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06-18-2009 Post does not mapped to Knowledge Tree
Telstar
Posts 30
Joined on 02-06-2008

Post #: 76
Post ID: 10832
Reply to: 8338
Computer transport
fiogf49gjkf0d
 scooter wrote:

* Regarding playback, I misspoke in my original post of an "Audiophile music server." Upon further reflection, this is an oxymoron, if not a moronic phrase. I think it is highly unlikely that a dedicated audiophile CD player will be beat by a home-brewed or commercially available music server:

- The microcomputer used for a music server is unable to perform specialized application roles at top-tier levels. The microcomputer's operating system and hardware are too generalist, too buggy, too noisy and have too much stuff going on in the background

- The apparently simple task of moving 1s and 0s from a hard drive to a DAC is still a challenge in 2008 (interference, jitter, data loss, etc.). Part of this is due to the limitations of a generalist computer system being utilized in a specialized application


This is not true.

Point one, "The microcomputer used for a music server is unable to perform specialized application roles at top-tier levels": what does this mean?
A today cpu is hundreds of millions times more powerful than any microcontroller used in a cdp or dac. It can perform anything that a cdp/dac can and many things more, all in realtime and without introducing significant delays. Generalist is just a false statement. There are very specific pieces software.
"too buggy": maybe

Point two, the simple task of moving 1s and 0s is still a challenge with any transport used. With a separate transport and DAC, the very same interface issues apply. the best combos have proprietary interfaces (usually based on i2s or ST) with their jitter and interference problems. A single box cdp has an easier life on this.

The issues are basically the same for the computer, as transport, namely (1)- interferences (RFI/EMI), more than within a cdp, also due to fans and other moving parts*, and (2)- jitter, which with some interfaces can be higher than from a SOTA transport, but from other sides can be lower because of no moving parts. it is surely very high using spdif, the worst digital interface ever created (TM by me). Anyway, it's always the DAC which accounts for the vast majority of the sound quality. This is why the guy above feels that his music server with Wavelenght dac sounds better. There is also the unknown rule that NOS sounds better with a computer (and nobody can explain why).

* A computer transport without fans and moving parts can be done with today technlogy, and for a very low cost. The critical point is the interface, the connection to the D/A. I do not have first hand knowledge of the top pro units such as Lavry, Pacific microsonics and Meitner, but i have listened to very good consumer setups. My favourite digital surce is the naim cd555. Yep, extremely regulated psu, in a separate box. Power matters. Everything matters.

06-18-2009 Post does not mapped to Knowledge Tree
Telstar
Posts 30
Joined on 02-06-2008

Post #: 77
Post ID: 10833
Reply to: 8390
The interface is the culprit
fiogf49gjkf0d
 manisandher wrote:

I am convinced that the soundcard/interface is the main cause of the inferior performance of HDD vs. CD.


It is.
06-18-2009 Post does not mapped to Knowledge Tree
Telstar
Posts 30
Joined on 02-06-2008

Post #: 78
Post ID: 10834
Reply to: 9201
Music servers pop like mushrooms
fiogf49gjkf0d
 alexd wrote:

As far as Music Server goes, the single board computer with linear power supply or battery supply is correct approach, I was talking about MAC or PC.


As long as you isolate the signal that goes to the dac from the noise of the computer, you do NOT need to design an expensive linear power supply for the entire computer.

This is just one of the dozens of so called "audiophile" music servers that are growing like mushrooms and that in common have only two traits: 1 bloated pricetag and 2 some modified soundcards and/or psu.
People with a bit of computer skills can build the same for a fraction of the cost, and tailored for their needs.
06-18-2009 Post does not mapped to Knowledge Tree
Romy the Cat


Boston, MA
Posts 9,545
Joined on 05-28-2004

Post #: 79
Post ID: 10835
Reply to: 10834
The main cause of the inferior performance...
fiogf49gjkf0d

 Telstar wrote:
I am convinced that the soundcard/interface is the main cause of the inferior performance of HDD vs. CD.

 I disagree. The main cause of the inferior performance of HDD vs. CD is not the soundcard/interface but the CD format itself and the fact that the CD are over-edited by deaf and semi-moronic people. If the CD be data CD and contain the raw 16/44 files than the qulety would be orders of magnetite more superior.

Also, yes, the music servers pop like mushrooms but it is meaningless. What those people put on their music servers? The copies from crappy CD? That is ridicules. The industry still holds control over music destitution and does not allow to have any row uncompressed files.  Without is anything else is garbage. If I did not have my FM source I would not even have and recognise not reasons in music servers.

The Cat


"I wish I could score everything for horns." - Richard Wagner. "Our writing equipment takes part in the forming of our thoughts." - Friedrich Nietzsche
06-19-2009 Post does not mapped to Knowledge Tree
tuga


Posts 174
Joined on 12-26-2007

Post #: 80
Post ID: 10845
Reply to: 10832
PC grounding.
fiogf49gjkf0d
 Telstar wrote:
 scooter wrote:

* Regarding playback, I misspoke in my original post of an "Audiophile music server." Upon further reflection, this is an oxymoron, if not a moronic phrase. I think it is highly unlikely that a dedicated audiophile CD player will be beat by a home-brewed or commercially available music server:

- The microcomputer used for a music server is unable to perform specialized application roles at top-tier levels. The microcomputer's operating system and hardware are too generalist, too buggy, too noisy and have too much stuff going on in the background

- The apparently simple task of moving 1s and 0s from a hard drive to a DAC is still a challenge in 2008 (interference, jitter, data loss, etc.). Part of this is due to the limitations of a generalist computer system being utilized in a specialized application


This is not true.

Point one, "The microcomputer used for a music server is unable to perform specialized application roles at top-tier levels": what does this mean?
A today cpu is hundreds of millions times more powerful than any microcontroller used in a cdp or dac. It can perform anything that a cdp/dac can and many things more, all in realtime and without introducing significant delays. Generalist is just a false statement. There are very specific pieces software.
"too buggy": maybe

Point two, the simple task of moving 1s and 0s is still a challenge with any transport used. With a separate transport and DAC, the very same interface issues apply. the best combos have proprietary interfaces (usually based on i2s or ST) with their jitter and interference problems. A single box cdp has an easier life on this.

The issues are basically the same for the computer, as transport, namely (1)- interferences (RFI/EMI), more than within a cdp, also due to fans and other moving parts*, and (2)- jitter, which with some interfaces can be higher than from a SOTA transport, but from other sides can be lower because of no moving parts. it is surely very high using spdif, the worst digital interface ever created (TM by me). Anyway, it's always the DAC which accounts for the vast majority of the sound quality. This is why the guy above feels that his music server with Wavelenght dac sounds better. There is also the unknown rule that NOS sounds better with a computer (and nobody can explain why).

* A computer transport without fans and moving parts can be done with today technlogy, and for a very low cost. The critical point is the interface, the connection to the D/A. I do not have first hand knowledge of the top pro units such as Lavry, Pacific microsonics and Meitner, but i have listened to very good consumer setups. My favourite digital surce is the naim cd555. Yep, extremely regulated psu, in a separate box. Power matters. Everything matters.


Telstar, what do you think would be the most efficient way to ground a computer? Would this affect performance as much as it does any other audio component?

Cheers,
Tuga


"Science draws the wave, poetry fills it with water" Teixeira Pascoaes
06-19-2009 Post mapped to one branch of Knowledge Tree
Telstar
Posts 30
Joined on 02-06-2008

Post #: 81
Post ID: 10848
Reply to: 10835
Interfaces, interfaces
fiogf49gjkf0d
 Romy the Cat wrote:

 Telstar wrote:
I am convinced that the soundcard/interface is the main cause of the inferior performance of HDD vs. CD.

 I disagree. The main cause of the inferior performance of HDD vs. CD is not the soundcard/interface but the CD format itself and the fact that the CD are over-edited by deaf and semi-moronic people. If the CD be data CD and contain the raw 16/44 files than the qulety would be orders of magnetite more superior.

Also, yes, the music servers pop like mushrooms but it is meaningless. What those people put on their music servers? The copies from crappy CD? That is ridicules. The industry still holds control over music destitution and does not allow to have any row uncompressed files.  Without is anything else is garbage. If I did not have my FM source I would not even have and recognise not reasons in music servers.

The Cat


I think I got your point. There are a few 16/48 files that they say are not a cd-rip. No doubt they sound better.

But i was referring to the old crappy redbook format, and a track ripped to a computer (HDD, although, ad hinted before SSD or other solid state memory sound better because there are no moving parts). That was my point, ceteris paribus, a solid state memory has the potential to sound better than any spinning transport, in reproducing the same crappy file.
Why this rarely happens? The main culprit imo is the interface. I still dont see the light there with any of the current offerings. OS and software can be tamed to the point of not being an issue. Same happens for the elimination of moving parts. But the interface? No, there's no clear answer there.

What's the point of having a music server you ask, well, lazyness i guess. Having thousands of CDs at click distance it is handy. There are slick interfaces that show all the information contained in the booklet. Redbook is garbage, sure, but if somebody has thousands of that garbage, a music server it's really helpful.

Let's talk more of such interfaces.
-Asyncronous USB as programmed by Gordon seem well done, but it's limited to 24/96 in the best case. And while he corrected the main issue, he's still left with lots of jitter and noise picked up by the usb plug. Not to forget that such computer should NOT use any other usb device or the pollution multiplicates. USb 3.0, which will be available from the beginning of next year seem to overcome some of these (i'm still skeptical about jitter, because it depends also on the usb receiver in the dac, better receivers are needed).
-Firewire (asyncronous by itself). It's probably still the best interface IMO. The jitter is not ultralow by itself, but there are good receivers and transceivers (in particular i know a very good firewire to i2s transceiver). Like anything coming from a noisy source that is not isolated by itself, it requires very good isolation. Fortunately, there are such devices to perform a complete galvanic isolation to every pin of the receiver. A mid-level (for this i compare to <5k$ cdp) firewire dac is the Weiss DAC2/Minerva.*

-In the two above the clock is in the DAC, as it should be. But there are also internal interfaces (pci/pci-express), namely souncards at pro and consumer level.
They have more issues (see below), but also the advantage of lower jitter. The issues are first and foremost the power source pollution, esp if the power is taken from the pci or pci-e slot; rfi/emi, even if we are using them only to get digital signal out of the computer (i'm not considering using the internal DAC, although there's people who even do that and that equals to very entry level CDPs), and lastly the interface. We can have all of the following, but in 99% of the cases there's just spdif:
-spdif coaxial rca: high jitter, sensitive to noise. You can hardly get worse. No wait, stereo minijack to rca is even worse.
-spdif coaxial bnc: 1/20th jitter of rca, besides, it's still spdif.
-spdif toslink: immune to noise, but several times MORE jitter than coaxial.
-AES: just a bit better than unbalanced spdif, still high jitter and not completely shielded.
-i2s (requires modding the card): lowest jitter, but very sensitive to RFI/EMI, very short cable required (which means that the rfi/emi from the computer can pass to other electronic equipment). Not common in DACs, but in some cases easy to mod, as i2s is used as internal data input for a vast number of converters.
-ADAT: potentially good, 8ch = 2ch 24/192. Unseen receivers in any commercial DAC.
-ST or double ST: probably the best interface. Low jitter, completely isolated. Unseen in any soundcard, requires modding. Very few DACs accepts it (to my memory Esoteric D70, audio sintesys dax, something from Zanden)

In all the above, the interface is penalized at the start, if it cannot be slaved to the DAC (i'm talking about clock).

[* I would like to know the impressions on the AF1 that Mani is using, I assume that improved things vs the Lynx through AES)]

06-19-2009 Post does not mapped to Knowledge Tree
Telstar
Posts 30
Joined on 02-06-2008

Post #: 82
Post ID: 10849
Reply to: 10845
Isolating a computer
fiogf49gjkf0d
 tuga wrote:
Telstar, what do you think would be the most efficient way to ground a computer? Would this affect performance as much as it does any other audio component?
Cheers, Tuga
Not grounding it? Wink
Seriously, there's people that found improvements removing the earth on the plug through which the computer was connected to the wall AC.

I would just use a good network filter between the computer and the AC, to protect the OTHER equipment from its noise (mainly due to switching psu).

This is easy, what's hard is to isolate the SIGNAL coming from the computer.
06-19-2009 Post does not mapped to Knowledge Tree
jessie.dazzle


Paris, France
Posts 456
Joined on 04-23-2006

Post #: 83
Post ID: 10852
Reply to: 10835
HDD sound vs CDP sound & NOS DACs
fiogf49gjkf0d
Manisandher wrote :
"...I am convinced that the soundcard/interface is the main cause of the inferior performance of HDD vs. CD..."

Telstar wrote :
"... It is..."

Romy wrote :
"...I disagree. The main cause of the inferior performance of HDD vs. CD is not the soundcard/interface but the CD format itself and the fact that the CD are over-edited by deaf and semi-moronic people. If the CD be data CD and contain the raw 16/44 files than the qulety would be orders of magnetite more superior..."


Yes, but it seems that's not what Manisandher is saying in the above quote.

Where Romy is comparing files from commercially marketed music CDs to raw 16/44 files, I believe Manisandher is implying the comparison between any file played back from a CD player, to that same file, once ripped and played back from a hard drive. 

Raw 16/44 files would sound orders of magnitude better than a commercially marketed music CD edited by deaf morons whether played back via a CD player or via a hard drive.

Telstar wrote :
"...it's always the DAC which accounts for the vast majority of the sound quality. This is why the guy above feels that his music server with Wavelength dac sounds better. There is also the unknown rule that NOS sounds better with a computer (and nobody can explain why)..."


Telstar, when you say NOS, I assume you are referring to what have apparently come to be known as NOS DAC modules, correct? The first time I heard about them was from the Wavelength web page, quoted below :

"...Some people call these NOS DACs or what I call zero DACs. The data input is the data output without any up/oversampling or other manipulation, which seems to make for a very analog presentation. [Wevelength markets them as their Transcendental DAC modules]. The reason I used this name [Transcendental] was because this technology is really an irrational approach that yields really good results. Basically Transcendental in math terms means irrational numbers..."

Why they are called NOS, I have no idea.

jd*


How to short-circuit evolution: Enshrine mediocrity.
06-19-2009 Post does not mapped to Knowledge Tree
ghpicard
Posts 12
Joined on 12-15-2008

Post #: 84
Post ID: 10853
Reply to: 10852
NOS -> Acronym
fiogf49gjkf0d
 jessie.dazzle wrote:

Why they are called NOS, I have no idea.

jd*


It usually stands for Non OverSampling

Gaston
06-19-2009 Post does not mapped to Knowledge Tree
jessie.dazzle


Paris, France
Posts 456
Joined on 04-23-2006

Post #: 85
Post ID: 10854
Reply to: 10853
Not vintage
fiogf49gjkf0d
Merci Gaston,
 
I was of course thinking "New Old Stock" as in vintage parts.

jd*


How to short-circuit evolution: Enshrine mediocrity.
06-19-2009 Post does not mapped to Knowledge Tree
Telstar
Posts 30
Joined on 02-06-2008

Post #: 86
Post ID: 10855
Reply to: 10852
NOS and filterless
fiogf49gjkf0d
 jessie.dazzle wrote:
 

Telstar wrote :
"...it's always the DAC which accounts for the vast majority of the sound quality. This is why the guy above feels that his music server with Wavelength dac sounds better. There is also the unknown rule that NOS sounds better with a computer (and nobody can explain why)..."


Telstar, when you say NOS, I assume you are referring to what have apparently come to be known as NOS DAC modules, correct? The first time I heard about them was from the Wavelength web page, quoted below :

"...Some people call these NOS DACs or what I call zero DACs. The data input is the data output without any up/oversampling or other manipulation, which seems to make for a very analog presentation. [Wevelength markets them as their Transcendental DAC modules]. The reason I used this name [Transcendental] was because this technology is really an irrational approach that yields really good results. Basically Transcendental in math terms means irrational numbers..."



Yep.
Gordon definition is quite good. Now I dont remember if he also avoid any analog filter after the D/A conversion. I dont like it either, it creates a pretty thick veil on the sound.
06-21-2009 Post does not mapped to Knowledge Tree
manisandher
London
Posts 156
Joined on 09-05-2008

Post #: 87
Post ID: 10868
Reply to: 10852
The PC interface
fiogf49gjkf0d
 jessie.dazzle wrote:
Manisandher wrote :
"...I am convinced that the soundcard/interface is the main cause of the inferior performance of HDD vs. CD..."

Telstar wrote :
"... It is..."

Romy wrote :
"...I disagree. The main cause of the inferior performance of HDD vs. CD is not the soundcard/interface but the CD format itself and the fact that the CD are over-edited by deaf and semi-moronic people. If the CD be data CD and contain the raw 16/44 files than the qulety would be orders of magnetite more superior..."


Yes, but it seems that's not what Manisandher is saying in the above quote.

Where Romy is comparing files from commercially marketed music CDs to raw 16/44 files, I believe Manisandher is implying the comparison between any file played back from a CD player, to that same file, once ripped and played back from a hard drive.
jd*

Yes, that's exactly what I meant in my original post. I've not really made a detailed comparison between the three firewire interfaces that I own, but have experienced enough 'strange goings on' for me to be pretty convinced that the interface is playing a major part in the overall sound.

 Telstar wrote:

I would like to know the impressions on the AF1 that Mani is using, I assume that improved things vs the Lynx through AES

I don't actually have the Lynx AES card to make the comparison (but I don't really like the idea of the D-sub to XLR cable).

Mani.

06-27-2009 Post does not mapped to Knowledge Tree
Romy the Cat


Boston, MA
Posts 9,545
Joined on 05-28-2004

Post #: 88
Post ID: 10935
Reply to: 8265
The "custom-built" by dealers audio computers.
fiogf49gjkf0d
As I learned recently that audio dealers have developed for themselves a new revenue source - they build custom music servers for their audio customers. They charge a lot of money for those machines and the people who play music on those custom-build custom music servers presume that if their music server was “build” by some kind of industry cretin then it is an assurance of some kind of “quality”. That is so fanny as all intestines of music servers are very common and the audio dealers, who build these machines and sell them 3-4 times more than they shall, invest no special knowledge and no special techniques to built custom music servers.

The Cat


"I wish I could score everything for horns." - Richard Wagner. "Our writing equipment takes part in the forming of our thoughts." - Friedrich Nietzsche
07-09-2009 Post does not mapped to Knowledge Tree
Romy the Cat


Boston, MA
Posts 9,545
Joined on 05-28-2004

Post #: 89
Post ID: 11019
Reply to: 8265
Weiss AFI1 joined the Interface Club
fiogf49gjkf0d

http://www.designwsound.com/dwsblog/?p=35

http://www.weiss.ch/afi1/afi1.htm

The Cat


"I wish I could score everything for horns." - Richard Wagner. "Our writing equipment takes part in the forming of our thoughts." - Friedrich Nietzsche
07-09-2009 Post does not mapped to Knowledge Tree
manisandher
London
Posts 156
Joined on 09-05-2008

Post #: 90
Post ID: 11020
Reply to: 11019
More about the AFI1
fiogf49gjkf0d
Romy,

I've been using the AFI1 for the last few months. I haven't done any detailed comparisons with my other two 'interfaces' (actually, they're multichannel ADC/DACs with AES ouputs), but I think the AFI1 is excellent.

The only issue that I have with it is that I can't use the PMII in Master mode at rates of 176.4 and 192KHz... the AFI1 cannot double the clock frequency from the PMII. But this will not be a problem for you, of course, as you don't use these sample rates.

I would be very interested in knowing whether the AFI1 shows any improved performance over the Lynx PCI card. The only advantage I can see really is in the dedicated power supply... possibly. But then maybe such an advantage (if indeed it exists) may be outweighed by having to use the firewire interface. But Daniel has done a good job here - he uses 'isosynchronous mode'.

Mani.
07-09-2009 Post does not mapped to Knowledge Tree
Romy the Cat


Boston, MA
Posts 9,545
Joined on 05-28-2004

Post #: 91
Post ID: 11022
Reply to: 11020
The power supply in DAWs
fiogf49gjkf0d

 manisandher wrote:
The only advantage I can see really is in the dedicated power supply... possibly.

The subject of power supply in computers that run sound is complicated subject and frankly I never heard anyone talk about it seriously or do any sensible about it. Everyone is bitching about PC power supplely but why no one make them better? Everyone is bitching about the “dirty” PC environment but no one told me why it I much PC “dirtier” then some kind of 16ch DAC running at 172kHz. I would not mention that why do we fell that Weiss, Lavry of any other “better” companies have better power supplies than anybody else. They all use simplistic CRC filtration, cheapest rectification, cheapest possible pars then can go ways with,  and the transformers that they sources from the same Chinese shops… BTW, I am not sure that he cheapest Chinese PS are the worst….

Anyhow. I use my XG Magnum PS, driver from PP2000 and I kind of like them.

http://www.xpcgear.com/psmg500.html

I did not make any sonic assessment of PS, I wish somebody do…

The Cat


"I wish I could score everything for horns." - Richard Wagner. "Our writing equipment takes part in the forming of our thoughts." - Friedrich Nietzsche
11-04-2009 Post does not mapped to Knowledge Tree
manisandher
London
Posts 156
Joined on 09-05-2008

Post #: 92
Post ID: 12143
Reply to: 11020
AFI1 now works 'properly' in dual-wire mode
fiogf49gjkf0d
 manisandher wrote:
The only issue that I have with it is that I can't use the PMII in Master mode at rates of 176.4 and 192KHz... the AFI1 cannot double the clock frequency from the PMII.


The latest firmware 'fixes' this issue. The AFI1 can now work in dual-wire mode at half or full audiorate clock frequencies.

What totally amazes me is the huge difference in sound between using the AFI1 with the PMII in MASTER vs. REF_CLK modes (at 176.4KHz and 192KHz SRs). The sound is transformed from a hard-etched one to a totally fluid one. I can only assume that this is down to the use of a poor* clock and/or power supply in the AFI1.

At these SRs, I've been using the Reference Recordings 176.4 material primarily. However, the difference is still pronounced when using 16/44.1 material and the 'Arc Predicition quad upsampling' feature in the XXHighEnd player. The latter is especially interesting with the PMII, as it has been designed to work with NOS DACs... and I believe that the PMII works in NOS mode at 4fs (I'd love to know this for sure though).

* 'Poor' in relation to the clock/PS in the PMII.

Mani.
11-04-2009 Post does not mapped to Knowledge Tree
Romy the Cat


Boston, MA
Posts 9,545
Joined on 05-28-2004

Post #: 93
Post ID: 12145
Reply to: 12143
Good for Weiss!
fiogf49gjkf0d

 manisandher wrote:
The latest firmware 'fixes' this issue. The AFI1 can now work in dual-wire mode at half or full audiorate clock frequencies.

Very good. I am impressed with Daniel’s responsiveness.  I wonder how many people would be affected and benefited with the new firmware. The 2-3 individuals around the word? Not a lot of manufactures would do it so fast and for free. Truly impressive!

 manisandher wrote:
What totally amazes me is the huge difference in sound between using the AFI1 with the PMII in MASTER vs. REF_CLK modes (at 176.4KHz and 192KHz SRs). The sound is transformed from a hard-etched one to a totally fluid one. I can only assume that this is down to the use of a poor* clock and/or power supply in the AFI1.

I do admit that in Master Mode Pacific better but I would not describe the situation when Pacific and Lynx run individual clocks as “hard-etched”. The difference is minor and I would say that I would not recognize it blindly with 100% certainty. Probably you are right and the reason is in a very poor quality of AFI1’s own clock or in bad implementation of the ways how AFI1 use own clock to slave the DAW’s resources.

 manisandher wrote:
At these SRs, I've been using the Reference Recordings 176.4 material primarily. However, the difference is still pronounced when using 16/44.1 material and the 'Arc Predicition quad upsampling' feature in the XXHighEnd player. The latter is especially interesting with the PMII, as it has been designed to work with NOS DACs... and I believe that the PMII works in NOS mode at 4fs (I'd love to know this for sure though).

I have no idea if Pacific does any oversampling. I however never do any upsampling or any rate change during recording or playback.  The worst that I had was the use of so fashionable nowadays Memory Players that can ease do rate chance.  I hated the result very much, in face I hate how all Memory Players sound.  I think the way how it shall be is a file shall be played in the rate and resolution it was recorded. Period. 

The Cat


"I wish I could score everything for horns." - Richard Wagner. "Our writing equipment takes part in the forming of our thoughts." - Friedrich Nietzsche
02-17-2010 Post does not mapped to Knowledge Tree
jessie.dazzle


Paris, France
Posts 456
Joined on 04-23-2006

Post #: 94
Post ID: 12959
Reply to: 12145
The sounds of silence; continued
fiogf49gjkf0d
Paul S wrote (originally posted in the thread discussing noise cancellation for Romy's HP servers)
http://www.goodsoundclub.com/Forums/ShowPost.aspx?postID=12957#12957

"...Jessie, are you saying that the RAID system is "quiet" with respect to regular old audible noise from the unit, itself?..."

Before adding the RAID array, the workstation was already super quiet; it uses several slow-running fans and has a case that is sound proofed.

Because of this, and because it was sitting there with 4 empty drive bays and a 1050 watt power supply (a sort of "industrial duty" workstation), I decided to install a RAID array as a place to store music files. The worstation is "seen" by the audio system (which has its own computer) as a network attached server (NAS).

The RAID array itself is extremely silent; it is built from 4 Western Digital high-efficiency "Green" 2TB drives running at 5400 RPM. The drives are model N° WD20EADS. (For anyone contemplating the same: If you want a RAID built from these drives to last, each drive will require a couple small tweaks to its firmware. WD sells drives that don't require this tweak for use in a RAID, but they cost a lot more).

"...How is it with respect to the filthy switching backwash on the AC line and EMI/RFI broadcasting?..."

The workstation and audio system are on separate circuits; as separate as possible. The power supply may or may not be of the switching variety; it is about the size of a shoe box, and weighs around 10 lbs. I've read that most PSUs used in computers are around 75% efficient (AC > DC); for this one, HP claims 94% efficiency, and that it runs cooler and hence more quietly than would be the case for a "normal" PSU.

The only part of the music server that is housed in the workstation is the library (the music files)... All processing of the data happens over at the audio system end, via a very silent Mac. The data stream goes from the PC workstation to the Mac via ethernet cable, and is then sent to a USB DAC.

There is no difference in sound quality since having moved the library to the RAID (it was previously housed in a collection of external drives, connected via FW800, daisy chained directly to the Mac).

jd*


How to short-circuit evolution: Enshrine mediocrity.
07-15-2016 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 161
Joined on 02-11-2009

Post #: 95
Post ID: 22679
Reply to: 10848
What is ST?
fiogf49gjkf0d
 Telstar wrote:

-ST or double ST: probably the best interface. Low jitter, completely isolated. Unseen in any soundcard, requires modding. Very few DACs accepts it (to my memory Esoteric D70, audio sintesys dax, something from Zanden)



What is ST?
never heard before

http://www.lavryengineering.com/wiki/index.php/Word_Clock
07-15-2016 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 161
Joined on 02-11-2009

Post #: 96
Post ID: 22680
Reply to: 22679
Lavry Digital wiki
fiogf49gjkf0d
now i am trying to setup my mac music server.
useful information :

http://www.amr-audio.co.uk/html/dp777_tech-papers_cmpDataBit4dummies.html
http://designwsound.com/dwsblog/hifi-computer-faq/
http://www.amr-audio.co.uk/html/dp777_tech-papers_OSX-Integermode.html
http://www.lavryengineering.com/wiki/index.php/Main_Page

This is Thorsten Article :  ---------------------------------------------------------------------------------------------------------------------

Computer audio has been oft-criticised for sounding mediocre. This should come as no surprise when one understands the history behind the handling of audio in the personal computer.

In the 1990's as computers became multi-media platforms, one of the primary issues was how to handle multiple audio streams (e.g. the system sound effects along with CD audio playback).

The solution was a software mixer, which was first and foremost, practical. It would convert all audio streams to the same:

  1. Sampling Rate (i.e. 44.1, 96 or 192kHz) and
  2. Data Length (i.e. 16, 20 or 24-Bit)

In Windows pre-Vista this was called "K-Mixer," for Mac "Core Audio" and "ALSA" for Linux.

alsaLinux
Sampling Rate Conversion for Convenience

If the incoming audio streams' sampling rates are different, for example: system sound effects (16-Bit@32kHz) and CD audio (16-Bit@44.1kHz), then the system software mixer would lock onto the first required sample rate and convert all subsequent rates to the same. The unavoidable result was artificial, digital manipulation.

A CPU is a generalist (it must handle email, web surfing, word processing etc…) that is optimised for multiple applications. With no onboard digital signal processing hardware, a dedicated sample rate conversion software program can load even a powerful Pentium 4 with 100% CPU usage when processing CD audio in real-time.

Therefore, in order not to burden the CPU too much, the default sample rate conversion inside the system software mixer is optimised first and foremost, for CPU efficiency; that is, a low CPU load. The conversion quality subsequently falls by the wayside.

This is one of the fundamental reasons why for lovers of music, the default audio playback on the computer system is found seriously wanting.

Data Length Conversion trying to be Clever

All modern PC's are at least 32-Bit machines. Naturally, they work best with 32-Bit data. However most if not all audio data is 16-Bit or 24-Bit and for the foreseeable future will stay that way too.

In order to convert the audio data to 32-Bit data, the operating system will simply add zeros to the end of the data to make it “32-Bit”.  16 zeros are added to 16-Bit data to make it 32-Bit. Similarly 8 zeros are added to 24-Bit data to make it “32-Bit”.

The issue arises when the audio device driver software - another piece of software to communicate with the external audio device (e.g. a DAC), does not accept 32-Bit data2 .

In an ideal world, the driver software should know that it is dealing with 16/24-Bit audio encoded in a 32-Bit data format, discard the redundant zeros and send the correct 16/24-Bit onwards to the external audio device.

In reality, nine times out of ten, this is not the case as the driver software will try to be smart and manipulate (e.g. add dithering etc.) the audio data to try to make a “better” truncation. Suchmanipulation is not transparent and most definitely not Bit-Perfect.

This is another critical reason why the default audio playback of computer systems is less than ideal.

The Solution

All of this only applies if the default audio system is used. If it is bypassed all these issues disappear.

Both the Steinberg ASIO interface (originated from professional audio use) and the WASAPI Exclusive mode (Vista and Windows 7 only) will allow the audio playback software to reliably bypass the default system software mixer and all associated “problem” areas.

J. River Media Centre

JRiverMediaCentre
Source:J. River

and

CPlay Interface

CPlayInterface
Source: http://www.cicsmemoryplayer.com

Presently, J. River Media Center and CPlay both have high levels of sonic performance when playing back lossless audio files.

On the Mac OS X, more and more auto-Sampling/Bit Rate programs are becoming available such as Audirvana which we have tested to be one of the most-sonically impressive computer audio playback software programs on the market today.sss

Audirvana in Action

Audirvana
Source: www.audirvana.com

However, setting up a PC to play back correctly (i.e. without the often sonically-degrading Sample Rate and Data Length conversion) is not trivial and most playback software still makes things very complicated for the music lover or worse still, is fundamentally incapable of such a “pure” configuration.

Despite this, with the breakneck pace of computer software development, we are optimistic that the future is one of easier configuration for dedicated audio playback applications.

Thorsten Loesch
Director – Technology ---------------------------------------------------------------------------------------------------------------------
07-15-2016 Post does not mapped to Knowledge Tree
Romy the Cat


Boston, MA
Posts 9,545
Joined on 05-28-2004

Post #: 97
Post ID: 22681
Reply to: 22680
Optimal Sample Rate.
fiogf49gjkf0d
The most interning in Larvy writing was the article about Optimal Sample Rate. Dan Lavry begin to advance this subject in his post at various audio boards from I think 2008 and then in 2012 he convert it into his “papers”  
 
http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf  
 
In my view it is very intelligent and considering from whom it is coming is the most credible position. Still, Larvy as any other industry participant I am sure protects his serf. At the time he begin to advocate his position Larvy did only 2X converters and was losing competition to 4X makers. So, he introduced a public idea of inferior sound with excessive rate. The funny part is that Lavry might be very much correct in in the subject but I do not know how to detach his position with him defending that his company never moved to 4X world with top of the line products.  
 
I am very much NOT in position to debate with Larvy. I kind of agree with his and I feel that 2X with no post-recording DSP of any kind (seldom seen) is all that we need. However, I have my own negative experience with my agreement with Lavry. A few years back I had my custom Stelovox with ½ inch master tape at 15 in/s. The quality was very nice and I decided to digitize it. I used Lavry AD and Pacific AD at 88kHz. I was not able to get the identical quality to the quality I was getting from the tape.  Then I switched the Pacific to 175 kHz and it was significantly better. I do not know why it was but it was. So, even I agree with Larvy regarding his Optimal Sample Rate position but I also feel that supporters of 4x have a merit.


"I wish I could score everything for horns." - Richard Wagner. "Our writing equipment takes part in the forming of our thoughts." - Friedrich Nietzsche
07-15-2016 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 161
Joined on 02-11-2009

Post #: 98
Post ID: 22682
Reply to: 22681
Perfect or good
fiogf49gjkf0d
it seems even Thorsten do not believe computer source could be perfect and he just guide us to reach max from computer.

" The computer is a great music server but also a source of jitter and other RF interferences that are detrimental the sound quality, even when bit-perfect reproduction is ensured."
http://www.amr-audio.co.uk/html/dp777_tech-papers_OSX-Integermode.html
---------------------------------------------------------------------------------------------------------------------------------
Beyond bit-perfect: The importance of the Player Software And MAC OS X Playback Integer Mode

Damien PLISSON, Audirvana developer

Abstract

In computer audio, the player software replaces the CD drive as the transport feeding the DAC. Ensuring bit-perfect output of the original audio signal is only a pre-requisite, while minimizing jitter and RF interferences are still strongly needed.
This paper explains the main factors impacting sound quality on the computer side, and the means that have been implemented in Audirvana player and the AMR DP-777 DAC to boost the audio experience to the next level above the normal iTunes.
These main means are bit-perfect, sample rate switching, asynchronous transfer and Integer Mode.

Introduction: bit-perfect as the only goal or the myth of the flat-square world

In the world of digital audio, the caveats of the CD player are well known, namely the read errors and the jitter induced by its mechanical transport.
It is widely thought that computer sources are immune to these issues, given that they are faithful to the original signal, that is are bit-perfect.
But unfortunately the digital world inside a computer is not a flat-square world composed of perfectly timed zeros and ones. The audio signal chain goes through different elements whose each can alter the sound quality.
In this paper we’ll look in details at these, and see what a “source direct” solution can be to minimize the adverse effects, and achieve very high sound quality, better than nearly all the CD transports.

1. Sources of non-quality

Assuming the output is bit-perfect, the computer as a source creates two main sources on non-quality:

Software-induced jitter

Digital signal is in fact an analogue waveform composed of two states separated by a voltage threshold (1 if above, 0 if under).
As presented in [MeitnerGendron91], the receiver detects the value change the moment the analogue value crosses the threshold. In addition, the shift from one state to another is not instantaneous but more slope like.

So a slight change in the reference voltage of the source will lead to a slight temporal shift in the value change detection.

voltage-jitter
Figure 1: Reference voltage induced jitter

So fluctuations in the source reference voltage create jitter, as explained in details in [HawksfordDunn96]. This is the same on the receiver side with measurement threshold fluctuations from its power supply and/or ground instability. Moreover the computer can still cause this as the grounds are linked most of the time through the same signal cables.

Computer load means rapidly changing power demands from the CPU and its peripherals, with peak demands that are directly related to the software behaviour.

Radio-Frequency & other interferences

In addition, computation, disk access, … activities mean complex current waveforms are carried on electrical lines and thus generate electromagnetic interferences. Apple computers are now made of “unibody” aluminium cases that are good protectors from inside RF interferences. But this is not sufficient as the cables connected to the computer act as antennas. And these current waveforms are also going back through the computer PSU, polluting the mains power supply.


2. The hidden audio filters of OS X

As a modern operating system OS X needs to offer shared access to the devices including the audio output to all running applications. But this is done at the expense of pure sound quality:

Audio mixer

Fortunately when only one application is playing audio, it doesn’t affect the signal and thus is at least bit-perfect in this case.

Sample rate conversion

In this shared model the device sample rate is not switched to match the original signal's, but it is this last one that is sample rate converted.
In addition a suboptimal algorithm is used to minimize the CPU load of this real-time operation.

Digital volume control

OS X offers through its mixer volume control (e.g. the one offered in iTunes). But as it operates on the digital signal, any volume value different from 100% means loss of bit-perfect and precision loss (e.g. a volume value of 25% means 2 bits precision loss).

3. The data transfer to the DAC

First way to connect to the DAC is to use the build-in TOSLINK output of the Mac. But this one should be dismissed for being too jittery for serious use.
Strong improvement comes by using “computer connection” to the DAC, being either USB or FireWire.
FireWire has long been the interface of choice for the pro-market as it is made by design to guarantee continuous streaming of AV data on large number of channels. Anyway its complexity of use (installation of driver required, hot plugging even strongly advised against by some manufacturers because of its potentially harmful issues, …) and its unclear future have made USB the widely used choice.
The first type of USB devices are called adaptive (or synchronous), meaning the DAC clock is slaved to the computer’s continuous stream of data.
More recent and advanced USB devices use asynchronous transfer mode where the DAC controls the flow of audio data, buffers it, and uses its own stable-low-jitter clock. Thus it is immune to short interruptions of USB stream (e.g. bus reset, other device burst transfer, …), and much less prone to computer jittery clock.

This combines the advantages of both worlds: ease of use of USB (no drivers), and stability of FireWire. This is a great step towards sound quality, but it is not decoupling completely the DAC from the computer, and the interferences, software-induced jitter still apply, starting by following the ground loops.

4. The player software impact

First of all the player should ensure bit-perfect reproduction of the signal by:

  • Adapting the DAC sample rate to each track native to avoid any unwanted sample rate conversion
  • Taking exclusive access ("hog mode") of the device to prevent other opened applications from interfering

Furthermore, as we have seen in section 1, the computer load (and its variations) has an impact on sound quality. Minimizing such current demands and sources of interferences is key:

  • Loading tracks before playback (“memory play”) to reduce disk access and its audible, power and RFI impacts
  • Minimizing synchronous CPU load taken for the audio data streaming operations. In addition to reduce jitter, this also helps to reduce audible RF interferences patterns, especially in low frequencies

5. Further optimization at driver level: Integer Mode

Audio playback in OSX is usually performed through a high-level framework, the Audio Units processing graph [AppleCoreAudio]. The first optimization of an audiophile player is to bypass these overhead facilities and address directly the CoreAudio lowest layer: the Hardware Abstraction Layer. (See figure 2)

player-vs-audiophile
Figure 2: Usual OS X file player vs Audiophile concept


In normal mode, all data exchanges performed across the user/kernel boundary are in PCM 32-bit float format, easing the different audio streams mixing process and associated soft clipping. [AppleHAL_1]
Note that it is anyway still bit-perfect up to 24bit definitioniv.

Integer mode

Addressing directly the HAL [AppleHAL_2] gives the possibility to bypass the two main overhead processes of the above standard mode:

Field Programmable Gate Arrays

  • Mixing buffer
  • Float to DAC native format conversion
  • gate-array
    Figure 3: Float vs Integer Mode

    In Integer Mode (see figure 3) the player software supplies a stream already formatted in the native DAC format, thus optimizing synchronous CPU load at the driver level.
    These operations performed inside the driver, in the kernel space, in real-time are on the critical path for sound quality as they are the most synchronous, happening at the very immediate moment of the data transfer to the DAC. So optimizing it is of great benefit, and this is only applicable to compatible DACs that offer this non-standard mode.

    Conclusion

    The computer is a great music server but also a source of jitter and other RF interferences that are detrimental the sound quality, even when bit-perfect reproduction is ensured.
    The player software needs to optimize and streamline the audio path to minimize these adverse effects essentially linked to the processing load synchronous to the audio streaming. Achieving “source direct” in addition to “bit-perfect” is key.

    This is what I’ve tried to get in the Audirvana player by streamlining to the maximum the real-time operations that are limited to simple data streaming in Integer Mode, while all the other processes (loading from disk, decoding, converting to DAC native format) are done

    Float mode

    offline in a preparation phase, before playback. This is called full memory play. Best results are achieved when feeding an Integer Mode, asynchronous USB DAC like the AMR DP-777 that can take advantage of all these optimization features.

    References

    [HawksfordDunn96] Bits is Bits ? in Stereophile 03/1996
    [MeitnerGendron91] Time Distortions Within Digital Audio Equipment Due to Integrated Circuit Logic Induced Modulation Products, Ed Meitner and Robert Gendron, presented at the 91st AES Convention, New York, October 1991, Preprint 3105
    [AppleCoreAudio] CoreAudio Overview: What is CoreAudio ? in Mac OS X Developer Library
    [AppleHAL_1] Audio Device Driver Programming Guide: A Walk Through the I/O Model in Mac OS X Developer Library
    [AppleHAL_2] AudioHardware.h documentation in Mac OS X Developer Library

    Replacing the HDD by a SSD removes the directly audible mechanical noise but not the other issues as it still requests important current waveforms to transit on lengthy wires. And the OS overhead is still present.
    OSX Audio low level subsystem typically requests data in 512 frames chunks, that is at a frequency of ~86Hz for a 44.1kHz sample rate.
    Note that bit-perfect playback can still happen if all effect filters (including software volume control) are deactivated. Thus stock iTunes can be bit-perfect.
    32bit float is composed of 1 sign bit, 8 exponent bits and 23 bits for the mantissa. Thus giving 24 bits of significant precision.

    -------------------------------------------------------------------------------------------------------------------------

    07-15-2016 Post does not mapped to Knowledge Tree
    Amir
    Iran Tehran
    Posts 161
    Joined on 02-11-2009

    Post #: 99
    Post ID: 22683
    Reply to: 22682
    CD ripping
    fiogf49gjkf0d
    about CD ripping :
    http://designwsound.com/dwsblog/hifi-computer-faq/cas-5-cd-ripping-for-mac-itunes/
    ------------------------------------------------------------------------------------------------------------------
    1. “Mac Quad G5” CDDA->AIFF (Just drag the file from the CD disc on desktop)
    2. “Mac Quad G5” internal DVD-Rom – iTunes 8 imports -> AIFF (1st time)
    3. “Mac Quad G5” internal DVD-Rom – iTunes 8 imports -> AIFF (2nd time)
    4. “PC QuadCore” internal Pioneer DVD – iTunes 8 imports -> AIFF (1st time)
    5. “PC QuadCore” internal Pioneer DVD – iTunes 8 imports -> AIFF (2nd time)
    6. “PC QuadCore” Plextor Premium Professional -> Extract Audio from CD -> Wav
    7. “Mac G4 Powerbook 1Ghz” External USB1.0 ASUS CD-Rom CDDA -> AIFF (drag file)
    8. “PC QuadCore” EAC -> Secure Rip -> Wav
    9. “PC QuadCore” Wavelab CD import -> Ultra Safe CD import -> Wav
    10. This file is created by our PC iTunes real time playback CD via Weiss AFI1 firewire interface -> AES digital output to our Crookwood mastering console -> loop back to Lynx AES16 digital audio soundcard -> capture input by WaveLab software.

    They are identical. No matter it is ripped by EAC, Plextools, Wavelab, or by PC iTunes, QuadCore Mac iTunes, Powerbook iTunes, USB 1.0 external Asus DVD drives, they are all bit transparent.

    In the file comparison tests, we prove iTunes 8 and up to date computer/CD/DVD drives, are capable to rip bit perfect data same as EAC, Wavelab and Plextools.

    ------------------------------------------------------------------------------------------------------------------

    about itunes version 8 :
    http://designwsound.com/dwsblog/hifi-computer-faq/cas-7-itunes-pc-81110-problem/

    ITunes PC & MAC Version 8.1.1.10 provides different results.

    Mac iTunes 8.1.1.10 produces bit transparent results on all our tests: 16/44.1 & 24/96 & 24/192

    PC iTunes 8.1.1.10 produces bit transparent result on CD format (16bit/44.1kHz). The digital data higher than 16bit (20bit-24bit) or sampling rate (96kHz & 192kHz) are changed by PC iTunes playback engine. ***(UPDATE: you can get bit transparent playback on iTUNES PC by setting up the correct QuickTime Preference setting. More info please read CAS8 : http://www.designwsound.com/dwsblog/?page_id=1720)***

    The following test is using Wavelab as playback and record engine on 24bit/192kHz with bit transparent result.

    wavelab-loop24192


    07-16-2016 Post does not mapped to Knowledge Tree
    wolfy
    Posts 6
    Joined on 07-16-2016

    Post #: 100
    Post ID: 22684
    Reply to: 22682
    Less is more
    fiogf49gjkf0d
    My understanding of all this is rather limited.  In my experience, much as I would like and occasionally do appreciate about the convenience of computer based server, the simple cd/dac interface is the most reliable and provides the best quality sound.
    PC based music server has too many variables and too many layers between the bit and the sound which increases problems and troubleshooting exponentially.
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