[romaz]

   
Wait. Did you just say this other thing is better than an Aurender W20.
 
Paul

No, because my microRendu hasn't arrived yet but that is certainly a question I want to be able to answer for myself.  I suspect the W20 is an obvious benchmark for anyone who's heard it.  Chris Connaker, in his review of the W20 last year had indicated that the W20 was the finest music server he had heard, better than even the CAPS servers he had been championing for years. While he hasn't completed his review of the microRendu, this is what he had to say about it recently:
 
"I've spent hours on end listening to music since I took delivery of the microRendu. I wanted to make sure I wasn't burned by expectation bias, so I compared it to many other sources and methods of audio playback (both blind and sighted). After all this, I can unequivocally say that with the microRendu in place, my audio system has never sounded better than right now."

 

[paulchiu]

  No, because my microRendu hasn't arrived yet but that is certainly a question I want to be able to answer for myself.  I suspect the W20 is an obvious benchmark for anyone who's heard it.  Chris Connaker, in his review of the W20 last year had indicated that the W20 was the finest music server he had heard, better than even the CAPS servers he had been championing for years. While he hasn't completed his review of the microRendu, this is what he had to say about it recently:
 
"I've spent hours on end listening to music since I took delivery of the microRendu. I wanted to make sure I wasn't burned by expectation bias, so I compared it to many other sources and methods of audio playback (both blind and sighted). After all this, I can unequivocally say that with the microRendu in place, my audio system has never sounded better than right now."

 
Even if the microRendu is equal or 99% of the W20, it's a no-brainer.
Microrendu is just 1/25 the cost of the Aurender.

 

[romaz]

   
Even if the microRendu is equal or 99% of the W20, it's a no-brainer.
Microrendu is just 1/25 the cost of the Aurender.

That's how many people feel and why it's caused such a buzz.  But to get the best from it, you will want to pair it with a very good 7V/2A linear PSU, particularly one with very low output impedance like the JS-2 that now serves as a paper weight on your desk, the upcoming "mystery " power supply by John Swanson (availability expected in August), the Sonore Signature power supply ($1,400), a Teddy Pardo power supply or a custom build from someone like Paul Hynes.  Even factoring this in, it is a bargain if it sounds as good.

 

[esimms86]

And, of course, to be complete in your comparison, you would need to A/B the microRendu with the SonicOrbiterSE whereby the SOSE is also connected to a LPS better than both the standard one it comes shipped and the ifi PS, if that's what you purchased from Sonore.
 
While many people say that computer audio is in its infancy, we may yet be in a golden era for music servers. There is some buzz about the Sound Galleries Music Server(SGMS) currently showing in Munich which converts all of your music files to DSD512 and sells for 16000 Euros(yes, you read that right). Chris Connaker will conclude and post his writeup of the microRendu soon and this will take place after his trip to Munich where he specifically comments that he had a chance to listen to the SGMS. If the microRendu can compete even respectfully against the likes of the Aurender W20 and the SGMS(I expect the jury to still be out on the SGMS since no reviewer AFAIK has had a chance to try one out in their own system), it will indeed be a game changer in this rapidly developing field. 

 

[romaz]

 
And, of course, to be complete in your comparison, you would need to A/B the microRendu with the SonicOrbiterSE whereby the SOSE is also connected to a LPS better than both the standard one it comes shipped and the ifi PS, if that's what you purchased from Sonore.

I agree with this but it' not so easy since the SOSE requires 5V and the microRendu requires 6-9V.  I will not have a top flight PSU that outputs 5V and so the fairest comparison would be to compare each using their respective stock PSUs or else to compare them both using my inexpensive HD Plex PSU which is capable of both 5V and 9V outputs.
 
 

  While many people say that computer audio is in its infancy, we may yet be in a golden era for music servers. There is some buzz about the Sound Galleries Music Server(SGMS) currently showing in Munich which converts all of your music files to DSD512 and sells for 16000 Euros(yes, you read that right). Chris Connaker will conclude and post his writeup of the microRendu soon and this will take place after his trip to Munich where he specifically comments that he had a chance to listen to the SGMS. If the microRendu can compete even respectfully against the likes of the Aurender W20 and the SGMS(I expect the jury to still be out on the SGMS since no reviewer AFAIK has had a chance to try one out in their own system), it will indeed be a game changer in this rapidly developing field. 

The SGMS will not be a good option for DAVE users and again, this is what Rob has to say about servers that upsample to DSD with respect to the Mojo but should be equally applicable to the DAVE:
 
Converting the original file into DSD or up-sampling is a very bad idea. The rule of thumb is to always maintain the original data as Mojo's processing power is way more complex and capable than any PC or mobile device.
 
DSD as a format has major problems with it; in particular it has two major and serious flaws:
 
1. Timing. The noise shapers used with DSD have severe timing errors. You can see this easily using Verilog simulations. If you use a step change transient (op is zero, then goes high) with a large signal, then do the same with a small signal, then you get major differences in the analogue output - the large signal has no delay, the small signal has a much larger delay. This is simply due to the noise shaper requiring time for the internal integrators to respond to the error. This amplitude related timing error is of the order of micro seconds and is very audible. Whenever there is a timing inaccuracy, the brain has problems making sense of the sound, and perceives the timing error has a softness to the transient; in short timing errors screw up the ability to hear the starting and stopping of notes.
 
2. Small signal accuracy. Noise shapers have problems with very small signals in that the 64 times 1 bit output (DSD 64) does not have enough innate resolution to accurately resolve small signals. What happens when small signals are not properly reproduced? You get a big degradation in the ability to perceive depth information, and this makes the sound flat with no layering of instruments in space. Now there is no limit to how accurate the noise shaper needs to be; with the noise shaper that is with Mojo I have 1000 times more small signal resolution than conventional DAC's - and against DSD 64 its 10,000 times more resolving power. This is why some many users have reported that Mojo has so much better space and sounds more 3D with better layering - and its mostly down to the resolving power of the pulse array noise shaper. This problem of depth perception is unlimited in the sense that to perfectly reproduce depth you need no limit to the resolving power of the noise shaper. 
 
So if you take a PCM signal and convert it to DSD you hear two problems - a softness to the sound, as you can no longer perceive the starting and stopping of notes; and a very flat sound-stage with no layering as the small signals are not reproduced accurately enough, so the brain can't use the very small signals that are used to give depth perception.
 
The second issue in using the transport to up-sample (44.1 to 176.4 say) is that the up-samplers in a PC or mobile device are very crude, with very limited processing power and poor algorithms. This results in timing problems, and like with DSD you can't hear the starting and stopping of notes correctly. These timing problems also screw up the perception of timbre (how bright or dark instruments sound), the pitch reproduction of bass (starting transients of bass lets you follow the bass tune), and of course stereo imagery (left right placement is handled by the brain using timing differences from the ears). Now Mojo has a very advanced algorithm (WTA) that is designed to maximise timing reconstruction (the missing timing information from one sample to the next) and huge processing power to more accurately calculate what the original analogue values are from one sample to the next. Its got 500 times more processing power than normal, and this allows much more accurate reconstruction of the original analogue signal.
 
So the long and the short is don't let the source mess with the signal (except perhaps with a good EQ program) and let Mojo deal with the original data, as Mojo is way more capable.
 
Rob
 
In case you've already forgotten, here is his response to your question back in November about upsampling:
 
Oh dear. Do NOT use your computer to up-sample or change the data when you use one of my DAC's.
 
All competent DAC's up-sample and filter internally; the issue is how well that filtering is done, in terms of how well the timing of transients is reconstructed from the original analogue. Computers are poor devices to use for manipulating data in real time as they are concurrent serial devices  - everything has to go through one to 8 processors in sequence. With hardware and FPGA's you do not need to do that, you can do thousands of operations in parallel. Dave has 166 DSP cores with each core being able to do one FIR tap in one clock cycle. That is incredibly powerful processing power way more powerful than a PC.
 
But its not just about raw processing power but the algorithm for the filter. The WTA filter is the only algorithm that has been designed to reduce timing of transients errors, and the only one that has been optimised by thousands of listening tests.
 
Rob
 
As to whether the DAVE upsamples, yes, it does and well beyond DSD512.  Here again is his response to me:
 
Oh dear!
 
No I over sample to 2048 FS, or a new filtered sample every 9.6 nS.
 
The first WTA stage is 16FS. Then the next WTA stage is at 256 FS. Then a three stage filter then takes it to 2048 FS.
 
Its done for a number of reasons - to reduce the timing of transients uncertainty problem, to enable the noise shapers to work at 104 MHz so that the noise shapers can reproduce depth correctly, and finally to allow no measurable noise floor modulation.
 
So there are a number of reasons why I oversample to such a high rate.
 
ADC ringing artifacts is not one of them, as that is irrelevant.
 
Regards
Rob
 
It seems to me that the rest of the audio world is trying to do what the DAVE already does better.  To feed the DAVE an upsampled DSD file that the DAVE would only further upsample makes no sense to me.

 

[shuttlepod]

 
It seems to me that the rest of the audio world is trying to do what the DAVE already does better.  To feed the DAVE an upsampled DSD file that the DAVE would only further upsample makes no sense to me.

 
Especially if that server will cost you roughly $20k USD.

 

[rkt31]

can't see the wav download link of those dac group test files. can anybody pm me ?

 

[esimms86]

DAVE obviously does not fit into the upsampling equation, which is why a DAC like the T+A was chosen for the SGMS demo. Likewise, the otherwise outstanding Phasure NOS1a doesn't fit as it is required to be driven by its own upsampling software(XXHE, which Rob counsels against using with DAVE for the same reason). So, yes, the processing purposely built into DAVE makes DAVE its own ecosystem with its own set of rules to play by.

Certainly, there exists a subgroup of computer audiophiles who find DSD more favorable to their tastes and also see the notion of upsampling to DSD512 an exciting option. I must admit that I love both PCM and DSD and I have easily over 100 music files in that format(some hi res downloads but the majority being files converted from SACD by a friend). What I can't dispute is just how wonderful DSD sounds through DAVE(with the DSD, not PCM, option selected, of course). Do I want to upsample any or all of my music to DSD512? No. I'm happy that something like the SGMS exists and that people are willing to push the envelope. Do I want one at even 1/10th the price? No. Am I excited that a multitude of music servers exist for various tastes, degrees of computer expertise and budgets? Unquestionably yes.

I'm learning everyday that there is no such thing as end game in this hobby. Rob and John will unquestionably be bringing more complex and musically satisfying DACs and digital amps in the future. Who knows, perhaps Rob will decide some day to produce a DAC that requires an order of magnitude more code than DAVE. For that matter, it's by no means a stretch to say that the day will come when consumer priced digital audio is inarguably superior to vinyl.

Regarding comparing the SOSE with the microRendu, keep in mind that the Uptone Audio JS-2 can be set to 5 volts.

 

[romaz]

DAVE obviously does not fit into the upsampling equation, which is why a DAC like the T+A was chosen for the SGMS demo. Likewise, the otherwise outstanding Phasure NOS1a doesn't fit as it is required to be driven by its own upsampling software(XXHE, which Rob counsels against using with DAVE for the same reason). So, yes, the processing purposely built into DAVE makes DAVE its own ecosystem with its own set of rules to play by..

Well expressed and I agree with your comments. Despite Rob's issues with DSD, he has suggested that you keep a file in its native format and so if a performance is recorded with a DSD recorder, you should keep it DSD and as you said so well, the DAVE does a wonderful job playing back DSD. Like yourself, I have a growing collection of native DSD files that I really enjoy, especially files recorded with a DSD256 recorder and as I discover more of these recordings, I will continue to buy them. I also have a decent SACD collection that I stream through my Oppo to my DAVE and they sound better through the DAVE than through my Oppo's own DAC.

What I don't agree with is the conversion of a native PCM file to a DSD file. I am not convinced this sounds better on the DAVE at all. If you look at the 2L test bench, as you know, you can download various test files in various formats, from 16/44 to DSD128+ although you will notice that without exception, all of these files were originally recorded in PCM (usually DXD). As I have listened to and compared these different file formats with the DAVE, to my ears, the native PCM recording sounds best. In comparison, the DSD sounds soft and at the expense of detail. With natively recorded DSD files, I have not found that to be the case and so this would support Rob's assertion that you should keep a file in its native format and let the DAVE handle any processing.

With that said, Rob has also been very clear that the DAVE's WTA filter and large number of TAPS are optimized for superior playback of PCM. He has expressed that with analog recordings that have been converted to both PCM and DSD, PCM (even 16/44) will sound better than DSD and I agree with this. As we know, Jazz at the Pawnshop was originally recorded via reel to reel and many jazz fans I know continue to believe that this is the best jazz recording there is. This analog recording has since been transferred to both PCM and DSD and I own a DSD128 transfer of this recording. As I compare this DSD transfer to the 16/44 PCM stream from Tidal on the DAVE, I actually prefer the 16/44 PCM stream. Again, the DSD copy I own sounds soft and less detailed in comparison. Of course, if this is how one prefers their music, that's fine but for those of us who are trying to experience the "live" from a live performance, it is my opinion that PCM sounds better. To be honest, as I listen to a really good Redbook file with the DAVE, I can't ever recall myself thinking that somehow I was being shortchanged.

As for vinyl, as the best recordings today are being recorded natively with either DSD or DXD recorders, I cannot envision how vinyl can top the DAVE with a modern recording since the analog transfer will have digital roots.

 

[romaz]

Regarding comparing the SOSE with the microRendu, keep in mind that the Uptone Audio JS-2 can be set to 5 volts.

I agree, this would be a great way to compare the two and up until last week, I had a JS-2 on order but for personal reasons, I have decided to go with a Paul Hynes custom build instead.

 

[esimms86]

Especially if that server will cost you roughly $20k USD.

I suppose this is a movement spearheaded by HQPlayer(whose designer is a major proponent of DSD) and the PS Audio Directstream DAC. In today's Stereophile report from Munich Jason Victor Serinus discusses the EMM Labs DA2 DAC which has an msrp of 25,000 USD. It uses its considerable processing resources to upsample all input to DSD1024 in real time(!). However impressive a feat that may be, Serinus found it to have an exaggerated midrange. "Smudged" is not the way you the sound of your DAC to be described by the audiophile press.

On the other hand, Michael Lavorgna had favorable things to say regarding his impressions of the SGMS at Munich.

 

[DaveRedRef-III]

I see the first major music company has signed up to MQA
http://www.mqa.co.uk/customer/news/post/warner-mqa-deal

I hope Chord join the growing list by retrospectively implementing with Dave

 

[Mython]

I see the first major music company has signed up to MQA
http://www.mqa.co.uk/customer/news/post/warner-mqa-deal

I hope Chord join the growing list by retrospectively implementing with Dave

I hadn't realised how much the aspect of timing fidelity is being emphasised, in the MQA promotional copy.
 
Since timing accuracy also happens to be a major emphasis of Rob's WTA filter approach, I am curious as to whether or not the two approaches are compatible, and I am also curious how MQA might be 'capturing' (digitising) timing details with (what seems to be being implied) greater accuracy than all other existing ADC approaches.
 
Lots of questions...

 

[DaveRedRef-III]

Hi Mython
I think for commercial reasons every hi end hifi manufacturer should be considering MQA as a bolt on because it stands to reason that with a complete MQA remastering, lesser Dacs with MQA will more than likely sound better than better Dacs without it. MQA will always have the upper hand in the equation because it has the better master for comparison purposes. I think that is why we are seeing a trend of companies like Pioneer and Onkyo providing 'free' upgrades. There is no reason that I can see why Dave would not sound superior with MQA handling than all other current Dacs. I suspect though that without it, Dave will be disadvantaged against others that can handle the MQA files.

I am part way through reading this recent lengthy report by The Absolute Sound magazine

Link from MQA site: https://mqa-production.s3.amazonaws.com/default/0001/01/6a10f3ba2385770ac3658df2cadc537ffcd09cd3.pdf

 

[Mython]

I'm neither pro nor anti MQA, in regard to whether or not Chord 'should' or 'shouldn't' implement it, or whether or not it would be competitively prudent to - I'm just curious about the technicalities underlying MQA.
 
I understand that they are accessing original master tapes (didn't Neil Young say he was going to do that, before his store just started selling all the same Hi-Res files as every other Hi-Res download store on planet earth?

)
 
Joking aside, I understand that MQA intend to dynamically 'calibrate' the playback chain (including the digital transport & DAC, unless I've misunderstood) to most accurately mimic the sound of each original analogue master tape (presumably via some kind of metadata embedded within each MQA file?). That'd be all well & good.
 
What I don't understand, however, is how the MQA approach can improve timing accuracy, unless their actual ADC itself (as a seperate entity from any subsequent codec-engineering, further along the process) is somehow a step ahead of all existing ADCs (and, as DAVE fans know, Rob happens to be working on an ADC following some of his WTA principles implemented in DAVE, Hugo and Mojo). The cleverest codec in the world cannot make a silk purse out of a sow's ear, WRT to timing accuracy, if the ADC itself is no better than any other ADC. Perhaps they are merely saying that the MQA codec retains maximum timing information available in the raw ADC'd PCM file.
 
I dunno - it just surprised me, a few minutes ago, when I followed that MQA link, posted above, and noticed, in another page of their website, that they are strongly pushing the aspect of timing accuracy in their approach, which is not something I have any recollection of seeing in their promotional literature, in months past. I can't help wondering (with a wry grin) if this might have been prompted, to some degree, by Rob's recent success in advancing the state of the DAC art, and his many public statements that timing accuracy is critically-important to convincing music reproduction, but I'm just amusing myself, with that thought, so don't take it too literally

.
 
 
.