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Chords New M -Scaler


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37 minutes ago, ElviaCaprice said:

If one excepts the premise that the upsampling and filtering of the Chord M-Scaler is far superior in SQ to any software approach via server, then I would be interested in why that is and what prevents the software approach via server from duplicating that accomplishment?  

The answer is nothing.  It is easy for the exact same approach as used by Chord to be accomplished in software.  In fact, I suspect that Jussi (miska) believes the oversampling/filters he has developed for HQPlayer are actually superior to what Chord does in Blu/Mscaler.  Rob Watts could certainly reproduce exactly the same (or even better, 2 million taps would be easy for a decent computer) via using his approach in a computer.

With Chord offering these as hardware solutions, we have only one way to get to somewhere, there is no reason we cannot to the same (or an even better) place through oversampling in software.  Anyone who believes differently is just fooling themselves, all it is required would be to run the same code int he software oversampling program.  Chord, of course, is a hardware company, so they offer their approach in hardware.  

If one wants to reproduce what Chord does in DAVE via software, take a look at the Stereopjhile test measurements of DAVE's filter response, and make an analogous filter in iZotope with Audirvana +, You will have to use isotope's RX-6 to get a graphic representation for comparing the results of the settings, but you can get very, very close to the same response as to what DAVE does.

 

Of course a listening test requires one to listen to other aspects of the DAC's performance, all aspects of sound quality are certainly not governed solely by the filter response-so how to do comparisons?  You cannot remove the other aspects of the DAC's performance from the equation.

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10 minutes ago, ElviaCaprice said:

This is where I would disagree.  A renderer is less flexible than a single server due to it's confines to lacking a flexible OS.  A well implemented server with clean power and clocking is far more flexible to stream to an M Scaler.  And until a software solution only upon a server is shown to match the performance of a hardware solution such as M Scaler, it will not be the desired approach, let alone the complexity of setting up that software for the common audiophile.    

In turn this is/will bode well for Chord DAC's and future DAC/AMP designs for driving speakers/headphones direct, bypassing any need for separate pre-amp/amp.

Say What???

 

My post re a Renderer was about using HQPlayer, first of all.  Second of all, a good Renderer will give better performance with either an mScaler or a DAC.  A full featured server will never achieve the lower noise footprint of a well designed, high end, Renderer.

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27 minutes ago, ElviaCaprice said:

Debatable at best.  I would say it's already been surpassed, the renderer.  Due to it's reliance on the preceding stream initiation.  It is becoming more and more an unnecessary fixer component that only adds inflexible complexity.

Just factually wrong, this is not debatable at all.  The noise footprint of a full server is easily measurable to be way larger than that of a well designed Renderer.

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5 minutes ago, ElviaCaprice said:

What facts?  Show us the listening test results and the comparable measurements.

You can choose to believe what you want, but that does not make it true.  This is simple stuff here, I have made the measurements, but one hardly needs measurements anyway.  It is pretty easy to see why a well designed, high end, Renderer has a lower noise footprints than any server, just power consumption alone will answer this for you, without even having to go into details such as onboard regulator types, or processor power, etc.

How do you suppose that a full computer using say even a low 60 watts, is going to produce less noise than a Renderer running on 5 watts?

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15 minutes ago, mansr said:

That difference only matters if the noise is coupled through to the DAC output at a level that is a) audible and b) correlated to the power consumption. If the computer manages to keep the noise contained, or the DAC can keep it out, your argument goes nowhere.

And then, for those who believe that the noise does not matter, they may as well just run a standard laptop into their DAC via USB, I am fine with that if that is what they want to do.

 

But I know form experience that doing so will leave a lot of audio performance on the table, I have have never heard a single DAC which does not produce significantly better sonic performance when fed from a high end, low noise source component.  It would be nice if DACs were indeed immune to source component quality, but in the real world I have never come across one that is.

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deleted, Chris is right...

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2 hours ago, ecwl said:

I think system synergy sometimes matters more. With class D amplifiers, the switching frequencies are usually in the 400-1000kHz range. If DAVE's transient timing is more accurate because of the upsampling to 768kHz 24-bit, that transient timing accuracy is going to be slightly lost due to switching frequencies of the class D amplifiers.

That point of view is well known to be nonsense.  Rise time of any decent class D amplifier well outstrips what is possible from the loudspeaker.

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Higher sample rates also allow for a very simple, discrete component, D/A conversion stage.  Such as Chord's Pulse Array, or Jussi's DSC-1 approach.

Many people feel there are sonic advantages to these simple, discrete component, conversion stages.

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25 minutes ago, esldude said:

I'm not looking for feelings.  I liked to know more than that. 

 

 

Then you should likely listen to one yourself.

 

Of course some of these style (simple, discrete) DACs are among the best measuring DACs for distortion and SNR, such as the Mola Mola, and DAVE.

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I do not really think measurements are going to show us anything here.  All the design approaches being discussed have examples with convincing measurements which generally say the approaches are doing what they say they are doing.  Certainly Rob Watts/Chord does.  The questions are really psychoacoustic: which things really matter (most) for best perceived sound quality.

Does the M Scaler offer what is important for sound quality?  Clearly it does what it is saying it does (as measured), but does that approach equate to the best sound quality?  To Rob Watts POV on psychoacoustics it does.

But, is that approach to psychoacoustics the "right" one... or the only "right" one...

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4 hours ago, Rexp said:

I think that the audible distortion is more often intrinsic within the recording rather than than caused by the DAC/conversion, what do you think?

It does not matter.  As audiophiles we have no control over how good the recording is (or not).  We can only try and find the best versions of the music we love.

So, we are concerned with the playback equipment and reducing its (unwanted) contributions to the music.

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1 hour ago, jabbr said:

 

There is little question that “jitter” or phase noise is a strong function of the IC chip and particular logic family. Lots of focus on getting a clock with perfect specs ...

I do not doubt the validity of the above point, but would also suggest, that given a specific "IC chip" and "particular logic family", IE what one is already dealing with,  a way to improve performance can be applying batter a better masterclock (given that the details are attended to).

In no way I am I suggesting avoiding the details of the former in favor of the latter.

In my experience of improving the masterclock (and I do not refer to applying an external clock here, but replacing the existing masterclock(s) with one of lower phase noise), I have yet to do so and not experience an improvement in sonics as a result.

 

I still have not got my hands on any of the really impressive XOs yet though (ones significantly better than NDK 2520 SDA series or Crystek 575 and 957 series, all of which are pretty closely grouped in general for close in phase noise).

 

 

 

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19 minutes ago, mansr said:

Perhaps they don't want that.

I do not think it would really matter in the end from a sales perspective.  Those who are going to prefer their Chord gear will still do so.  I have worked in the high end for long enough to know that manufacturers are not sitting around and scheming about ways to dupe people, manufacturers are trying really hard to improve system performances and sound quality, and to do so also by making products which can distinguish themselves int he market place: that Chord has a somewhat unique approach which accomplishes that.

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I would expect the M Scaler to include Chord's isolated USB interface which they use in their non-portable products (Qutest, Hugo TT2 and DAVE) vs the non-isolated interface used int eh Mojo.  My understanding is that RW feels the portables do not need the isolated interface as they will be hooked up to a portable player/computer running on batteries.

 

Bottom line, I think we can be confident the M Scaler will have a very good USB interface, as does DAVE and Qutest.

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  • 1 month later...
2 hours ago, Advieira said:

How do you guys compare PSAUDIO Direcstream with Hugo 2/Qutest "scaled" by Blu Mkii dual BNC?

I would expect folks to hear big significant between the DS and Chord DACs, after all, the PS Audio DS uses a single bit/DSD approach, where Chord uses a multi bit approach.

It might take awhile for people to report in on this, as the M Scaler is only just becoming available, and I doubt anyone is using the Blu with Qutest or Hugo 2. 

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  • 3 weeks later...
1 hour ago, Shadorne said:

I am unable to eek out any benefit from upsampling with Roon with the DAC 3

Right. Is not this because no matter what you do externally, the DAC 3 is always sending the ESS 9028 211 kHz?  If you cannot defeat the built in OSF of the benchmark, then you cannot hear (potential) benefits of oversampling externally...  Or can you defeat the Benchmark's OSF? 

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  • 3 months later...

We know more than that about ESS.  My understanding for the 9018 is that it first makes an 8x pass, this is controllable and can be defeated (although this where the "jitter eliminator" works, as an asynchronous sample rate converter), then there is another step of up conversion, to very high MHz levels at 6-9 bits (bit rate is user selectable).

I am almost certain, that with DSD input, there is an SDM step, but I am not sure about how the sample rate is handled.  Remember that ESS volume control works with DSD, so there is processing going on.  Although ESS does seem to remain tight lipped about exactly what happens.  The ESS chip will not convert DSD if the first stage of processing (the "OSF") is turned off (true for 9018 and 9038).  It is clear the ESS chip is not operating as simply as a discrete DSC-1 type DAC, or even as simply as the recent AKM chips in pure DSD mode.

 

That said, I am very happy with the sonics of the ESS 9038 feeding it exclusively DSD 256, so I optimized my DAC build for DSD 256 and it sounds great.  I also run it synchronously, supplying Bit Clock and Master Clock from the same oscillator such that the DPLL completely drops out.

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7 minutes ago, Miska said:

Not many DACs (end-user products) really do something like that, most just use analog filters optimized for PCM. Still they tend to perform better when running at DSD. And could be even better if optimized for DSD.

That is pretty cool if people take advantage of it.  I changed the analog filter in my DAC a bit (relaxed it for DSD 256 input).

Jussi, are the recommended analog filters in the AKM data sheet?

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43 minutes ago, Em2016 said:

So we still don’t know to what MHz sample rate?

I used to know for the 9018, cannot remember now.  There was a huge thread )probably still there) at diyaudio.com about the ESS 9018, and designer Dustin Foremen talked a lot there about how the chip operates.  My understanding is that the final rate is related to the master clock rate (which has a wide possible range).

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