Jplay : Wow

Yes.



Nothing substantial.



It isn't.

I have not heard this particular player but I have heard and blind compared two different players (one of which was Media Monkey and don't remember the other). I could pick it out 100% of the time and so could anyone else. Was it level matched to .000000000001 db. NO. It did not need to be. We could change the volume and make one louder and then the other louder and we still picked the same one.

Your off-handed "it isn't" clearly demonstrates your total bias on the subject. (Do you even have a server?). Ever compared the one he is asking about. My bet would be no.
 
From what I've been able to surmise, different player softwares use CPU in varying amounts. Higher amounts of CPU usage can inject noise into the system and cause various audible effects. Enough CPU drain can cause "starvation" to the playback system, resulting in audible ticking during playback. In fact, I experienced all this in my own system and it was absolutely audible to anyone listening.

Lee
 
Ok, forget about me saving 99 Euros...it was a nice idea.

I am sure it isn't the most accurate comparison but may be there's something to hearing differences in digital music file players with same specs - similar to what I personally hear the differences in between a Philips DVD 963SA (for that matter SACD1000) as a transport vs Emm Labs CDSD transport that shares same laser and mechanism...through same outboard DAC.
 
Aside from a few bloody noses, did we learn anything here? The more I learn about digital, the more confused I become. People swear different software makes their digital files sound better. Hell, I thought Foobar sounded better than MediaMonkey, but that may just have coincided with my switch from a tube preamp to a SS preamp. Some people swear that a digital file is a digital file and software really can’t have much if any influence.

People even argue about what sampling rate sounds better. Ever since I switched over to Windows 7 from XP Pro because of my new laptop server (thanks again Gary), my E-MU 0404 won’t lock on any higher than 96 kHz (and I presume this is caused by the Beta drivers from Creative Labs for Windows 7), and my DAC has never sounded better which confuses me even more. And I say this in a general sense as my 24/192 files are outnumbered by my less higher sampling rate files.

It’s a digital jungle out there I tell ya.
 
There is data, noise and timing, until the file is converted to analog. If the data is bit perfect we're done with that part. There is a theory that the noise of a computer gets carried to the DAC along with the data, that the noise gets in the converted audio, and that by limiting the function of the computer playing the music file, you limit that noise. People actually not only use bare bones software but hobble the functionality of perfectly good hardware to minimize this noise. Galvanic isolation between the computer and the DAC makes a lot more sense. Then there is timing, jitter. Some people believe that a fully functioning computer not only sends noise nasties to the analog system but increases jitter that effects the functionality of the DAC. I suppose that's possible, too. But an asynchronous DAC, or re-clocking outside of the computer is, again, a much more elegant and effective solution. It's more efficient, too, because you don't have to dedicate a computer to playing music.

You can get rid of the noise and timing nasties, rest easy knowing that your digital file has been purified, sanctified, cannonized and cleansed, and, at the same time, use a good piece of software like JRiver or iTunes that gives you good information about the music you're playing, a lot of options in your file mangement, pretty pictures, and allows you to come here and argue with us while you're listening with, no negative effects. Win-win. Win.

Or you can use special software, or hobble a perfectly good computer, and believe it's all better until someone comes along with a big, heavy box with pretty lights on it, which is always better.

P
 
I'm afraid that the usual canards are being perpetrated here - digital is considered to be perfectly transposed - yes if it remains in the digital domain but we are talking about a mixed domain here unless you are able to listen directly to the bits? If you don't understand this then you shouldn't be involved in digital audio. If you do understand this then the bits are bits & bit-perfect mantra are disingenuous & misleading. Yes Jplay sounds better than any playback software I have heard - the evidence is growing & overwhelmingly obvious to any who listen & have the equipment to hear it :)
 
I believe you are saying is that digital is perfect. I agree.

Your comment about "mixed domain" must refer to analog.

All media players operate with digital data. The "mixed domain" begins at the DAC. No media player controls what happens at that point. The media player doesn't even know what is there. It only talks to the software interface (digital) that communicates with the DAC. WASAPI or ASIO, for example. Up to and including this interface, there is no "mixed domain". It is purely digital.
 
I believe you are saying is that digital is perfect. I agree.

Your comment about "mixed domain" must refer to analog.

All media players operate with digital data. The "mixed domain" begins at the DAC. No media player controls what happens at that point. The media player doesn't even know what is there. It only talks to the software interface (digital) that communicates with the DAC. WASAPI or ASIO, for example. Up to and including this interface, there is no "mixed domain". It is purely digital.

Yes but your blind spot is that you don't recognise that there could be some noise riding on the digital signal that has no effect on digital processing because it is not sufficient to cause a bit misinterpretation but is sufficient at the D/A stage to effect the translation. So how does different playback software sound different - because each one has different noise riding on the digital signal. Jplay addresses this, I believe. You seem to ignore it?
 
... there could be some noise riding on the digital signal that has no effect on digital processing because it is not sufficient to cause a bit misinterpretation but is sufficient at the D/A stage to effect the translation.

The players have no ability to affect any timing or other issues with the digital signal. They just fill the buffers of the interface and let the hardware do its job.

If the hardware interface isn't perfect, that's an issue between the PC and the DAC, and it would affect the sound of both. A good DAC should take care of the problems.

I have no desire to convince you, but I think you're spreading information that is not accurate, so I thought I would answer.

I wish you a nice weekend.
 
The players have no ability to affect any timing or other issues with the digital signal. They just fill the buffers of the interface and let the hardware do its job.

If the hardware interface isn't perfect, that's an issue between the PC and the DAC, and it would affect the sound of both. A good DAC should take care of the problems.

I have no desire to convince you, but I think you're spreading information that is not accurate, so I thought I would answer.

I wish you a nice weekend.

I was wondering when you would come out with this next over-used term - it's usually expressed as "devices that are adequately designed" but I see you have gone a step further & inadvisedly used the term "if the hardware isn't perfect". So please nominate what hardware that you know of is perfect and also please demonstrate it's measured perfectness for us. This is again the problem of those who seem to think in binary - something is either perfect or it's broken - the real world is a place without these absolutes & the real world is where we reproduce our music. Dealing with this imperfection is what sets apart the great products from the me-too meh products

Have a good weekend too but please stop spreading your ill-formed & ill-advised ideas - particularly when you set out to criticise a product which actually worked in synergy with your own & wasn't in competition with you. You really have done yourself & continue to do yourself and your product no favours. Your comments reveal an inadequate view of the market your product targets.
 
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So how does different playback software sound different - because each one has different noise riding on the digital signal. Jplay addresses this, I believe. You seem to ignore it?

One thing is obvious, sending a signal to a DAC is done using a 100% analog wave.
But can software affect this wave?
An explanation might be the way the data is processed.
Is it done in bursts or is it throttled. The burst might induce periodic jitter, the throttle a constant jitter level.
However, if this is true I want to see measurements e.g the eye pattern when playing software A or software B.
Personally I don’t mind any claim about improved sound quality by any piece of software but if it does something I like to see at least a measurable difference.
 
One thing is obvious, sending a signal to a DAC is done using a 100% analog wave.
But can software affect this wave?
An explanation might be the way the data is processed.
Is it done in bursts or is it throttled. The burst might induce periodic jitter, the throttle a constant jitter level.
However, if this is true I want to see measurements e.g the eye pattern when playing software A or software B.
Personally I don’t mind any claim about improved sound quality by any piece of software but if it does something I like to see at least a measurable difference.
I put forth one possible mechanism whereby the D/A conversion could be effected - getting measurements for such is another problem altogether! There seems little desire from the objectivists to do so as they "know" it's bit-perfect & therefore perfect. The subjectivists hear the difference & don't feel the need to prove that they do. Hence the impasse. There are some that are neither Obj or Subj (the real world remember not the binary one) but I can't see this happening until we know what to measure & at what level. In the meantime, just enjoy the music

Just to complete this - asynchronous USB is theoretically meant to make the USB receiving device immune to the PC but it isn't immune as can be demonstrated! So how does this happen?
 
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I put forth one possible mechanism whereby the D/A conversion could be effected - getting measurements for such is another problem altogether! There seems little desire from the objectivists to do so as they "know" it's bit-perfect & therefore perfect. The subjectivists hear the difference & don't feel the need to prove that they do. Hence the impasse. There are some that are neither Obj or Subj (the real world remember not the binary one) but I can't see this happening until we know what to measure & at what level. In the meantime, just enjoy the music.
This test is on my TODO list! I have just been too lazy to do it. The issue is that my Audio Precision analyzer does not go very low in jitter measurements. So it may not work out but I like to see if there are gross enough jitter levels to be detected this way.
 
There seems little desire from the objectivists to do so as they "know" it's bit-perfect & therefore perfect.
Soory but I strongly disagree.
This is not "objectivist", the "bits=bits" philosophy is plain ignorance as it leaves the timing out of the equation.
I do think it is almost trivial to get bit perfect output.
To get timing perfect output is a bigger problem.
Then there are third order effects like the “noise” you mentioned.
 
Soory but I strongly disagree.
This is not "objectivist", the "bits=bits" philosophy is plain ignorance as it leaves the timing out of the equation.
I do think it is almost trivial to get bit perfect output.
To get timing perfect output is a bigger problem.
Then there are third order effects like the “noise” you mentioned.
I agree with you about the timing - I was just reflecting the attitude I had perceived a couple of posts above & it seems to be prevalent viewpoint in the more extreme obj's - but you are correct ignorance not = objectivist & I apologise to all objs
Yes, bit-perfect output is a given by now in computer audio or at least should be & as you say is trivial to achieve.
Jitter & noise are so inextricably linked that I'm not sure which is most important. I believe that we categorise a lot of distortion into the jitter bin but in reality it may well be CM noise - sibilance for instance. Now, it may be that the ultimate agent is jitter at the DAC but the cause is CM noise at the receiver? All these things need to be teased out to have a full grasp of at least one aspect of computer audio. We are at the discovery phase at the moment so guesses & claims & denials, etc are par for the course as witnessed here.
 
This test is on my TODO list! I have just been too lazy to do it. The issue is that my Audio Precision analyzer does not go very low in jitter measurements. So it may not work out but I like to see if there are gross enough jitter levels to be detected this way.

Look forward to any results you might present but I don't know if you will find it in jitter levels? It's true that if there is an audible difference then it should be seen in the analogue output but what measurement do we use & what level are we viewing down to. I contend that FFT analysis may not be the correct test method as it uses multiple samples to average over time - will it for instance show the time-smearing of different digital filters? The focus in audio measurement still is rooted in the frequency domain (except when speakers are being measured) - I believe this focus has to change & perhaps new measurements introduced.
 

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