The volume control technology - most DACs use resistive attenuators, volume chips or gain control - not good. Never use 100% digital volume control, ala iTunes.
6) the DAC analog stages - most use op-amps and have mediocre power systems - not good
Notice how far down the list the DAC itself is. The digital jitter can be affected by the DAC too, particularly if it is an upsampling DAC or if it is a USB or Firewire DAC, then it is a BIG part. If neither, then it is less important. Just get a good one.
"Is it possible to get equal/better than CD sound from Itunes?"
Yes, but probably not as good as a $15K CD player. Not without Amarra or PM. Then the answer is definitely it will beat the CDP. Also depends on the Jitter above. It's #1 remember.
"Is it possible to get equal/better than CD soound with a wireless connection between a streamer (Apple TV) and the hard drive?"
Absolutely, but the high quality devices necessary to deliver this are not available yet IME. Again, its beating the $15K CDP that is interesting. If you are only interested in beating a $300 Costco DVD player, then the answer is yes. Squeezebox Touch will do this.
Steve N.
From your post, it wasn't clear what volume control technology would be better than "not good" according to your reckoning.
The SABRE chip performs digital volume control at 32 bit precision and the chip itself has about 130 dB S/N ratio. So it would seem that one could do a fair amount of digital volume control before it would be "not good", even when playing 24 bit material and especially when playing 16 bit material.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
"The SABRE chip performs digital volume control at 32 bit precision and the chip itself has about 130 dB S/N ratio."
I doubt the output stage of any Sabre DAC would come near a SN of 130dB.
Let alone the rest of the system as a whole.
"I doubt the output stage of any Sabre DAC would come near a SN of 130dB.
Let alone the rest of the system as a whole."
The link shows claimed S/N numbers for several models. The actual S/N ratio depends on the model number (e.g. the more expensive chips are selected for lower noise) as well as the configuration (more switches are used in parallel when the chip is run in mono than in stereo or multi-channel).
There is no theoretical reason why these chips couldn't meet their claimed specifications. However, the numbers are likely to be as good as any available test instruments, making it difficult to argue with the advertised results. ESS has a white paper that describes how the chip works.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Tony, you appear to have misunderstood me.
Most power supplies don't achieve 100dB SN. It is unlikely in the real world that the Sabre can achieve anywhere near its theoretical SN ratio.
As I recall the test setup for the test circuit for the SABRE DAC calls for an active power supply filter right at the DAC chip to reduce noise. There was a test board and measurements. I don't think the quoted numbers were total BS. Whether an actual product does as well is problematic.
In any event, my point was that there were other limiting items that lowered resolution more than the DAC chip, since what was being discussed was digital volume controls. The most likely source of noise in a 24 bit digital recording is going to be the microphone preamp or even the random microphone noise caused by air molecules striking the microphone's diaphragm, at least for classical recordings where the microphones are some distance from the instruments. Most microphones have self-noise of 10-20 dB or so and SPLs might peak at 110 dB, giving a maximum S/N ratio at the microphone of 90-100 dB. When played back at similar volumes (very loud) this noise will be below ambient room noise.
Perhaps someone who has the Mytek DSD192 DAC can compare the results of using this products digital volume control vs. its analog volume control.
I was going to order one of these, but was put off by the big queue. This DAC uses the "cheap" 128 dB version of this chip set.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
No worries Tony, I see that you understand SN ratio in a real world situation.
S/N specs tell you very little. Why? Because of the measurement conditions. Steady-state sine-waves. There is nothing in this to tease-out the dynamic performance of the system. The closest thing we have to that is an impulse response, which is lame at best.
One that is definitely "good" is a transformer-based linestage like the Music First. There are others, but nothing that is analog active, resistor passive, gain control or volume-control integrated analog chips IME.
#5 Steve: whats the type of vol control used in the Sabre DAC (when ruuning eg from a 95 Oppo directly to a power amp)? TIA
The Sabre volume control is done in the DSP section of the chip, so its digital.
So helpful; thank you!
How would you correct this??
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
The Wolfson WM8804 and WM8805 S/PDIF transceivers buffer and reclock the incoming S/PDIF stream so that the outgoing signal jitter is entirely dependent on the chip's driving clock. Their approach and measurements are published in an AES paper A high performance S/PDIF receiver [PDF]. I happen to use this chip for this reason.Custom-designed solutions may also buffer and reclock so the outgoing signal does not get unduly influenced by the incoming signal. I know there are DACs out there that take this route.
Edits: 04/07/12 04/07/12
====
buffer and reclock the incoming S/PDIF stream so that the outgoing signal jitter is entirely dependent on the chip's driving clock.
====
I do not think that is what the paper describes. It shows an improved version of PLL with less jitter than older "conventional" methods. But the output clock is still a (less) jittery PLLed clock to keep the elastic buffer as they call their reference buffer all the time partially filled. And that is always the case if two independent clock domains are to be aligned (input SPDIF, precise clock for the receiver). IMO the only way to go is to use some form of feedback, somehow controlling the pace of the incoming data. Be it PCI via IRQs, or USB/firewire async via feedback control messages.
From my understanding of the paper, the elastic buffer is used to buffer the data and also help to determine when a bit of data should be "released" from the buffer. And if I'm reading correctly, the eye diagram in Figure 4 illustrates the the reconstructed signal transitions at regular intervals, independent of the amount of jitter in the S/PDIF signal, for the range of jitter variation used in the experiment.
I guess should have been more careful in my words. My statement was an oversimplification as the resulting outgoing signal appears to have a clock independent of the incoming signal jitter, for the range of jitter in for which it is spec'ed.
Conceptually, this is the same to me as controlling the pace of the incoming data, because doing that would mean the transport side needs a similar buffer to hold data until a signal triggers its release, and if the buffer got close to underflowing then it would have to speed up the reads off its media while if the buffer got close to overflowing it would have to slow down the reads. This is how we do buffering and streaming of A/V data to ensure that the video and audio comes out at the right time, independent (again, seemingly) of your Internet connection.
======
Conceptually, this is the same to me as controlling the pace of the incoming data,
=======
I am afraid it is not the same. There are two buffer scenarios:
Controlled input, fixed clock output:
-------------------------------------
* CD player - the amount of buffer fill controls the CD spinning speed
* PCI card - the amount of how much of the DMA region/buffer is read by the card controls via IRQs the re-filling of the DMA region/buffer
* USB asynchronous - the fill of short buffer in the USB device determines the feedback messages sent to the player - send more data/send less data
Uncontrollable (independent) input, controlled output to fit the input pace
-----------------------------------------------------------------------
In the buffer usage of the SPDIF receiver there is uncontrolled jittery input, with output clock being continuously adjusted to keep the buffer fill optimal via advanced PLL. No fixed clock at the output, unlike in the previous scenarious. And that is a major conceptual difference.
Very nice summary!
And how well we implement the "controlled output to fit the input pace" tells the story...
There are a slough of approaches to clock recovery but there will ALWAYS be some residual jitter (at the D/A) so the ultimate focus is on reducing it's amplitude and forcing it out of band (for the listener) while maintaining adequate lock range and speed to track the W/C input. In my view the topology and implementation are equally important.
Rick
Uncontrollable (independent) input, controlled output to fit the input pace
-----------------------------------------------------------------------
In the buffer usage of the SPDIF receiver there is uncontrolled jittery input, with output clock being continuously adjusted to keep the buffer fill optimal via advanced PLL. No fixed clock at the output, unlike in the previous scenarious. And that is a major conceptual difference.
Again, I think we're in agreement but you are using what I consider to be the edge cases to argue a point. It's a valid point; I'm afraid I'm not the best at articulating what I'm thinking.
The difference between adjusting input or adjusting output only kicks in if input jitter is too high, forcing the WM8804 output clock to move past its "intrinsic" jitter specification.
However in both cases, attempting to control the output rate or the input rate, if there is data in the buffer then your output will be reliable and identical. Assuming that the small amount of jitter at output is negligible, then it doesn't matter if you are adjusting the input or output rates. (In the case of a uncontrolled input, you want the buffer to stay in the middle, so you avoid underflow or overflow.) Of course, one can reasonably argue that it is a lot easier to keep data buffered when using a computer and requesting data, since a computer can have much larger buffers.
My comparison to streaming A/V could be described as follows, and had more to do with A/V bitrate than frames per second.
If you don't have a super fast Internet connection, then even if you are controlling when you ask for data, if you don't get the data in time, you are in trouble. So you adjust the bitrate instead. The corresponding way to achieve the same result when you are adjusting the output clock is to have a super big buffer (so you don't have to adjust the output clock).
A computer and audio DAC falls into the category of a super fast connection. Wolfson states that their buffer is big enough to handle incoming jitter of 0-0.5UI @ 1kHz with 50ps of output jitter which they categorize as negligible.
I hope that gets across what I was thinking a little clearer. I completely agree with what you said; I was just thinking something different. And maybe still not explaining it very well.
AKM receiver chips reclock too. They are still subject to the incoming jitter. There are no DACs that are immune to incoming jitter IME.
Do you have proof that your Wolfson chip makes it immune? Any change in sound noted with cable changes or source changes?
Every product has its own intrinsic jitter measurement so even if it could filter out incoming jitter there will be the issue of self jitter.
Many different approaches but in the end you just have to settle for at least intrinsic jitter and design the product from that point on...after all, the output stage makes a huge difference.
Seperate power supplies for digital and analog then going further to have seperate power supplies for right/left analog each heavily regulated.
Fully descrete/fully balanced with no opamps, quality parts throughout.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
It is the buffering + reclocking that allows the Wofson chips to handle much higher jitter without affecting the output signal jitter. Reclocking without buffering first would not provide the same level of benefit.
It is not 100% immune to jitter, as there is a limit of course. This limit is included in the measurements in their paper. I have not done extensive testing, and none where I have captured measurements. I do think it is reasonable to accept the measurements included in the paper as evidence in support of their design; those measurements were made with controlled S/PDIF signals containing varying amounts of jitter rather than a random sampling of devices.
Using a poor transport with the WM8804 can still sound bad, but the one case where I noticed a difference large enough that it was definitely not just in my head was with a very cheap DVD player over coaxial. I didn't notice it with optical, so I attributed it to a poor coaxial output circuit. I regularly use what are believed to be high jitter optical sources like Macs. With other (higher quality, more expensive) transports connected via coaxial I have not heard a difference between the optical or coaxial connections on my DAC.
The measurement dont mean squat IMO. It's the SQ that you hear. If you are hearing differences, then case closed.
OK, I was just reacting to the statement that the output jitter is entirely dependent on the chip clock, as would theoretically be the case with async communications. IMO a short buffer is a key part of any SPDIF receiver. E.g. many SPDIF receivers can be put into slave mode, accepting external clocks as master. Without a short buffer for reclocking (SPDIF clock in, master external clock out) it would not work.
I would not read too much into this "short buffer". Most receivers have this. Its always a PLL with internally generated self-timed clock. They can reduce jitter a little, but certainly not below audibility. Nothing like a free-running clock.
Yeah, I think we're totally on the same page. :)
If you have a jitter problem due to the interface on your DAC, then one way is to get a USB converter with lower jitter and drive the DAC S/PDIF or I2S if it has this.Another way is to have a reputable modder install a good power supply and low-jitter clock in the DAC.
A third way is to put an in-line jitter reduction reclocker in the S/PDIF cable to the DAC.
A fourth way is to replace the entire DAC with one with a lower jitter interface.
Steve N.
Edits: 04/04/12
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Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
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