[x RELIC x]

These two videos of one of Rob's talks should explain.

Part 1

Part 2

Thanks @Triode User.

Part 2 is along the lines of what I was thinking, especially in his first sign wave illustration of the problem when reconstructing transients. The key is the sinc function filter with infinite sampling in order to properly reconstruct the original waveform from sampled data. This all based on proven sampling theory. If I recall correctly Rob’s WTA filter improves a basic sinc function by an order of 10x so the 1 million TAPs are that much more effective.

 

[simorag]

I’m asking for a more detailed explanation of what the Mscaler is doing to the file... extracting more data from it? Lessening the amount of data that’s lost? And whichever (or both) it’s doing, at the exact same rate of resolution in theory would the files sound the same or different if they were a native high-res file or if it was upscaled by the mscaler?

The good thing is that, according to digital sampling theory, it is possible to rebuild the exact initial analog data from the sampled data by using an ideal (sinc) interpolation function. This means that you could indeed "extract more data" from a digital file (redbook or higher resolution) by filling the gaps between samples with zero error compared to the original analog signal which was recorded.

The bad thing is that using an ideal interpolation function would take an infinite computational processing capability, so we need to live with a certain error, and all DAC designers have their take and preferred approach to how to reduce this error as much as possible.
​

The ultimate achievement of the MS (+dual BNC DAC) is to rebuild the original analog signal with an accuracy better than 16bit (16.6bit actually).
This in turn produces a timing error which is close to - or better than - the threshold of human brain (a few microseconds according to literature) of discerning transients.
In order to do this, the approach selected by Rob is to use a very long (1Mtaps) sinc function interpolation filter, and to upscale the sampling rate to 705.6/768kHz.

The true "magic" of M Scaler resides on using its full 1Mtaps filter length / WTA combination algorithm, because only by doing so you would bring the timing accuracy to a level where the brain should not be able to discern transient errors. RW insists that transient errors are at the root of all evils for audiophiles because they degrade also timbre and bass pitch perception.

Therefore, to answer your questions based on my (admittedly limited) grasp of all this matter:

a- yes, in a way (see above) the MS 'extracts more data' and 'lessens the data that's lost' from files with resolution lower than 705/768kHz;
b- if you had a native file of a certain resolution (say 24/192), and use the MS by starting from a lower res one (say 16/44) to upscale "only" to the native level (i.e. to 24/192 in this example), the MS would not produce any advantage, as I understand it
c- in today's world, native files at 705.6/768kHz are not easily available to us consumers, so by using the MS basically all practically available (PCM) digital music can be improved to some extent

As an M Scaler owner, I can testify that I can hear improvements from the HMS even when playing high res (e.g. 24/192) files, and that there is a marked difference from using the HMS at full throttle (1Mtaps, hence everything upscaled to 705.6/768kHz) compared to using its upscaling capabilities only partially (which you can do by adjusting the settings of the HMS OP SR button to passthrough - low - medium - high).

In addition to the videos already linked by @Triode User, here you find some other interesting material:

https://www.head-fi.org/threads/watts-up.800264/page-12#post-13150760
https://chordelectronics.co.uk/wp-content/uploads/2018/07/The-theory-behind-M-Scaler-technology.pdf

Hope I have not posted anything misleading, if so other more knowledgeable people and of course Rob Watts can straighten this out.

 

[apmusson]

Nick, all the classic soundstage expansion cues: better height resolution, then adding in significant more depth resolution, more relaxed presentation. Absolutely delightful (and a delightful surprise to know that there is even more SQ to reveal...mScaler is the gift that keeps on giving).

I have been hearing similar things with my end point experiments: NUC running real time OS (Audio Linux) in memory with Squeezelite with extremely large memory buffers on the network side and the ALSA driver side. The SQ lift from these experiments (significant) is very similar to what I hear with the ISORegen/LPS 1.2 combination. In fact, by chance I had 2 ISO Regens in house, and when I put them in series, it is the closest I've come to not hearing a difference with the end point optimizations I've been doing (which is awesome).

Clean power on Vbus to HMS has great intuitive appeal to me (LPS 1.2 on the ISO Regen), and I get squint my eyes and say "yeah, that makes sense that it makes a difference". The only hypothesis I have for how the hell real time end point optimizations or multiple USB signal regeneration steps can be so audible is some sort of signal integrity induced noise on the USB receiver side. If ever there was a clutching at straws hypothesis, this is it. Much more thinking and experimenting to do to see what is really going on, but the pragmatic result is awesome.

In the interim, Alex and team are extremely generous with their home trial set up (pay for shipping, try it for 30 days). If others try ISORegen and LPS 1.2 with their mScaler setups, please share your experiences. I don't know how much of what I'm hearing is unique to my setup and house, and what may be tickling at a more general noise vector. I look forward to hearing what other people are hearing.

Hi Ray,

I'm now also using audiolinux and a NUC but with optical connection to my HMS.

I already have an LPS 1.2 but would you recommend USB via ISO regen over optical?

Ade

 

[PY034]

Need some advice/help here.

I ordered a bluesound node 2i and thinking of connecting it to HMS using optic cable. Just wondering if I can use RCA output of the node2i to feed the BNC input?

 

[simorag]

If others try ISORegen and LPS 1.2 with their mScaler setups, please share your experiences. I don't know how much of what I'm hearing is unique to my setup and house, and what may be tickling at a more general noise vector. I look forward to hearing what other people are hearing.

I have the ISORegen (IR) with LPS1 (grounded via iFi Groundhog) and my experience is similar to what @ray-dude reported.

I initially purchased the IR when I had an Audio GD NOS11 DAC/amp connected to a laptop, and after each upgrade I made to my digital front end (DAVE, ZENith SE and M Scaler) and interconnects I tried to remove the IR for the sake of reducing the number of boxes, cables etc. (including their induced noise, distortion), but I always ended up putting the IR back in the chain.

While the effect is now more subtle than when I was using it within in my older, less sophisticated setup, it is still worthwhile.

Namely I am hearing a tighter bass, more solid imaging and better defined soundstage depth, cleaner decay with the IR.

Conversely, it am not hearing significant effects on treble smoothness / digital glare with my current setup.

 

[dmance]

Hi Ray,

I'm now also using audiolinux and a NUC but with optical connection to my HMS.

I already have an LPS 1.2 but would you recommend USB via ISO regen over optical?

Ade

Ade,
I'll jump in here to offer my reply. Ray has many more hours of evaluation than I do on this topic, but I just wanted to (again) clearly state that the changes in the character of the sound are due to analog noise/disturbances not digital errors. My way of rationalizing what's going on to the digital signal that can affect SQ is based on:
- electrical leaking current movements (ground loops or gradients) ...solved by going optical
-jitter in the waveform - solved by the reclocking of all Robs DACs
- high frequency noise (spikes, overshoots, ripple) on the waveform ...mostly fixed by attenuation or modulation of various sorts.

But then there are the head-scratchers ...like when a better PSU on a NUC waaay upstream of a DAC and through an optical bridge is still audible. There are some people on the track of what this may be (phase pumping of energy). I've also got my own ideas.

So, put your DAC on an island of isolation. They are incredibly sensitive and our ear/brain picks up almost anything you can do to improve isolation.

 

[ray-dude]

Hi Ray,

I'm now also using audiolinux and a NUC but with optical connection to my HMS.

I already have an LPS 1.2 but would you recommend USB via ISO regen over optical?

Ade

Ade, optical input to HMS is outstanding. I’m glad you got your setup working. If you have primarily redbook or Tidal content, sit back and enjoy.

As Dan mentioned above, there are tweaks even for optical, but for me they are maddening to contemplate (how the heck can this possibly be happening?) I’m in full denial as I eagerly wait for Audionauts like Dan to emerge from the madness with an answer

I bounce back and forth between USB and optical, using optical as the consistent baseline as I evaluate the more complex USB chain/experiments. With that, I’d say my answer to your question is a very unsatisfying “give it a try and see which you prefer”

To pile on what Dan said above, none of these tweaks have any impact on the 1’s and 0’s. Unless you have defective cabling or some other gross issue, a bit perfect digital signal is getting to hMs perfectly in all scenarios. If it wasn’t, you’d be hearing major drop outs and loud clicks.

This is all about either transmitted noise or induced noise that is somehow getting through to your DAC (my guess right now is what Dan and I are hearing on optical and USB is some sort of signal integrity induced noise, but that is a completely wild ass guess).

This stuff is insidious. It also varies from setup to setup, so the best thing is to give them a try and see if they make a difference for you (If you do, awesome, and if you don’t, even more awesome!)

 

[Gibson59]

The good thing is that, according to digital sampling theory, it is possible to rebuild the exact initial analog data from the sampled data by using an ideal (sinc) interpolation function. This means that you could indeed "extract more data" from a digital file (redbook or higher resolution) by filling the gaps between samples with zero error compared to the original analog signal which was recorded.

The bad thing is that using an ideal interpolation function would take an infinite computational processing capability, so we need to live with a certain error, and all DAC designers have their take and preferred approach to how to reduce this error as much as possible.
​

The ultimate achievement of the MS (+dual BNC DAC) is to rebuild the original analog signal with an accuracy better than 16bit (16.6bit actually).
This in turn produces a timing error which is close to - or better than - the threshold of human brain (a few microseconds according to literature) of discerning transients.
In order to do this, the approach selected by Rob is to use a very long (1Mtaps) sinc function interpolation filter, and to upscale the sampling rate to 705.6/768kHz.

The true "magic" of M Scaler resides on using its full 1Mtaps filter length / WTA combination algorithm, because only by doing so you would bring the timing accuracy to a level where the brain should not be able to discern transient errors. RW insists that transient errors are at the root of all evils for audiophiles because they degrade also timbre and bass pitch perception.

Therefore, to answer your questions based on my (admittedly limited) grasp of all this matter:

a- yes, in a way (see above) the MS 'extracts more data' and 'lessens the data that's lost' from files with resolution lower than 705/768kHz;
b- if you had a native file of a certain resolution (say 24/192), and use the MS by starting from a lower res one (say 16/44) to upscale "only" to the native level (i.e. to 24/192 in this example), the MS would not produce any advantage, as I understand it
c- in today's world, native files at 705.6/768kHz are not easily available to us consumers, so by using the MS basically all practically available (PCM) digital music can be improved to some extent

As an M Scaler owner, I can testify that I can hear improvements from the HMS even when playing high res (e.g. 24/192) files, and that there is a marked difference from using the HMS at full throttle (1Mtaps, hence everything upscaled to 705.6/768kHz) compared to using its upscaling capabilities only partially (which you can do by adjusting the settings of the HMS OP SR button to passthrough - low - medium - high).

In addition to the videos already linked by @Triode User, here you find some other interesting material:

https://www.head-fi.org/threads/watts-up.800264/page-12#post-13150760
https://chordelectronics.co.uk/wp-content/uploads/2018/07/The-theory-behind-M-Scaler-technology.pdf

Hope I have not posted anything misleading, if so other more knowledgeable people and of course Rob Watts can straighten this out.

Now THIS is the explanation I was hoping for! Thank you for taking the time to write this up. Very interesting that science has allowed us to manipulate digital data in conjunction with such a deep understanding of how we perceive sound.

The explanation kind of makes me feel better about my overly simplistic way of describing what using the Mscaler sounds like to me... that it just sounded natural (which I feel like is the very best compliment I could give), and the timing and rhythm were just so perfect, it almost sounded slow at first but I think that’s just because it was perfectly synced with my natural expectation of what instruments should sound like. I think normally when I listen to music I have a preconceived sense of how it will be reproduced with regards to speed, timber, volume etc and in a way that’s almost subconsciously startling... whereas with the Mscaler in the chain (using Qutest and HEKse) I wasn’t startled or artificially aroused (place immature joke here lol) at all, it felt absolutely pure and organic and that it just fell seamless in place with my natural rhythm. I don’t mean to sound like I’m on an acid trip here, but that’s the only way I can think to explain what Chord has achieved here in my elementary terms.

 

[GreenBow]

HiFi+ has reviewed the Chord Hugo TT2 and M-Scaler, in the current issue. May 2019 issue 171.

More of the review is based on what the reviewer thought of the TT2 with the M-Scaler. It still clear however that Hi-fi+ love the TT2. (I don't want to risk a copyright by describing what they said.)

 

[Uncle Monty]

Hi Ray,

I'm now also using audiolinux and a NUC but with optical connection to my HMS.

I already have an LPS 1.2 but would you recommend USB via ISO regen over optical?

Ade

Rob recommended to me using optical over USB with Mscaler

I also got improvement over standard BNC cables by using WAVE Storms - expensive but then so is HMS

 

[musickid]

Something funny. Imagine someone new to the hobby with more dosh to spend on the hobby slowly becoming available with time.

Imagine the same headfier having only limited experience in listening to non chord gear.. in fact he loved the mojo sound at richer sound and hugo 1 sound more. Both were listened to for around 10 minutes but that was enough.

The same headfier i.e. me managed to progress to a hms/h2 and the music sounds so beautiful that he spent nearly 2.5 years studying the science behind it to feel better about the whole thing and a desire to understand how and why.

The really funny thing is that due to the limited listening experience, i have nothing to really compare the sound of my hms/h2 with apart from to say that it could be easily taken for granted when reading about dave/tt2 and the constant chase for those blissful sparks of music, the pure bliss you can get now drives him to reach for more. I did audition dave a long time back but it's still a leap of faith as it always has been with chord and even though we all interact here it still seems to be a solitary and lonely leap forward in the hope that more resolving gear will bring you closer to the music you hold so close to heart..

 

[PJDubyaM]

I ordered a bluesound node 2i and thinking of connecting it to HMS using optic cable. Just wondering if I can use RCA output of the node2i to feed the BNC input?

Can't help directly with your question, but I feed my M Scaler with a Node 2, and I do it via optical. From having read this thread (all of it!), optical's pretty much the way to go.

You can set the Node to have fixed volume output as well, which I think shortens the path out of it.

 

[514077]

Something funny. Imagine someone new to the hobby with more dosh to spend on the hobby slowly becoming available with time.

Imagine the same headfier having only limited experience in listening to non chord gear.. in fact he loved the mojo sound at richer sound and hugo 1 sound more. Both were listened to for around 10 minutes but that was enough.

The same headfier i.e. me managed to progress to a hms/h2 and the music sounds so beautiful that he spent nearly 2.5 years studying the science behind it to feel better about the whole thing and a desire to understand how and why.

The really funny thing is that due to the limited listening experience, i have nothing to really compare the sound of my hms/h2 with apart from to say that it could be easily taken for granted when reading about dave/tt2 and the constant chase for those blissful sparks of music, the pure bliss you can get now drives him to reach for more. I did audition dave a long time back but it's still a leap of faith as it always has been with chord and even though we all interact here it still seems to be a solitary and lonely leap forward in the hope that more resolving gear will bring you closer to the music you hold so close to heart..

Sounds like me as well, MK. I started with the Hugo1, not having been exposed to other products from competetors. What's the saying: When you go Chord, you never go back.

 

[TheAttorney]

I have been hearing similar things with my end point experiments: NUC running real time OS (Audio Linux) in memory with Squeezelite with extremely large memory buffers on the network side and the ALSA driver side. The SQ lift from these experiments (significant) is very similar to what I hear with the ISORegen/LPS 1.2 combination. In fact, by chance I had 2 ISO Regens in house, and when I put them in series, it is the closest I've come to not hearing a difference with the end point optimizations I've been doing

Just to be clear, are you saying that, with 2 IRs in series, the quality of the source no longer mattered much? I.e. any old PC source would sound almost as good as your fully optimised NUC?
Or is it more the case that you could no longer hear the effects of incremetal tweaks to your already tweaked NUC?

My single IR (with microRendu) definitely improved the USB input to DAVE, but I haven't tried removing it since gettiing my HMS. I'm aiing to replace my laptop source with a ram-booted NUC one day, but still waiting for the dust to settle on all those experiments over at AudiophileStyle.

 

[STR-1]

If others try ISORegen and LPS 1.2 with their mScaler setups, please share your experiences. I don't know how much of what I'm hearing is unique to my setup and house, and what may be tickling at a more general noise vector. I look forward to hearing what other people are hearing.

I've not had chance to hear with an LPS-1.2, but I heard similar things powering my ISO Regen with an LPS-1 when used with my mRendu.

I did try the IR with my Zenith SE when I first got that but felt that the SE's warmth and confident bass didn't need similar input from the IR. I found the tX-USBultra (more detailed but a little leaner than the IR) powered by an SR4 (reported by some as being slightly warmer/fuller than the LPS-2.1) to be both superior technically and to better synergise tonally with the SE. I also use Lush^2, USPCB and Sablon Reserva Elite USB options depending on mood.

Conversely, it am not hearing significant effects on treble smoothness / digital glare with my current setup.

Try one of the WAVE High Fidelity BNC options to reduce that glare. I have tried both Storm and Stream. They both work but I decided on the cheaper Stream as a better tonal fit for my system.