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 1 
 on: February 01, 2015, 09:16:17 pm 
Started by Jim_F - Last post by Jim_F
Nick,

You wrote:

> But again re reading your first post I think the laptop is still
> reading its disk but there are newly introduced SPDIF and upsampleing
> steps in the data path.

I would suspect that in fact it's not using the disk at all (that is, on the laptop that's "bridging" the RME Babyface USB input to the Phasure NOS-1a USB output via the built-in Windows 7 playthrough capability).

Remember, there are actually **two** computers in the system I'm describing.  The computer that has the music files and the music player is the one I usually use for audio playback -- an HP minitower (quad-core i7, with Windows 7 64-bit).

The (current) architecture of the whole system is not unlike that of a dual-PC system using HQPlayer on a PC connected over a LAN to HQPlayer's NAA (Network Audio Adapter) on a second PC that's actually performing the job of feeding the audio to a USB DAC (or the similar arrangement using JRiver Media Center on one PC connected over a LAN to a second PC running Jplay and feeding audio to a USB DAC).  Except that instead of using Ethernet to connect the "control PC" to the "audio PC", I'm just sending the data via S/PDIF (or ADAT).  And it's the RME Babyface USB interface that's getting the ADAT (or S/PDIF) -- the laptop only sees USB data coming in and going out; it's certainly not doing any sample-rate conversion, and it's not reading or processing audio files.

I've actually got a sort of "optical bus" running around the house.  The source PC (running foobar2000, usually) is connected over Firewire to an RME Fireface 400 that's clocked by an Antelope Isochrone.  The Fireface sends optical output to the first audio-system "station" on the bus -- the "station" is either an Apogee Big Ben or an RME ADI-192DD that takes in the optical and then passes it along to the next "station" (the optical links are up to 50' Hosa glass cables).  The Big Ben or ADI-192DD can then feed a DAC, or a signal processor (like the Purcell), etc., etc.

I also installed that "Fidelizer 6.5" program on the laptop (the Toshiba laptop's also a quad-core i7 machine running Windows 7 64-bit; both machines have plenty of memory -- 6 or 8 GB, I can't remember exactly), and put Fidelizer into "Extremist" mode, so the laptop should be pretty quiet --
I don't think Windows would have any need to page to the hard drive, and I can't see why it would be putting any audio on the hard drive -- I would think the buffering would all be done in memory.

Oh, speaking of the Purcell -- I just noticed that in fact when you're doing upsampling from 44.1k to either 176.4k or 192k, with 24-bit output word length, both dither and noise shaping are automatically turned off, and cannot be turned on, at least with the firmware in the unit I'm using (these things go through multiple generations). I hadn't noticed this before (I'm used to using the Purcell in other modes, such as doing 44.1k->96k and then
reducing output word length to 18 bits to feed an Audio Note NOS DAC capable of taking 96kHz). So much for the theory in that IAR article about noise shaping!

And as far as playback glitches are concerned -- they do happen from time to time (every 15 minutes or so maybe?) and are quite audible, even with Windows 7 playthrough and Fidelizer.  They sound like buffer underruns to me -- the audio drops out, comes back, drops out for about a second or two altogther, then goes on for a long while before the next one.  There were a lot more of them when I was using Audacity to do the playthrough (and they were different -- raspberry-sounding things: brrrrrr for a second or so.  A buffer overflow in that case?).  So again, I'm not really recommending this to anybody.

 2 
 on: February 01, 2015, 06:56:31 pm 
Started by Jim_F - Last post by Nick
Ok. I wouldn't know how "hash improved" bass sounds.

And to be really honest, what description is that ?
Of course it is not important at all, but hash improved highs I would understand right away.

Peter

Peter hi I agree, more likely to hear reduced hash is the highs not in the bass. I was typing a little too fast on a tablet  Happy. Sorry for the confusion.

Regards,

Nick.

 3 
 on: February 01, 2015, 06:52:39 pm 
Started by Jim_F - Last post by Nick
Jim hi,

Love the story about the ESL63s I certainly understand why you have had them so long. I also had a used pair form many years, I was so attached to them I could not bring myself to part with them for 5 years after I had bought a set of horn speakers. They do things others just cannot manage :-).

Thanks for the description of sound. Now I read your first post again I think the laptop playing music in the pass through USB mode maybe using its disk sub system after all to read the music data.

The playback setup is sort of smart with the dCS and word clocks, really understand where your coming from with the flying pig comment, you just would not think it would work  Happy There is certainly a lot that could change the sound, word clock quality, over sampling algorithm or other things. Your post caught my eye because I'm looking at the PC's storage sub system at the moment. What you have described is the sound is sort of consistent with the PC's disk subsystem being improved or in this case perhaps removed from the replay chain. But again re reading your first post I think the laptop is still reading its disk but there are newly introduced SPDIF and upsampleing steps in the data path.


Kind regards,

Nick.



 4 
 on: February 01, 2015, 06:36:32 pm 
Started by Jim_F - Last post by PeterSt
Ok. I wouldn't know how "hash improved" bass sounds.

And to be really honest, what description is that ?
Of course it is not important at all, but hash improved highs I would understand right away.

Peter

 5 
 on: February 01, 2015, 06:18:21 pm 
Started by Jim_F - Last post by Jim_F
Nick,

You wrote:

> Great post and a result getting a recombined 192k stream into your NOS1a.
> I would be interested to hear what the resulting sound is like compared
> to playing at 192k directly to the NOS1a. . .

Well, I'm not actually seriously recommending a setup like this to anybody.  My intention was more along the lines of announcing "Look at the flying pig!" and suggesting a way in which somebody with a lot of electronics at their disposal (USB interfaces, digital format converters, what-have-you) can play with their toys.

But as to sound quality -- well, this might be rather system-dependent in my case.  The system I'm experimenting with is rather a "dog's dinner", as the British say.  My Quad ESL-63 electrostatics started showing their age a while ago (I've had them for more than 20 years, and I bought them used) -- they started arcing-over occasionally, while turned on but not playing, in response (it seemed to me) to the downstairs neighbors' cooking smells (like an audio smoke detector, if I'm not totally imagining the correlation).  So it's time to have them cleaned and/or serviced and/or refurbished (and/or recycled -- but I don't want to think about that ;-> ).

Anyway, in the meantime I looked around for a pair of relatively modest speakers that wouldn't be too embarrassed to stand in for Quads, and I bought (used, unheard) a pair of Paradign Signature 2 v2 (with the beryllium tweeters).  These are being driven at the moment by an unmodified Carver Pro ZR1600 (a Tripath-based amp -- full-range Class D -- that some folks were raving about 11 years ago) whose balanced inputs are being fed by an Audio Experience (a Chinese direct-sale brand) "Balanced A2" tube balanced line stage.  (I've basically retired my tube amps.  Unless I experience a change of heart, which I suppose is always possible, I'll be using full-range Class D [Tripath is defunct, but ICEpower, Hypex UcD et al. are still around] for power amplification, and tubes only in line-level preamp stages, for the foreseeable).  So anyway, like I said, a bit of a dog's dinner.

As I mentioned, I've been experimenting with software upsampling ever since I stumbled across a computer program called Eximius DVD2One more than a decade ago.  Most recently, I've been playing with realtime upsampling in XXHE, HQPlayer, and foobar2000 (with the SoX plugin).  And even more recently I've been playing around with an iFi iDSD Micro, to see what upsampling to DSD sounds like (in HQPlayer, foobar2000 with Maxim Anisiutkin's ASIO Proxy driver, or offline with Yuri Korzunov's -- Audio Inventory's -- "AuI ConverteR 48x44 PRODuce-RD").

In fact, it was after hearing the DSD results that I got really annoyed with my collection of upsampled PCM files, so just on a lark I decided to try out the dCS Purcell again -- to go back to the unit (or at any rate the consumer successor to the dCS 972 pro unit that started it all) that created the "upsampling" juggernaut back in '99 (Jonathan Scull's review of that in _Stereophile_ remains to this day the most affecting piece of audio porn I've ever read, http://www.stereophile.com/digitalprocessors/260/ ).

So what's the difference with the Purcell (in comparison to a batch of files I upsampled with iZotope's 64-bit SRC to, variously, 192k, 96k, and 176.4k -- [settings: Steepness 4; Max filter length 2,000,000; cutoff scaling 1.28; Alias suppression 200.00; Prering 0.00%]).  This is totally subjective, and may have more to do with my current mood than anything else, but -- a bit more midrange (not getting lost between the bass and the treble), plenty of ambience, but with a bit less "hollowness" than before, and most importantly, a bit less sharpness in the upper midrange (especially on piano tone -- it's important to be able to listen to Alfred Brendel play Mozart piano concertos on Philips without getting a headache ;-> ).

It's perfectly possible one might be able to match the sound of the Purcell with a different software upsampler, or with different settings.  There's been a lot of discussion on Computer Audiophile (and even Hydrogen Audio) about trying to match the characteristics of Ayre's "apodizing" filter by playing with settings in SoX or Izotope to minimize pre-ringing, use minimum-phase filtering, etc., and I was going along with all of that.  Maybe I was barking up the wrong tree (I doubt if the Purcell follows that mantra).  On the other hand, dCS's filter algorithm (**whatever** it is) **was** the one that got everybody's juices flowing in the first place, 15 years ago (assuming it wasn't just a marketing stunt that all the reviewers got suckered in by).  On the other hand, I've seen an article on line from International Audio Review ( http://www.iar-80.com/page21.html ), presumably from back around the turn of the century, that claims that the Purcell's "magic" had nothing to do with upsampling per se, but everything to do with the noise-shaping that it applies (which is actually a user-selectable option) when converting back to fixed-point output after doing its internal arithmetic).  I have no idea if the author of that article knows what he's talking about (I fear the worst ;-> ).  So who knows?

 6 
 on: February 01, 2015, 06:13:38 pm 
Started by Jim_F - Last post by Nick
Quote
Hash improved bass,

Was this intentional, Nick ?
Seems like a nice phenomenon. Wink

No not intensional  Happy, just what changed with the sound ?

There is another large difference between the two test setups that is not being considered. Intermodulation of the two word clocks is going to have an impact sure but there is something else. Think where the data originates from in the two setups. Its very different in the two cases, there is a whole subsystem of the PC not being used when the pass through setup is being used. This really really matters :-).

Regards,

Nick.

 7 
 on: February 01, 2015, 05:52:51 pm 
Started by Jim_F - Last post by PeterSt
Quote
Hash improved bass,

Was this intentional, Nick ?
Seems like a nice phenomenon. Wink

 8 
 on: February 01, 2015, 05:51:11 pm 
Started by Jim_F - Last post by PeterSt
Jim,

Quote
So, sooner or later, depending on how closely the clocks are matched (and the master clock in my system is an Antelope Isochrone, so I imagine the two clocks are pretty close, but still not absolutely identical), there will be a playback glitch eventually, because a buffer in the computer will either overflow or run dry.

This description is pretty much what I recall. Could work for a while and then disotortion started or whatever. But the "while" is key.

Btw wanted to look into it again today but I forgot.

Thanks again,
Peter

 9 
 on: February 01, 2015, 05:46:46 pm 
Started by Jim_F - Last post by Nick
Jim_f

For sure two word clocks will run out of synchronisation and buffer glitches must happen as a result at some point. I have plenty of experimental evidance of the impact on sound quality that relative clock speed in the replay chain (5 clocks looked at so far more on the way) has on sound quality. This is both within the PC in what would have been considered the data domain and withing the DAC itself.

My guess however is that the word clocks running at different speeds is a secondary effect here (unless the two word clocks are grossly out that is, think 100s to 1000s of beats per second out).

There is another significant difference between the two setups you have tried that is likley to be the prime reason that the sound is changed. It would be helpful if you can you post your impression of how the sound has been changed ?

Hash improved bass, clearer highs, more detail, dynamics ?

Thanks,

Nick.


 10 
 on: February 01, 2015, 05:15:50 pm 
Started by Jim_F - Last post by Jim_F
Peter,

You wrote:

> What I recall is that when W7 was around XXHighEnd did not support KS
> yet, and one of the first things I tried was that PassThrough. So must
> have been WASAPI. However, it did not work well enough. . .

One thing that occurs to me is that with the setup I'm using (with Windows 7 playthrough), there are two independent clocks -- the one in the S/PDIF or ADAT coming in, and the local clock in the Phasure NOS-1.  So, sooner or later, depending on how closely the clocks are matched (and the master clock in my system is an Antelope Isochrone, so I imagine the two clocks are pretty close, but still not absolutely identical), there will be a playback glitch eventually, because a buffer in the computer will either overflow or run dry.

I know there are DACs out there that are deliberately designed to work this way (Doede Douma's DDDAC comes to mind, and there are others) -- "exotic" D/A boxes that clock their DAC chip locally rather with the extracted clock from incoming S/PDIF, and that let the two clocks freewheel against each other, glitches be damned (for the sake of the sound quality in between the glitches ;-> ).

It's considered bad engineering practice, but so was NOS once upon a time (and still is, at least at 44.1k).  "Good" engineering practice would be to put an asynchronous sample-rate converter (like a CS8420, AD1896, or SRC4192) in between the two clocks.

There was an amusing thread on Computer Audiophile a while back in which "Miska" (of Signalyst/HQPlayer fame) debated with a non-technical audiophile about this sort of thing.
http://www.computeraudiophile.com/f6-dac-digital-analog-conversion/mytek-stereo-192-a-5555/index113.html
The audiophile was using a setup with an independent clock in the DAC, and raving about the sound quality, and Miska was trying to explain why this inevitably leads to glitches and is considered bad engineering practice.  They never did reach a mutual understanding.

The answer to this conundrum is that a "civilian" audiophile is perfectly entitled to put up with glitches at home for the sake of the sound in between, but in a studio or other professional setting this would be considered completely unacceptable.  But even the "civilian" audiophile is not entitled to complain about the glitches, under these circumstances.

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