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HOLO Audio Spring DAC - R2R DSD512


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  • 2 weeks later...
2 hours ago, barrows said:

No, this is a myth.  And, when considering digital volume control, at full scale, it is not "there" at all.  In other words, there is no compromise having a digital volume control at all for those who duo not use it.

 

Then, where you attenuate the volume?

 

From A+ & Hq Player (the two players I use) there is availability for software volumen control.

 

From my own experience the analogue volume control in a very good preamp is the most expensive part !

 

My only and main fear is to destroy the speakers using digital control, because it is easier and more practical through a preamp. I know that this is a very discussed topic previously, but it is worth analyzing it again!

 

A question to @Miska too ...!!!

 

Best,

 

Roch

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2 minutes ago, Miska said:

 

So far I've only used it mostly with Linux, never on Mac. So never tried it with DoP...

 

No problems so far on Linux.

 

So, you have your Mac PC running under Linux?

 

I refuse to learn Linux like Zamenhof 's Esperanto language O.o

 

Best,

 

Roch

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  • 2 weeks later...
8 hours ago, doraymon said:

I need your help to clarify something about the output voltage of my Spring L3.
As you can see in the picture below, the MAXIMUM balanced output voltage should be 5V rms (PCM).
Is this valid for each of the two channels? Or each channel has 2.5V for a total of 5V?

Also, I am confused by the fact that in the specs for PCM they mention only 48k NOS, what about the other frequencies?
For DSD the maximum balanced output voltage is half, I guess this depends on the SPring's architecture?

 

I have already asked Tim of Holo USA and waiting for a reply but in case anyone already knows that would help!

(Just to put this in context, I am asking this because my amp has an attenuator switch to handle high voltage inputs from the DAC, which seems to be the case of the Spring.
I sought advice from the amp manufacturer to decide wether switching the attenuator on or off and they asked me this question on the MAX output voltage of my DAC.)

 


Spring_output.png

 

Please wait for Tim answer... we are on Sunday.

 

Output voltage is 2.5Vrms for RCA per channel and 5Vrms  under Balanced (XLR) per channel.  More or less the same of other DACs I know.  This is under NOS setting in the DAC (no upsampling).

I guess the the other output voltages in DSD 64X is for DSD setting in the DAC.

I use NOS only.

 

Regarding frequencies you can play all you need, from 44.1 -> 352.8 and 48 -> 384.  Remember there are two family rates on PCM and DSD, 44.1 & 48 family rates.

 

Roch

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On 11/7/2017 at 9:26 PM, Superdad said:

Well I own a Holo Spring Level 3 and can assure you that the ISO REGEN is a terrific addition to it.  But obviously I am biased.  Yet there are a great many other Spring/ISO REGEN owners who will back me up on this...:D

 

Hi Alex,

 

Not in my case.  Maybe from an already galvanic isolated DAC (like in the exaSound) ? Or, system matching and listener taste ...!

 

I'm on Mac OS X (Mac Pro) like always, several USB cables tested with ISO Regen, but finished liking better the Lush.  Switch on the IR to I or not, ,etc.

 

But it works wonders in the SQ of the Lampizator B7 !

 

Best,

 

Roch

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8 minutes ago, doraymon said:

Yes Roch I know it's Sunday in the US, I'm not complaining but nothing wrong to post the question here right?

 

Of course, nothing wrong about to post a question.  At least you get my answer, but maybe is better to wait for Tim.

 

8 minutes ago, doraymon said:

My doubt about the frequencies was driven by the fact that in the specs the output voltage is given on specific frequency (48kHz for PCM and DSD64 for DSD).

Does that mean that the maximum output voltage varies with the source file sampling frequency?

I understand that it can vary between PCM and DSD due to the Spring's architecture but the source file's sampling frequency??

 

 

As I said before, I believe the output voltage is 2.5 & 5 (for SE & BAL) under NOS setting on the DAC Over Sampling button for PCM & DSD. The only I use: NOS.  This in all frequencies in PCM & DSD. I didn't measure the output, just but by listening.

 

The other spec (1.25 & 2.5) could be from the DSD setting on the DAC Over Sampling button, setting that I never use.

 

Best,

 

Roch  

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3 hours ago, Miska said:

 

DSD output is 6 dB lower than PCM output. Thus the output voltage is half. Which is expected, because DSD sources sometimes exceed 0 dB level. So the technical absolute maximums match.

 

I wouldn't use NOS mode for any rate lower than 352.8k. It will generate a lot of high frequency hash/images/distortion.

 

Thanks for the explanation. When I mentioned NOS was with respect to the button in the DAC, I always do it by means of software, HQ Player or A +, but with this DAC I sometimes find excellent recordings that practically do not require upsampling. Not many, but there are.

 

The 6dB difference between PCM and DSD is well known to DSD lovers (like me), but I find the explanation in the published specifications for this DAC somewhat confusing.

 

Best,

 

Roch

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1 minute ago, pkane2001 said:

 

10x better in what sense? I understand the theoretical benefit, but what's the practical one? I certainly don't hear a 10x better sound quality in NOS mode O.o

 

I feel the same as Ted, and also playing the Holo by A+....

 

But this is in the very individual taste territory, where nobody is wrong, nobody is right, but happy :)

 

Roch

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7 hours ago, Miska said:

 

For proper reconstruction of RedBook, the analog filter would need to have 96 dB attenuation by 24.1 kHz, but nothing yet at 20 kHz. So 96 dB attenuation in 4.1 kHz wide band. That won't happen, not even nearly. Only practical way to do it is a digital filter.

 

Another thing is that the "ringing" is originally created when the recording is created. Either at ADC when if it is running at 44.1 kHz, or at later time when for example 96 kHz master is converted to 44.1 kHz for distribution. You need to have an apodizing upsampling filter to later change/reduce it.

 

 

That way you'll also get 3 dB HF roll-off... And quite a bit of timing uncertainty for transients, depending on where sample happened to be in regards to transient timing.

 

This is how 19 kHz sine wave looks at 44.1 kHz sampling rate from a NOS DAC:

musette-19k-44k1.thumb.png.668d6989705b4964c53eeb5728e55e84.png

 

And same after upsampling to 384k sampling rate:

musette-19k-384.thumb.png.bc17fd2da7f522d3d7543a435173ab30.png

 

Makes quite a bit of difference?

 

P.S. If you have recordings in DXD or DSD, you can play it through as-is and it'll come out just fine... And you don't have any of the problems compared to reproduction of RedBook.

 

Fully agree with you !

 

But finding a lot of well-recorded music still persists :D

 

Roch

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29 minutes ago, barrows said:

But, although many DACs do not have a "NOS" mode, if you feed DSD 128, 256, 512, or PCM 352.8/384 to most DACs, what will happen is a "bypass" of the first filter stage.  You do not need a switchable "NOS" mode to take advantage of external oversampling, or to defeat the first filter stage in most DACs-just by feeding a DAC an already oversampled signal you will bypass the first onboard filter stage.

Now in many DACs there will be a second oversampling stage, which will not be bypassed by feeding 352.8/384 PCM, but this stage, if well implemented at all, will have virtually no audible consequences (as all oversampling artifacts at this high a rate of oversampling will be in such high frequencies as to inaudible).

 

Maybe the problem is that most of the music that I have and I like is in 16/44 ...!

 

Roch

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  • 4 weeks later...
5 hours ago, TubeDriver said:

Has anyone tested whether the Holo Spring inverts absolute polarity?  In my system, while it sounds very good, it does have a sound (especially with bass response in room) that reminds me of my older DAC when absolute polarity was reversed?  Just wondering...

 

I think not, but hard to tell with a lot of out-of-phase recordings. And even worse when they use many microphones, some in phase and others not.

 

Roch

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  • 2 weeks later...
16 hours ago, Miska said:

 

I wouldn't be able to use SE in first place between pre- and power-amps, it would be too noisy. There's 5 m of cable...

 

My preamp has three balanced inputs and five SE inputs. All are in use... Holo Spring, T+A DAC8 DSD and RME ADI-2 Pro are connected through balanced. It is actually hard to find preamp with more than eight inputs in total!

 

I have a Conrad Johnson pre, SE only as all CJ products, then I managed to 6' (183 cms.) interconnects. I have a wood rack with 3 shelves, then it's easy to me not to exceed interconnect length.

 

On 5 meter I agree you need XLR / Bal cable.

 

Roch

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1 hour ago, Bimmer100 said:

I apologize for bringing up the subject of balanced and SE. good lord we have gone off on a tangent when the topic is suppose to be the Dac. Lol

 

I should have known better. :) 

 

Anyway Tim, the Holo is so good that his intrinsic SQ will be more than OK to 'balanced' people (like you) and to the 'unbalanced' (like me).  The rest is only a matter of taste, system matching or whatever!

 

Roch

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  • 10 months later...

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