|
Audio Asylum Thread Printer Get a view of an entire thread on one page |
For Sale Ads |
76.25.70.1
All the major PC motherboard makers have announced motherboards with built-in Thunderbolt support. More Thunderbolt devices on the way, too.
Follow Ups:
Love the guys who have really good gear but dump it in a heartbeat for something not nearly as good, as long as it has something his doesn't.
Which guys are you talking about ?
The guys who will need to have Thunderbolt.
I think everyone will dump their computer systems and purchase a T'Bolt MB.
Not!!
It means about as much as USB3, faster transfers but no audible improvements.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
it usually takeas a year or two before the MB/interface makers sort out the issues - such as USB3 which required several updates and is still not as consistent in transfer behaviour as spec'd. Then they may move onto another interface before fully sorting it.
It's called an ugrade.
.
Why is Thunderbolt not as good?
"Love the guys who have really good gear but dump it in a heartbeat for something not nearly as good"
Doesn't matter if it's better or not. It's "faster, newer, better". Here's the same DAC you have, but it has Thunderbolt. You want this one!
system that they deserve.
Yes, - sometimes, - people jump on the "latest" fad. But, - after garnering more experience, - most folks dial in their "right" combination of things that work well, and work well for them.
"Asylums with doors open wide,
Where people had paid to see inside,
For entertainment they watch his body twist
Behind his eyes he says, 'I still exist.'"
at 10gbps, a full backup of a 1tb external hd would be almost, ummmmm, a joy!!!
Makes no difference to me so long as the backup completes while I am still ZZZ...
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
But I'll still argue that with computers, faster is better.
What are the real advantages of TB?
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
The real advantage of Thunderbolt is speed. This is useful in some cases with audio, e.g. when copying an album or backing up a library, but for real time playback the other interconnects are more than adequately fast.
"Does Thunderbolt affect sound?" This is an ill formed question that can be asked by someone who doesn't understand the technology, isn't thinking logically, is being careless, or is peddling something. The correct question is, "How does the combination of Thunderbolt and these other components sound, compared to an alternate configuration?"
It may be that Thunderbolt plus an external drive is better than some alternative, e.g. an internal SATA drive, an external ESATA drive, an external USB drive, etc. But if that is so, it is because that alternative is affecting the sound quality. All computer audio systems use a significant amount of RAM memory for playback. This means that at least several seconds worth of audio can be played back without the use of any mass storage device as a base line to see whether any storage device or combination of devices "affects" sound quality. (I am assuming that devices that are powered down and not connected in any way to an audio system and which may have been removed from the listening room do not affect sound quality.)
Even when a storage device affects sound quality it can do so only indirectly (unless it is way too slow or gets a data error). The effect is through other parts of the system designed to produce sound on the basis of the bits taken off of mass storage. If these devices work well then none of the storage devices would affect sound in any way. Since they generally don't do so well there might be changes in sound quality, but these will be characteristic of an interaction between the storage device or storage cabling and the audio components.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
"Even when a storage device affects sound quality it can do so only indirectly (unless it is way too slow or gets a data error). The effect is through other parts of the system designed to produce sound on the basis of the bits taken off of mass storage. If these devices work well then none of the storage devices would affect sound in any way. Since they generally don't do so well there might be changes in sound quality, but these will be characteristic of an interaction between the storage device or storage cabling and the audio components."I think this sums things up very well. I have 2 Thunderbolt drives; a Promise Pegasus and a G-RAID Thunderbolt. Both seem to sound better than my stable of OWC drives using Firewire 800 or eSATA to my MacBook Pro 17.
But to add more confusion to the discussion, the two Thunderbolt drives sound different from each other. So I question if speed is the only determining factor here.
Add the Synergistic Active SE Thunderbolt cable, and the sound changes again. (For the better in my opinion).
Would a PC with Thunderbolt using the same drives have the same outcomes I have had with the Mac? Who knows?
So in the end, each individual will have to try these things to really know if there is a difference in his system. General comments like FOS, etc, don't really explain what is going on.
Edits: 07/09/12 07/09/12 07/09/12
Have you tried my suggestion of loading the music into RAM and then powering down and disconnecting the external storage?
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
I will this weekend.
Mind if I borrow your Synergistic Base and a few cables?
But really, everything does make a differene...and YES even computers, OS's, RAM and anything else you can think of in or having to do with the musical chain.
That being the case, I can't try everything. At some point people just have to be satisfied.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Or nothing does. Or so I have read. Since not even the people who design and build the stuff agree, who cares? If it sounds good, that's good enough. How fast does it need to be to serve music to a DAC?
""Does Thunderbolt affect sound?" This is an ill formed question that can be asked by someone who doesn't understand the technology, isn't thinking logically, is being careless, or is peddling something."
IOW, the entire audio industry is FOS.
Its all about making money.
The true value of something is always less than the market value otherwise they can't make money. With a little Capitalistic motivation and Marketing know-how they are able to drive up the perceived value of a product and increase profits.
FOS, well its all a part of the game. 24/192 sounds better than 16/44.1, this wire sounds better than that wire. Marketing magic helps to influence perception and all of a sudden WOW it does sound better....I can hear it! After that, bada-boom bada-bing, the worth of a product just increased.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Sure everything affects SQ!! - including your altitude, humidity, size/shape of your room, size of your head, length of your hair, color of your eyes, shape of your music server, the rake angle of your ears, the condition of your hearing, the health of your sinuses, and the number of gnats farting in your listening room. So why not a new I/O port on your music server, whether it is active or not. ;-)
.
"Asylums with doors open wide,
Where people had paid to see inside,
For entertainment they watch his body twist
Behind his eyes he says, 'I still exist.'"
There used to be a guy on AA who claimed that the finish of the paint on the walls affected sound reproduction in the room. Semi-gloss was brighter than satin, etc. I can't remember if he made any claims about particular colors, but wouldn't be surprised.
Audibility is another matter alltogether, of course.
I'm not familiar with how chemically different are different kinds of finishes of water-based paint. However, with polyurethane for instance, the amount of sheen is regulated by addition of aluminum oxide powder (the more matte the finish, the more of it is in the mix). There's no reason why something like this should not - in principle - affect sound-reflecting properties - for high frequencies, of course.
I'm not surprised. It's a well known fact that audiophiles perceive silver colored gear to be brighter sounding than black.
This is precisely why I use a black disk drive to offset the light colored Mac. I also have a black DAC to offset the silver integrated amp, which offsets the black speakers.
''Since they generally don't do so well there might be changes in sound quality, but these will be characteristic of an interaction between the storage device or storage cabling and the audio components.''
you do agree that computer components affect sound.
If a computers make acoustic noise and this is deemed to affect the sound adversely then it should be moved out of the listening room or replaced with silent equipment. If a computer generates SPDIF then it contains a sample clock and can affect the sound. There is little reason to configure a computer this way, however. If a computer is performing DSP (intentionally or unintentionally) then it definitely affects the sound, and presumably the user can configure this for advantageous use or else disable it.
Apart from the above considerations, the computer is not in the audio signal path and there is no reason why it should affect the sound. If it does so it is only indirectly by creating a noisy electrical environment for the DAC and amplifiers. With good equipment any differences here should be small, minute in fact compare to a slight change in recording technique or a slight adjustment of speaker position. There are people who worry about these minute changes, but IMO most of them suffer from obsessive compulsive disorder.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Your claim that the computer is not in the audio signal path is perfectly logical but that kind of logic clearly does not have any traction in the Asylum! Its almost impossible to prove a negative.
Go back to the OP and read between the lines. The fellow is coming from a Grand Tradition of over 50 years of audiophilia. One of the most spectacular bits of pseudoscience is that cables have a sweeping impact on the quality of reproduced audio. Certain effects, such as using a grossly undersized wire for speaker cable, are clearly audible in repeatable experiments. Beyond that, an entire industry has arisen and rides on our backs peddling things like $1000 AC power cords.
We want to believe a data cable has "sound" because we can easily buy a shiny new one, plug it in and say we hear improvement in the music. We choose to ignore all the hideous connectors, crappy power supply, unshielded cables, noise of every imaginable sort, etc. just a few centimeters away inside the PC case, because its hard to get in there and work on any of that.
Why do I complain? Because cable mania sends an unfortunate message to the audio industry that "We want to be ripped off!" Instead the message could be "We want better recording quality, less compression and processing, more and better downloadable music" - things that could have a huge positive impact on the listening experience.
''If a computer generates SPDIF then it contains a sample clock and can affect the sound ..........'' etc
This is the serial kind of assumption you have consistently made which led to my comments. It is not just the sample clock, but signal integrity that has to be generated and maintained that matters.
I am not going to explain any more; I am tired of doing so onto deaf ears.
Fred, in the past I have posted various links to how digital logic works and the physical principles on which it is based. You have never replied with any indication that you have read these, let alone understood them.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
In the real world, with real-world GOOD equipment, these differences are pretty far from small - and certainly much greater than those resulting from "slight adjustment of speaker position".
In your opinion, people who worry about these changes suffer from obsessive compulsive disorder. In my opinion, people who disregard them are missing out on very good chance to drastically improve performance of their systems.
Seems to be a case of 'conviction' denial, like bits are bits.
People will claim bits-r-bits but at the same time they buy expensive Dacs which is not needed if bits-r-bits, only difference would be the output stage.
PureMusic, cMP and all other High-end music software...not needed as long as its bit perfect, only difference would be ergonomics and file management.
Fact is bits are not bits and the Audio-world knows it.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
I think you should look a little more closely at the parts of a DAC. The output stage is the least of the problems, there are many other components that affect the output waveform, starting with the clock and associated clock circuitry. In addition, the power supply for all the critical circuitry affects how these critical components operate. Then there is the critical input circuitry that must extract the bits from the noisy input waveform and send clean bits on to the rest of the DAC. There is not much in a DAC that isn't critical.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
.
Your tongue in cheek dismissal ignores the fact the bits delivered are indeed the same bits. It is easy to prove that exact same data is delivered by many of these programs so something else is going on because they ARE the same bits.
So it has to be some interaction of the computer and the DAC = some difference in timing. The solution is to use the computer only for storing and arranging the data and then deliver the song file all at once to the DAC. Then let the DAC deal with it... take the computer out of the playback equation.
Now I see above a proposal to address the chip delivering spdif directly? IMHO a huge step backwards since you still have all of the timing issues. Get the computer out of the equation.
If you are concerned about these differences then you should definitely go about taking steps to eliminate them. They are the result of some combination of poor or defective equipment, cabling, or system setup.
There is not going to be a substantial improvement possible in my system. It already does more than a credible job reproducing most recorded music, limited primarily by the recordings themselves and evaluated by how far playback fails to capture the essence of live performances. There is a significant deficit in the low base below 30 Hz, which is partially an equipment limit and partially a room issue. This is obvious on recordings of organ music, but not any other recorded music that I have. It may be possible to get slightly better bass by further tweaks of the digital room correction, but the room is too small to ever achieve really low bass and too small to accommodate the necessary large speakers. You must understand that I am interested in reproducing music. I am not interested in reproducing sound for its own sake.
I listen only to recordings of acoustic music, and frankly do not believe that the qualities appropriate to other types of music are relevant to the recordings that I work with. Without a reference to some kind of constant standard, it is not possible to tell whether a difference is small or large.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
You must understand that repeating the same one more time doesn't make it more valid to anyone else. It's obviously already valid to you personally, and to those who agree with you.
The fact is, it still sounds like nonsense to me, borne out of limitations of your system's resolution, ear/brain sensitivity, and possible (to much lesser extent) the kind of material you listen to.
As I said, in fantasy - or ideal, if you prefer - world, what you said would be true. In real world, with real-world equipment like DACs that are NOT immune to all kinds of interference/noise (from computer, power line, airborne RFI etc.), it's true only if your own agenda dictates it to be.
No,They are likely the result of pretty good equipment, not working well together, or possibly mis-applied. Where with such an amazing disparity in "world view" amongst designers, - what is good for one person, is "different" for another. Hence, Krell vs Cary. Two great designs, - but very different sounding.
""There is not going to be a substantial improvement possible in my system.""
You will never know that until you test it. Experience is closer to knowledge/reality than anything else.
"Asylums with doors open wide,
Where people had paid to see inside,
For entertainment they watch his body twist
Behind his eyes he says, 'I still exist.'"
Edits: 07/10/12
"They are likely the result of pretty good equipment, not working well together, or possibly mis-applied. Where with such an amazing disparity in "world view" amongst designers, - what is good for one person, is "different" for another. Hence, Krell vs Cary. Two great designs, - but very different sounding."
I think we agree when it comes to equipment not working together properly. That's what I mean by "system setup". One can mix and match excellent components and the result may be poor, but this is the fault of the person assembling the system.
""There is not going to be a substantial improvement possible in my system.""
I measure the quality of system playback by the difference that I hear between live performance of acoustic music with recordings of similar music. Since this difference is small with my present system, there aren't going to be what I would call "large" improvements possible. Of course there can be small improvements, but at some point a music lover has to stop making small improvements to playback and get on with acquiring and listening to music.
With non-acoustic music there is no such reference and one never knows how much further "improvement" is possible. However, this is inapplicable in my case because I seldom listen to this type of music and generally do my best to avoid listening to it at all.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
again....
It's not all bad....
Some people just have different goals.... it doesn't mean that their choices are wrong, their gear is defective, or they have not bought, or set up a system to its optimum capability.
From your writing, it appears as though people are doing the wrong thing, buying the wrong gear, or are just incorrect, - rather than making different choices.
If someone chooses an amp to bring out great techno music with excellent bass slam, - it's naturally going to be a different kind of amp than one that makes a traditional classic string section sound lovely.
Manufacturers produce equipment to sound exactly like they want it to. And if that sound is different, (sometimes dramatically), to a different product/topology, - it doesn't mean that they've erred. How many people died in fires caused by "defective" high-end equipment exploding?
"Asylums with doors open wide,
Where people had paid to see inside,
For entertainment they watch his body twist
Behind his eyes he says, 'I still exist.'"
N/T
f
"Asylums with doors open wide,
Where people had paid to see inside,
For entertainment they watch his body twist
Behind his eyes he says, 'I still exist.'"
seem to want to generalise on the basis of your needs and setup. We are pointing out that this should not be the case.
As I have said many times, with the assumption that the PC or Mac is driving a good external DAC, the computer music server itself affects the sound very little in the grand scheme of things, relative to the rest of the system.
So I have to agree with Tony. An adjustment to the speakers, adjustments to room acoustics, and upgrades to other components downstream of the computer will have a much bigger impact on the sound than tweaking the computer music server to death.
I have used every OS available in every set up I can imagine and have always heard differences, some of which were huge....none were so small that a slight tilt of a speaker could resolve.This type of inquiry is known as Qualitative Research, were the researcher investigates what is felt or perceived by those being researched. In Qualitative research the researcher is not supposed to project their own notion of what 'they think' is real into the research but take information provided by the informants as real. In this case we have two groups that perceive different things.
At this point we can use Quantitative Research to hold some of the variables constant while manipulating one in order to find the true cause. We might even include the group that does not perceive significant changes in sound to increase the sample size.
In the World of Audio...everything is subject to the DBT. Funny how audio-enthusiasts let a test determine reality. In the real world science is interested in discovering reality not ruling it out, as what a DBT would do. DBT says, lets have a group of people listen while we run some tests and at the end either we hear it or not, the DBT is the law of reality. Good thing the world does not use DBT otherwise many of the fine nuances of life would not be recognized and only the large macro aspects perceived by the greatest number of people would be real.
The point is, this small discussion and debating back and forth trying to use deductive reasoning to make specific claims based off of general observations does not amount to anything. Neither side will concede to the other.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Edits: 07/10/12 07/10/12
If you take a careful note of what I have written you will see that there are two limitations to my comments about large changes in the sound. First, the same bits have to be sent to the DAC. Second the DAC has to send timing information to the computer, rather than the usual way (e.g. SPDIF) whereby the computer sends timing information to the DAC.
You don't post your system and you write about various DACs. When you hear large differences it may be that your system is running in a mode where the DAC is not in charge of timing. In this case, it would not surprise me that you hear large differences when making changes to the computer.
You haven't explained how you verified that changes to the computer still sent the same bits to the DAC. Unless one performs careful tests to verify that the same bits are being sent, then the most likely cause of a large change in sound is a change to the bits.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
First, the same bits have to be sent to the DAC.Second the DAC has to send timing information to the computer,
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
My most recent experience.Gear:
Linux OS 32bit and 64bit Lubuntu
Wadia 121 Dac
Dynaudio SpeakersBoth Linux versions running on identical SSD's, with MPD configured exactly the same, ALSA out via USB, bitperfect 'hw' used not 'plughw'.
Music was fed to MPD:
Once with music stored on the SSD
Once with music fed via USB external HD
Once with music fed via Newtwork from another machine
All music was RBCD 16/44.1 flacAll electronics plugged into a Balanced power transformer and remained plugged in for the duration. Analog being fed to Dynaudio speakers from Wadia Dac via 6ft Balanced cable.
32bit sending bit-perfect data out via USB Asynch to the Wadia 121 Dac
64bit sending bit-perfect data out via USB Asynch to the Wadia 121 DacThe sound coming from the 64bit OS was noticeably dull when compared the the exact same tracks from the 32bit OS. So much so that it was not listenable. Highs did not extend, vocals were not as clear, bass did not have the same impact.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Edits: 07/10/12
Have you verified that both O/S configurations are bit perfect? If so, how did you do this? Note that 32 and 64 bit operating systems require different instructions and this means that after compiling and loading there may be different results. It is even conceivable that there are CPU bugs affecting one mode of addressing. (Unlikely, but if so it wouldn't be the first time that Intel had a bug in CPU design.) The phrase is "trust but verify" but when it comes to computer audio my experience is that should forget about trust and go straight to verification unless one personally knows and trusts the design engineers.
Incidentally, it's not clear to me how to verify that USB output is bit perfect. One way would be to use an external USB to SPDIF adapter and record the SPDIF output by feeding it into another computer. However, even this won't be perfect, because the USB "enumeration" process sends configuration from the USB device back to the operating system and this will cause different code paths to execute in the computer, which may or may not matter. (It won't matter if the code is bug free, but then is there much bug free code?) If one gets similar sonic results when comparing the two OS through a USB to SPDIF converter to your DAC and if this converter is powered separately from the computer (e.g. has a source of power other than the USB) then this may be a useful way to see if you are getting the same bits from the two operating systems. The advantage of looking at the USB output stream is that you can easily verify that this stream is bit perfect by recording it on another computer and comparing files. If the results only appear with USB, then getting to the bottom of the problem is likely to be very difficult due to the complexity of USB. If I were designing a USB DAC I would certainly do this, but it would require specialized equipment that creates and records USB protocol. I assume this equipment exists, but (fortunately) I haven't found the need to use it. Some "USB protocol analyzers" showed up that were priced under $1000 in a Google search, but I don't know if these are any good.
I looked at the Wadia web site to see how their USB input works. I didn't get a good feeling for what I read. It looks like copy written by the marketing department. (When this happens, usually it means that the product feature is missing and the marketing department doesn't really want to ask the engineering department how the product works, lest they then be guilty of writing misleading copy. I've been on both sides of the marketing-engineering fence, so I know how this game is played.) I would not be surprised if the Wadia 121 USB implementation is not really asynchronous. Perhaps someone has looked carefully at this product and done some degree of reverse engineering and can comment more about this.
Small differences in sound will be much harder to track down as they won't show up reliably in listening tests, creating a constant source of noise in experimental results that make troubleshooting difficult. (Anyone who has diagnosed a broken electronic component will appreciate how much difficult it is to diagnose an intermittent device.) In your case, you've said the differences are large, so isolation may be (relatively) easy, particularly if you have a variety of equipment you can mix and match for listening tests.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Have you verified that both O/S configurations are bit perfect? If so, how did you do this?
Two ways.
One send a higher sample rate to the Dac and watch the light reflect sample rate.
Two:
cat /proc/asound/card0/pcm0p/sub0/hw_params
This will give you the result of sample rate and bit depth.
Also like I said....using "hw" to set up MPD is bit-perfect.
CPU bugs affecting one mode of addressing.
Kind of hard to swallow that my CPU has a BUG which contributes only to digital audio. Everything works perfectly in 32 and 64 bit.
I looked at the Wadia web site to see how their USB input works. I didn't get a good feeling for what I read. It looks like copy written by the marketing department.
The Wadia uses XMOS as the USB interface, if there is a fault or defect in the device it would fall into XMOS's lap. BTW I do have another USB-to-SPDIF converter that is Asynch. In fact I have two, the Musical Fidelity V-Link and the M2Tech HiFace 2.
In your case, you've said the differences are large, so isolation may be (relatively) easy, particularly if you have a variety of equipment you can mix and match for listening tests.
Yep, I have tons of gear, converters and Dacs with a variety of input methods. Really easy to start changing variables.
Either way, I am not an isolated case. Others hear differences as well, we can not assume that everyone who hears a difference has a Bug in their CPU or defective gear. At some point we have to accept the reality that people can perceive differences in sound from the computer.
The only way to get to the bottom of this is acknowledge all perceptions and start to isolate and manipulate variables. Still, given a large enough sample size there will be those who fall at the ends of the Bell Curve [no difference at all -and- a marked difference] and those who fall at various points within the curve at hearing a difference when certain variables are manipulated.
This kind of study may be useful to a manufacturer who wants to spend time/money on variables that are perceived by the greatest number of people as having a impact on sound.
People on one end of the spectrum will likely not notice any advantages to features developed to enhance sound. While people at the other end will likely notice differences in sound related to enhancements. Of course those who do not notice any difference will not understand or believe that anyone could legitimately discern a difference.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
What you described shows that the sample rate getting to the DAC is what you expect. It does not show that the bits are correct. Similarly, showing the setup parameters of the MPD shows how the software is "supposed" to work, but not that it actually is working this way.
My comment about CPU bugs was in the spirit of showing that trusting a complex piece of computing equipment may not be wise if it is possible to perform tests to verify correctness. Intel got to the problem with CPU bugs in the Pentium, where certain floating point calculations were incorrect. They were going to cover this up rather than spend the million(s) of dollars to scrap and retrofit, but were convinced by some experienced people that this would ruin their reputation. We didn't have this problem with the VLSI CPUs designed at Digital Equipment Corporation, but only because we had previously experienced problems with inaccurate arithmetic. At least one published scientific paper in a refereed journal had to be recalled after errors in calculation were found to have arisen due to a logic design error in one of our CPUs. As a result of this experience, an entire department of "computation quality" was established at DEC to prevent these kind of errors from happening again. Fortunately, no one was killed, as might have happened if the error had appeared when designing a bridge or airplane.
Everything does not work perfectly. If it did you wouldn't be hearing differences. Since you are hearing differences, something is not working perfectly. If you care about these differences then with sufficient determination you can get to the bottom of them.
The reason why I don't presently hear differences in my transport is that I have systematically exterminated them to the point where the ones that remain are beyond the resolution of the rest of my system and my hearing. If I had not gotten to this point I wouldn't be posting now, I would be trying to figure out what is causing the differences that I hear and what I could do about them. In the case of any new piece of equipment this process can be very time consuming, whether it is a computer, a new OS, a new player, a new DAC, new amplifiers or speakers, etc. I am talking about months of testing and tweaking.
Saying that "everything matters" in audio is a cop-out. If everything matters then nothing matters and one will be constantly spinning one's wheels. Focus is needed.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Everything does not work perfectly. If it did you wouldn't be hearing differences. Since you are hearing differences, something is not working perfectly.
this is really a tautology, and while true in some sense, just seems to beg the question.
If one can isolate the cause of a difference to a part of the system that is relatively easy to improve one can make progress. This can be done by careful experimentation involving a combination of listening tests and measurements.
It is inexpensive but not necessarily easy to achieve bit perfect audio from a computer system. This is is one of the easy steps that can make a big difference. Some posters seem to believe they have achieved this, but from reading their posts it seems likely that they have been fooled. (This is based on my personal experience of having been fooled more times that I would want to admit.)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
I can see that you are firm in your position.
Stubborn....hmmmmmm
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
You haven't shown that you are getting bit perfect audio from the two versions of Linux. I would bet that you aren't, since the history of digital audio shows that bit perfect sound is rare. Here's a little history to consider:
Going back about 10 years when pro audio software started to be used to produce recordings on computers there were many packages that had poor quality DSP and that were not bit perfect in "bypass mode". This was the rule, rather than the exception. After bad press, articles, books and tools appeared this situation started to change. Computer audio then moved into the audiophile arena. Going back about four or five years, all the Apple fanboys used to complain about how bad Windows was, how it it wasn't bit perfect, etc. Eventually people figured out how to work around these limitations. Then about two years ago the Apple fan boys started noticing that there were problems with the Apple audio software and a proliferation of audiophile products started appearing. Perhaps it's now time for Linux and we will start seeing the issues appearing and getting solved. Or perhaps there are other forums where these issues have already surfaced.
Even when software is known to be bit perfect there are still so many combinations of options, knobs, switches, etc., that it is never clear whether a particular setup is actually bit perfect. The only way to know for sure is to test it. Software is so easy to "update" and "improve" that even if things are bit perfect today, tomorrow they may not be. So it is necessary to periodically vet the software.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
You haven't shown that you are getting bit perfect audio from the two versions of Linux.
----------------------------------------------------If I send 3 different files recorded at 3 different sample rates and both the proc information from the computer and the Dac say they are sending and recieveing the sample rates correctly how is this not proof of bit-perfect?
If I used the guide lines provided by the audio software [MPD] and ALSA to configure bit perfect audio, how is this not proof of bit-perfect?
I have been using Linux for the past 6+ years and have always gotten bit-perfect. I've been into Computer Audio for nearly 10 years so the pro's surely have been using it longer than me. Bit-Perfect from Linux is no secret and it surely does not lag behind Windows and Mac.
If you don't think people can and do get bit-perfect audio from Linux then what can I say? I guess Linux will be bit-perfect when you say it is....and Linux users will know how to get bit-perfect when you show us how because we are all doing it the same way.
BTW, Pro audio was using computers a lot longer than 10 years ago. ProTools has been out for more than 20 years.
Wiki:
The first version of Pro Tools was launched in 1991, offering 4 tracks and selling for $6000USD. The core engine technology and much of the user interface was designed by and licensed from a small San Francisco company called OSC. OSC was known at the time for creating the first software-based digital multi-track recorder, called DECK, in 1990.[5] That software, manufactured by OSC, but distributed by Digidesign, formed the platform upon which Pro Tools version 1 was built. The OSC designers and engineers responsible for that technology, Josh Rosen, Mats Myrberg and John Dalton, split from Digidesign in 1993 in order to focus on releasing lower-cost ($399)[6] multi-track software that would run on computers with no additional hardware. Although the original design remained largely the same, Digidesign continued to improve Pro Tools software and hardware, adding a visual MIDI sequencer and more tracks, with the system offering 16 bit, 44.1 kHz audio recording. In 1997 Pro Tools reached 24 bit, 48 track versions. It was at this point that the migration from more conventional studio technology to the Pro Tools platform took place within the industry.[7]
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Edits: 07/10/12 07/10/12
.
I don't see how any of your tests prove anything about the data other than it is the same depth and rate.
Bit perfect to me means if the original file sample stored on the computer is 1101 1001 1111 0101 then what ends up being fed to the DAC chip is the same 1101 1001 1111 0101 and no samples are altered.
How does any of what you stated prove that is true? How did you prove that the data had not been somehow changed?
.
I've tried various Macs and a couple PCs and quite frankly, I don't hear a huge difference. A difference yes, but nothing mind blowing. On the other hand, it has been my experience that the components downstream of the computer, starting with the DAC, contribute more significantly to improvements or differences in the way the audio system sounds. I can only tell you what I hear.
Hi Abe...even though I put my post under yours it was really not directed towards you but rather a general post at all the comments.
I don't dispute what you hear or don't hear, it just shows that perceived changes in sound has a range from little to none -to- quite a bit.
This might show up to look like a bell curve if enough people were asked.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
Chalk me up as the one who does not understand the T'Bird technology. Even after your explanation, the cost to benefit ratio seems very low.
Dynobots Audio
Music is the Bridge between Heaven and Earth - 音楽は天国と地球のかけ橋
SPEED. If you were doing a lot of high resolution video editing, then you might see the benefit of Thunderbolt in terms of responsiveness.
I do a fair amount of audio editing, but there is little benefit from higher speed I/O since most of the files are small enough to be cached in the RAM memory several times over. This means that I can move the files between disk drives and even across a gigabit Ethernet without having to wait at all most times, even when making a backup. However, 100 Mbps Ethernet was noticeably slow for the audio work I was doing.
As to having active cables, this is really a connector issue more than anything else, electronics on one side of the connector vs. electronics on the other side. That's about it. If the connectors are bypassed by direct connection there is no difference.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Cost to benefit ratio? What does it cost? To the consumer, having a Thunderbolt port doesn't cost much at all. All Macs have it built-in, some PC laptops have it, and every major motherboard manufacturer has announced boards with built-in Thunderbolt ports. Yes, it costs a little more, maybe $25 - $50.
Some benefits of Thunderbolt, not necessarily for audio, include:
1) It has the bandwidth to accommodate faster external storage, at least 2x of USB 3.0 today. The theoretical peak limit for USB 3.0 is ~640MB/s but actually somewhat less due to protocol overhead. Modern SSD and NAND flash modules are already achieving nearly 600MB/s in Read/Write performance which is bumping the limits of USB 3.0. If you were to stripe two SSDs for ~double the throughput, USB 3.0 would be the choke point, Thunderbolt would handle it. 50Gb/s Thunderbolt is already in the works at Intel slated for ~2015.
2) It will make HD video capture, editing, and transfers much less time consuming.
3) Thunderbolt is essentially a connection into the PCIe infrastructure which means just about any PCI-e card can be accommodated via Thunderbolt, via a Thunderbolt enabled chassis or dock. Lets say you have an Ultrathin lightweight laptop that you carry for travel but need high performance graphics and connectivity at your desk. A Thunderbolt docking station could tie into high performance graphics cards and other devices when your laptop is docked.
Here's an interesting product that takes the Mac Mini and allows it to connect to PCI-e cards via the Thunderbolt port:
Another product that extends the Thunderbolt port for laptops:
Hi Abe,
I have read up on TB and still don't see how it will help audio. If you are doing massive backups from one hardware controlled SSD RAID0 array to another hardware controlled SSD RAID0 array that is one thing, but that is not audio.
24/192 audio is a lot smaller than USB3, SATA3 or TB. I use SATA3 SSDs that load complete tracks into memory and then play out of memory. How would TB improve playing out of memory? I am already getting parallel 500MB read/write speeds (one for OS and one for data) and would need a dedicated PCIe hardware RAID0 card to get faster speeds or buy a multi-thousand $$$ FUSION card.
I have yet to see a USB3 DAC. If much faster above 24/192 speeds was better, the DAC manufactures would already be building USB3 DACs.
Also, TB is 2X USB3 speeds, but my Z77 mobo's already have multiple USB3 and SATA3 controllers, so I already have more speed than TB that I am not using to its fullest.
Any thoughts as it applies to audio?
Thanks,
Tim
For audio, I think it is more applicable in the studio.
"Thunderbolt began at Intel Labs with a simple concept: create an incredibly fast input/output technology that just about anything can plug into. After close technical collaboration between Intel and Apple, Thunderbolt emerged from the lab to make its appearance in Mac computers."
"MacBook Air, MacBook Pro, iMac, and Mac mini now give you access to a world of high-speed peripherals and high-resolution displays with one compact port. That’s because Thunderbolt is based on two fundamental technologies: PCI Express and DisplayPort."
.
The cables use active components so they're going to be more expensive than simple wires. Unless of course we're talking about 'audiophile wires' in which case a $50 Thunderbolt cable is a relative bargain!
Interesting article linked below that attempts to describe the Thunderbolt cable and high cost.
Flight of fantasy?
The questions are:
How dirty is the signal
How much 'compensation' is done by the chip and what is the extent of the 'clean-up'
Dirty signals - prevention is better than cure.
Sources within the telecom industry told ArsTechnica that active cables are usually used at data rates in excess of 5Gbps. Chips at either end are calibrated to the attenuation and dispersion properties of the wire in order to "greatly [improve] the signal-to-noise ratio."
"Active cables are copper cables for data transmission that use a silicon chip (semiconductor) to boost the performance of the cable. Without a chip, a cable is considered a 'passive' cable. Passive cables are liable to degrade the data they carry, due to such "channel impairments" as attenuation, crosstalk and group velocity distortion. In active cables, one or several semiconductor chips are embedded in the cable to compensate for some or all of these impairments. This active boosting allows cables to be more compact, thinner, longer and transmit data faster than their passive equivalents."
Active cables offer benefits over their passive equivalents:
- An overall improvement in signal integrity
- Cables can support much longer reach
- Cables become thinner, lighter, more flexible and more compact
- Cables have a tighter bend radius
- Cables can be more easily routed
- Cable conduits are smaller, allowing for better air flow
- Cables require less signal processing by the host system, leading to savings in power consumption
Active cables have been used for years in various applications.
There is no quantifiable measure; just more IT speak
Mercman has gotten to the bottom of why these cables are active.
You can be sure that the people building the active Telecom cables cited in the linked article are quantifying their results. These cables are going into systems that will be accpetedas working or rejected as not working according to measured bit error rates. The people building these cables will have the appropriate test equipment to measure and tweak components as required. If a communications channel is pushing the state of the art when designed, then it is much harder to design and specify individual components so that "mix and match" combinations will work reliably. This also tends to over constrain tradeoffs that may arise in the future, e.g. if better transceivers become economical cost may be reduced by using them while substituting lower performance and cheaper wiring.
The Telecom industry is not to be confused with the Information Technology "industry". The Telecom industry has a tradition of careful engineering going back over 150 years backed by known laws of physics and mathematics.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
This is well known. The bottom of it seems to be that the signal is dirty and that active electronics is needed to 'compensate'. This adds complications and it is not surprising that different cables affect signal transfer which can result in sonic differences.
External power supply for each end of the cable and custom "audiophile" dongles. How is this an "improvement" if our precious audio signal has to make it's way through two additional dongles before hitting the DAC?
I never said it was an improvement for audio quality, and I couldn't speak from experience because I don't have Thunderbolt or a Thunderbolt DAC. I suppose it could be argued that with active transceivers on each end of the cable, they will be less prone to external influences and potential noise.... but hey, I'm just speculating like most everyone else here.
Heck, I don't even own a Thunderbolt disk or a computer with Thunderbolt I/O (yet).
Post a Followup:
FAQ |
Post a Message! |
Forgot Password? |
|
||||||||||||||
|
This post is made possible by the generous support of people like you and our sponsors: