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MAC vs PC -> DAC (Digital to Analog Conversion)


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Peter,

 

bla bla bla jitter bla bla bla jitter.

 

Come on... Tell me, in any asynchronous environment with a dac how your software is going to effect the jitter?

 

It can't... so if you start there and your wrong there then how is anyone going to believe the rest of what your saying?

 

What next are we going to ask you what kind of test equipment you have?

 

No because then we would have to know if you can use it.

 

Ok here's a question or a hypothesis of sorts...

 

If your software does effect the jitter in some kind of way that makes it less jitter output then how do you know how much to do? Like for example the jitter modulation on say my dac would not be anything like say an adaptive piece... so how do you know what modulation of jitter you should be correcting for?

 

With that hypothesis in mind the next question would be how many devices have you verified that said software lowers the overall jitter as heard by the user?

 

I mean if your perfect dac is the model, then maybe the playback for any other device would be wrong!

 

I would imagine that if I sent the JTEST out your software and compared it to say J River in exclusive mode that the results on my Prism dScope III would be pretty much the same.

 

Gang this is just advertising... I don't like it.

 

Thanks

Gordon

 

 

 

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You know, anyone will be able to test Peter's thinking by trying his DAC when it's released. I for one will be comparing it, as objectively as I can, to my DAC and sharing my thoughts on this forum. Of course, I hope my DAC trounces it, which'll save me a bunch of money. But you know what, having heard a prototype of Peter's DAC, I wouldn't bet on it...

 

Mani.

 

Main: SOtM sMS-200 -> Okto dac8PRO -> 6x Neurochrome 286 mono amps -> Tune Audio Anima horns + 2x Rotel RB-1590 amps -> 4 subs

Home Office: SOtM sMS-200 -> MOTU UltraLite-mk5 -> 6x Neurochrome 286 mono amps -> Impulse H2 speakers

Vinyl: Technics SP10 / London (Decca) Reference -> Trafomatic Luna -> RME ADI-2 Pro

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Gordon, I appreciate your concerns on behalf of readers. No problem here. What I don't understand though is why you suddenly title my posts as "advertising" (which you did not only today) while I am saying exactly the same for at least 18 months on this very forum, you always being around. I didn't change a bit of my thinking nor expressions and I only repeat what has been said many times before by me.

 

As a matter of fact I recall one of your first posts here at CA, telling that my software sounded cold on your DAC. Yes, your asynchronous DAC from back then.

 

I would imagine that if I sent the JTEST out your software and compared it to say J River in exclusive mode that the results on my Prism dScope III would be pretty much the same.

 

You, like Steve, may be short on memory somehow, since it looks like you have forgotten the debates about tests regarding exactly this. I referred to this in the other thread today, and while Steve was keen to explore further (but banned himself from here), you were keen not to help me while I asked you with the same Prism you just mentioned.

 

But I went on without your help. Sorry, but I did.

Afterwards you helped me greatly on another subject which funnily enough was about Steve's devices (over at AA). Also afterwards I pointed you at my means of filtering. No different story from today's, although nobody sees through it but you. Well, that is what I assume.

So suddenly you call this "advertising" ? Come on now, you must have better means to protect your market share.

 

About your 10 or so phrases in your post ... they almost all miss the point; Although my english won't be the best, please distinguish from what I said and what your assumptions are, you lacking my knowledge on this. Well, as it appears, ok ?

 

Thanks,

Peter

 

Lush^3-e      Lush^2      Blaxius^2.5      Ethernet^3     HDMI^2     XLR^2

XXHighEnd (developer)

Phasure NOS1 24/768 Async USB DAC (manufacturer)

Phasure Mach III Audio PC with Linear PSU (manufacturer)

Orelino & Orelo MKII Speakers (designer/supplier)

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You wrote:

 

when I questioned the lack of Hog mode, you commented that Amarra didn't need it because it wrote to the HAL ... but surely the advantage of the Hog mode in Pure Music is that it avoids any need for the mixer being utilised

 

Hey Eloise,

 

Please read my comments to idiot_savant re, among other things, “mixers” versus accumulators above [Riding an ox, in search of that ox].

 

therefore there IS a need for a Hog more as you (I think) and Gordon have both said that it's best to avoid mixers which (with Amarra) can't in some case be avoided. Likewise in Windows applications where they use WASAPI exclusive mode.

 

There are situations where hog mode is a good choice to solve a problem but, it’s not required for LPCM data transfer. The product you like uses it, other products do not because the designer has decided they can get the job done without it. I cannot speak for the folks at Channel D, maybe they would comment on why they use hog mode.

 

Designers make many, many choices when creating a product and listeners make many, many choices when purchasing a new piece of kit. A diversity of opinion make the world go ’round.

 

Regards,

______________________________________________

O.A. Masciarotte - http://www.othermunday.com

______________________________________________

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Oliver - I know you merely talk about the Apple environment, but since you responded to Eloise's "WASAPI", allow me to have a small addition;

 

In the Windows environment, when the sound device is not used exclusively - or the mixer is involved, or the least what will happen is that dither keeps on being applied. This latter happens in XP.

Btw, in XP quite some means exist to "hog" the sound device afterall, one of them being special drivers some brands have (like from RME).

 

Under Vista/W2008/W7, these "driver" means do not exist, and the only options are WASAPI Exclusive Mode ("hog mode"), Kernel Streaming (the officially never supported means to create sound in Windows) and ASIO (just going around everything and all).

 

Maybe also interesting to know is that Windows/WASAPI/Exclusive does not allow "designers choices" to use the mixer or anything. So it's really all or nothing. And as far as I know this also works the other way around : once not in any of the "hog modes" (like Direct Sound), there's no way the designer can choose to avoid mixing etc. So it really all starts with the method used (or depends on the driver used under XP).

 

Regards,

peter

 

Lush^3-e      Lush^2      Blaxius^2.5      Ethernet^3     HDMI^2     XLR^2

XXHighEnd (developer)

Phasure NOS1 24/768 Async USB DAC (manufacturer)

Phasure Mach III Audio PC with Linear PSU (manufacturer)

Orelino & Orelo MKII Speakers (designer/supplier)

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Peter,

 

You, like Steve, may be short on memory somehow, since it looks like you have forgotten the debates about tests regarding exactly this. I referred to this in the other thread today, and while Steve was keen to explore further (but banned himself from here), you were keen not to help me while I asked you with the same Prism you just mentioned.

 

Sorry but I don't read everything you write.

 

As for the rest of this post... I am just more confused.

 

Gordon

 

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I've been so far quite polite, but you seem to insist, and now are quoting your replies, so I'll make this as succinct as possible so my views are clear:

 

If your "Audio Engine" provides the same bit-perfect output as the default one, it doesn't matter what convoluted process you go through to generate that same output.

At the end of the day, if the Core Audio stack loses bits, and I quote "And, this is floating point not fixed point arithmetic, which means the low order bits (the low amplitude data) are sacrificed, again and again, on the altar of gain staging." then all the measurements thus far provided are wrong. Can you provide any measurements proving that the low amplitude data is being sacrificed?

 

The "HAL" layer will read the same buffer using iTunes or the SSE engine at the end of the day. This isn't magic, and if there is a difference it can't be due to the standard waffle people (like you) spout about 64 bit floats, dodgy mixers, mysterious Core Audio effects, etc or the data would not be bit-perfect

 

 

Additionally, Amarra cannot be using less CPU/have better latency than iTunes by virtue of the fact that iTunes is concurrently playing the same music.

 

I am not stating that it's impossible for SSE to sound better than Core Audio - indeed, if EQ/volume is used, it's quite believable, but please stop talking nonsense about mixers/summing points when the arithmetical outputs are the same

 

your friendly neighbourhood idiot

 

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You wrote:

 

I've been so far quite polite, but you seem to insist, and now are quoting your replies, so I'll make this as succinct as possible so my views are clear:

 

If your "Audio Engine" provides the same bit-perfect output as the default one, it doesn't matter what convoluted process you go through to generate that same output…

 

Hey idiot_savant,

 

Yes! I agree! I was simply trying to address your, ahem, focus, on “mixers,” latency, etc. Given bit perfect output, any noticeable audible difference between two players will be how they accomplish the process of getting data from storage to output spigot.

 

I am not stating that it's impossible for SSE to sound better than Core Audio - indeed, if EQ/volume is used, it's quite believable, but please stop talking nonsense about mixers/summing points when the arithmetical outputs are the same

 

Not nonsense, simply not germane to this discussion…thank goodness. BTW, thanks to PeterSt for your info on current designer choices for Windows.

 

Regards,

______________________________________________

O.A. Masciarotte - http://www.othermunday.com

______________________________________________

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