Chords New M -Scaler

[mansr]

2 minutes ago, Shadorne said:

Wow. A bit condescending. First you say oversampling to 250 GHz is impossible and now it is nothing special or extraordinary.

He's right. Conversion from one rate to another can be done by first upsampling to the lowest common multiple, then down to the target rate. However, and as stated in the quote from Benchmark, almost all of the calculations when doing this would be multiplications by zero. Eliminating these leaves you with something commonly called a polyphase filter. Calling it by any other name can only be intended to confuse.

[Miska]

2 hours ago, Shadorne said:

Wow. A bit condescending. First you say oversampling to 250 GHz is impossible and now it is nothing special or extraordinary. So which is it? My BS detector has gone off - you talk like a leading authority on D to A conversion  (bored by ESS chips) but I think I can see behind the mask someone is struggling. I notice that almost all of your posts tend to have a marketing angle intended to promote what you hawk. Perhaps Benchmark are simply guilty of the same - hardly surprising. Kettle calling the pot black me thinks?

 

This would have been much less eyebrow rising if you would have immediately quoted the original text... So it is virtual intermediate like I was speculating before. So there are no samples produced at 250 GHz rate ever. You said first that it upsamples to 250 GHz. But from what you quoted it specifically says it doesn't do that. What you quoted describes a polyphase filter.  The output rate is just 211 kHz which is the rate it really operates at. HQPlayer has quite a bunch of such, but operates and outputs for example at 24.576 MHz output rate. But it never crossed my mind to brag about the "virtual intermediate" rate (which can be easily in GHz range) or claim that such way to process a filter would be special. For me, more interesting is the filter design itself, instead of how it is applied.

 

Even Windows and macOS have such resampling filters built-in, just for small change ratios and poor filter designs to keep computational cost as low as possible.

 

[Shadorne]

On 10/10/2018 at 7:50 AM, Miska said:

 

This would have been much less eyebrow rising if you would have immediately quoted the original text... So it is virtual intermediate like I was speculating before. So there are no samples produced at 250 GHz rate ever. You said first that it upsamples to 250 GHz. But from what you quoted it specifically says it doesn't do that. What you quoted describes a polyphase filter.  The output rate is just 211 kHz which is the rate it really operates at. HQPlayer has quite a bunch of such, but operates and outputs for example at 24.576 MHz output rate. But it never crossed my mind to brag about the "virtual intermediate" rate (which can be easily in GHz range) or claim that such way to process a filter would be special. For me, more interesting is the filter design itself, instead of how it is applied.

 

Even Windows and macOS have such resampling filters built-in, just for small change ratios and poor filter designs to keep computational cost as low as possible.

 

 

I don’t think it was bragging. It was an application note on their website. John Siau writes many of these notes to explain about the hardware they design and the choices made. The 250 GHz sample rate is used to make 4 picosec timing adjustments so it is integral to the way the Benchmark DAC 3 stays phase locked while using the asynchronous clock in the DAC. For sure there is a marketing spin to everything they write - John Siau very likely  believes he is making wise and rational choices to provide optimal performance at a price point - this is no different from any technical company trying to win customers. These technical notes make it quite clear that Benchmark are not simply using the ESS 9028 chip straight out of the box - they are customizing and I guess that is a marketing differentiating point vs competitors that they are trying to make.

 

Anyway thanks for your effort and time to explain things. I take it that HQPlayer can’t really do anything for any of the Benchmark products because they are highly customized and “hardwired” unlike most DACs which just connect a power supply and inputs and outputs to the DAC chip and use the chip as it is.

[jabbr]

1 hour ago, Shadorne said:

 

I don’t think it was bragging.

 

 

My new USB cable does 1 Thz virtual upsampling (and then downsampling). Don’t mean to brag but it’s amazing ... and very carbon neutral  

[Miska]

3 hours ago, Shadorne said:

Anyway thanks for your effort and time to explain things. I take it that HQPlayer can’t really do anything for any of the Benchmark products because they are highly customized and “hardwired” unlike most DACs which just connect a power supply and inputs and outputs to the DAC chip and use the chip as it is.

 

I wouldn't say it is "just connects", there's quite a bit more involved. However I'm most interested on DACs that don't use any DAC chips, but instead have more original, discrete, design. Like Chord which this thread is about. Or Holo Audio, Denafrips, dCS, Playback Designs, etc.

 

Based on my measurements, ESS chips perform the best when running at DSD512. But Benchmark DACs don't support that. Or even PCM inputs higher than 192 kHz. So it is not really interesting to me. There are other ESS based products that offer more.

 

I also like more DACs that offer capabilities of the AK49xx chips.

 

[mrkoven]

Any other Mscaler owners care to share their impression? Thinking about adding this to DAVE but a bit hesitant on the benefits..

[ecwl]

47 minutes ago, mrkoven said:

Any other Mscaler owners care to share their impression? Thinking about adding this to DAVE but a bit hesitant on the benefits..

Haha... I just flipped through this thread and realized despite all the technical discussions, only @romaz and myself actually own M-scalers in the form of Blu2 and DAVE (and we have both tried HQPlayer) and we barely talked about our subjective sound quality impressions of M-Scaler. And @romaz and I basically owned both Chord products around the time when they came out.

The best way to describe M-Scaler over DAVE is to describe DAVE over other DACs you might have had before. I don't know if you've noticed DAVE, with some instruments or vocals, the timbre of the instruments and the voice just sound more realistic than what you're used to with other DACs. Or when you're listening to percussion instruments, going from guitar plucks to finger snapping or hand clapping all the way to strikes of drums, DAVE sounds more realistic and dynamic compared to most DACs you know. Well, whatever DAVE to the other DACs is M-Scaler to DAVE.

I almost never listen to DAVE alone but ironically in the past two days I did. My dealer had a new setup so I dropped by to listen to some unfamiliar pieces through his DAVE (direct to PS Audio BHK 300 mono amps) and things sound great except the instruments just didn't sound as realistic as I'm used to so when I got home, I listened to the same songs through Blu2 and the transients and timbre of instruments were at another level that I really enjoyed.

Because I disconnected my video system from my Blu2, I decided to watch TV using DAVE alone (instead of the 0.6M taps of M-Scaler) and within 2 minutes, I just didn't appreciate how the sounds of explosions or people fighting didn't seem quite right for me, in addition to people's voices not sounding as realistic as I'm used to. Plugging Blu2 back into the video system had a nice improvement.

I would say that if you own DAVE and can afford it, M-Scaler is a no-brainer. The only caveat is that there is a possibility that your downstream devices after the DAVE, e.g. preamplifier (if you have one) or amplifier might be a limiting factor in hearing all the additional improvements. I think that's why the Head-Fi forums tend to rave about Chord products more because there are fewer intermediate factors that can reduce the performance of the DACs or cause synergy issues.

Anyway, that's my personal subjective (and obviously biased) opinion. Hope it's helpful to you.

[mrkoven]

46 minutes ago, ecwl said:

Haha... I just flipped through this thread and realized despite all the technical discussions, only @romaz and myself actually own M-scalers in the form of Blu2 and DAVE (and we have both tried HQPlayer) and we barely talked about our subjective sound quality impressions of M-Scaler. And @romaz and I basically owned both Chord products around the time when they came out.

The best way to describe M-Scaler over DAVE is to describe DAVE over other DACs you might have had before. I don't know if you've noticed DAVE, with some instruments or vocals, the timbre of the instruments and the voice just sound more realistic than what you're used to with other DACs. Or when you're listening to percussion instruments, going from guitar plucks to finger snapping or hand clapping all the way to strikes of drums, DAVE sounds more realistic and dynamic compared to most DACs you know. Well, whatever DAVE to the other DACs is M-Scaler to DAVE.

I almost never listen to DAVE alone but ironically in the past two days I did. My dealer had a new setup so I dropped by to listen to some unfamiliar pieces through his DAVE (direct to PS Audio BHK 300 mono amps) and things sound great except the instruments just didn't sound as realistic as I'm used to so when I got home, I listened to the same songs through Blu2 and the transients and timbre of instruments were at another level that I really enjoyed.

Because I disconnected my video system from my Blu2, I decided to watch TV using DAVE alone (instead of the 0.6M taps of M-Scaler) and within 2 minutes, I just didn't appreciate how the sounds of explosions or people fighting didn't seem quite right for me, in addition to people's voices not sounding as realistic as I'm used to. Plugging Blu2 back into the video system had a nice improvement.

I would say that if you own DAVE and can afford it, M-Scaler is a no-brainer. The only caveat is that there is a possibility that your downstream devices after the DAVE, e.g. preamplifier (if you have one) or amplifier might be a limiting factor in hearing all the additional improvements. I think that's why the Head-Fi forums tend to rave about Chord products more because there are fewer intermediate factors that can reduce the performance of the DACs or cause synergy issues.

Anyway, that's my personal subjective (and obviously biased) opinion. Hope it's helpful to you.

 

That is helpful thanks for the response. I split my DAVE use between headphones and active speakers so not much downstream implications. The sound is so pure on it's own, I think I want to keep it that way. I will probably go for the Mscaler, my curiosity is killin' me! 

 

[ray-dude]

6 hours ago, mrkoven said:

Any other Mscaler owners care to share their impression? Thinking about adding this to DAVE but a bit hesitant on the benefits..

 

About a year ago I (elsewhere) posted my way too long review of the Blu2:

 

https://www.head-fi.org/showcase/chord-blu-mk-ii-digital-cd-transport.22848/reviews

 

Having listened to a buddies HMS with Dave, except for all the ferrite crazy, same impressions stand.  

 

As amazing the experience when you first hear mScaler, it really does feel like you’ve lost something integral and real when it goes away.  Over the past year, it revealed so much to me that I’ve completely revamped my system end to end (including parting with my beloved B&W 802d3’s, after having spent multiple decades dreaming of getting 800 series speakers).

 

Don’t give it a listen unless you’re willing to go all in. It really is a red pill moment. 

 

 

[EnjoyTheMusicNow]

On 8/2/2018 at 11:05 AM, BigAlMc said:

Just read all ten pages of this thread and not one of you bozo's answered the all important question! Will this thing improve @BigAlMcs system! ?

 

Short answer I suspect is not without significant changes, as I'm not even close to being willing to consider parting with my Directstream DAC and because I have a heavily optimized USB flow.

 

But would appreciate some more expert input on the following. The M-Scaler sounds very impressive but it has USB input but no USB output. Therefore I could potentially put it after my TX-USBultra but not before.

 

So if I have a TX-USBultra (clocked by an SoTM OCX-10 reference clock) providing a very, very nice USB signal to my DAC. In theory would feeding that USB signal into an M-Scaler have potential benefits or be a dumbass idea?

 

Would the M-Scaler take that very nice USB signal and scale it to 386Khz or whatever making it even nicer? Or would the M-Scaler completely reclock/rejig the signal to the extent that the efforts (money spent!) on the TX-USB-Ultra/OCX-10 were rendered pointless or lost?

 

Cheers,

Alan

 

Did you ever get a response to your question? I have the same question. I have the SOtM master clock and SMS200 ultra neo and an M scaler on order. Trying to figure out if anyone has used and M scaler with the SOtM USBultra and master clock.

[BigAlMc]

39 minutes ago, EnjoyTheMusicNow said:

 

Did you ever get a response to your question? I have the same question. I have the SOtM master clock and SMS200 ultra neo and an M scaler on order. Trying to figure out if anyone has used and M scaler with the SOtM USBultra and master clock.

 

Not really. There was a suggestion that the M scaler makes the most sense with a Chord dac. But the combo of super clocking and super scaling wasnt really opined on.

 

Be interesting to hear how you get on.

 

Cheers,

Alan

[asdf1000]

On 10/12/2018 at 5:19 PM, Miska said:

Based on my measurements, ESS chips perform the best when running at DSD512.

 

Do you think the ESS chips noise shaping operate at 256fs or 512fs? In this old 2013 article they said the ES9018 could be operating at 256fs but also possible at 512fs (guessing).

 

From your recent measurements and looking at ultrasonic noise patterns does it look more like newer ESS chips (like in the Pro-Ject S2 DAC for example) are running noise shaping at 512fs? Or likely 256fs from your measurements/guess?

 

Or still too hard to guesstimate?

 

"The requirement for feedback-based noise shaping is still there, though: the ESS Sabre 9018, which is probably the most advanced delta-sigma converter in current production, most likely operates at 11.2896MHz, or 256fs. (Published data on the internals of the 9018 is not readily available; the 256fs speed is a best-guess. It might go all the way up to 45.1584MHz.) "

 

https://positive-feedback.com/Issue65/dac.htm

 

This ES9023 spec sheet says:

 

"For best performance. 256fs or greater is recommended for 32kHz to 96kHz sampling."

 

But I don't know if this has anything to do with the noise shaping sample rate?

 

 

https://www.computeraudiophile.com/applications/core/interface/file/attachment.php?id=6757

 

 

[ecwl]

7 hours ago, Em2016 said:

 

Do you think the ESS chips noise shaping operate at 256fs or 512fs? In this old 2013 article they said the ES9018 could be operating at 256fs but also possible at 512fs (guessing).

 

From your recent measurements and looking at ultrasonic noise patterns does it look more like newer ESS chips (like in the Pro-Ject S2 DAC for example) are running noise shaping at 512fs? Or likely 256fs from your measurements/guess?

I suspect ESS chips when fed DSD512 just plays it like a DSD DAC as if it's a 64-element shift register DSD design. So I don't think there's any additional noise shaping involved.

 

But when you're playing a PCM file on an ESS chip, I'm not sure if we can easily know whether it upsamples and noise shapes to 5-6 bits at 64fs, 128fs or even at higher fs. We also don't know whether it upsamples to say 16fs 24-bit first and then to say 128fs 6 bits or whether it'll be a one-step process.

 

But I think whatever the process is, it's almost a little irrelevant. First of all, I am quite certain ESS chips do not upsample from PCM and noise shape directly to DSD512. Moreover, the computational power involved in the ESS chips would be dwarfed by what HQPlayer can do when it upsamples and noise shapes to DSD512.

 

Similarly, if ESS chips upsamples to 16fs first like M-Scaler, the computational power for the upsampling filter with ESS would be dwarfed by what the M-Scaler can do. And the subsequent upsampling or noise shaping to 5-6 bits at say 128fs to be output to the 64 elements SDM would be computationally significantly less intensive than say the Chord Qutest upsampling from 16fs to 256fs and then to 104MHz then noise shaping for playback on the 10 discrete elements of the pulse array DAC.

 

I think the bottomline, whether you're going with a final DAC output of DSD512/DSD1024 or a multi-element discrete SDM design is that the more computational power (and superior algorithm) you can throw at it for upsampling and noise shaping, the better the sonic result. (Unless you believe that all DACs measure the same and all noise/distortions measured are already beyond the threshold of hearing in which case you should stick with your headphone amp/DAC/jack that comes with your cellphone...)

[Miska]

8 hours ago, Em2016 said:

But I don't know if this has anything to do with the noise shaping sample rate?

 

 

Not much, these are just clocking requirements. The chip needs at least one clock cycle per output sample (at it's native SDM rate), but it is not unusual to need more than one clock cycle per output sample. Higher grade ESS chips can use up to 100 MHz clock. Sometimes chips can adapt and also accept lower than optimal clock rate, but then they make a shortcut at DSP side for example and quality degrades to some extent.

 

In these "traditional" implementations everything runs out of single clock (unlike in separate upsampler + DAC case). Which has challenge that when you increase clock frequency your phase noise figures get worse. And if you lower clock frequency you limit the DSP capabilities. If you run the DSP and DAC asynchronously from completely independent clocks, you can do whatever you like in DSP and you can choose optimal clock frequency for the DAC purely from the conversion stage perspective. But so far, all DAC chips run out of single MCLK.

 

For comparison, you can run the DSP stuff on CPU running at 4 GHz and DAC having oscillator of it's own running at 22.5792 MHz for DSD512 (44.1 x 512). DAC only converts samples to analog as-is. And computer only does DSP. Each optimized for their own domain of work.

 

[barrows]

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.

[Miska]

23 minutes ago, barrows said:

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.

 

AKM also goes a bit further and has always been recommending a different (reference) analog filter design for DSD usage than for PCM.

 

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.

 

For example T+A DAC8 DSD is quite unique in a way that it really has two different analog filter settings you can select.

 

23 minutes ago, barrows said:

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).

 

It has two cascaded programmable FIR filters. First stage is 64 taps and second stage is 16 taps. This is also what is used for "MQA rendering" with MQA's own filters.

 

[barrows]

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?

[mansr]

4 minutes ago, Miska said:

It has two cascaded programmable FIR filters. First stage is 64 taps and second stage is 16 taps. This is also what is used for "MQA rendering" with MQA's own filters.

The second stage is symmetrical, so there are twice as many taps as programmable values. The 9018 datasheet gives this number as 16 in one place and 14 in another. Datasheets for more recent chips say 16 values can be written but only 14 are used.

[Miska]

1 hour ago, mansr said:

The second stage is symmetrical, so there are twice as many taps as programmable values. The 9018 datasheet gives this number as 16 in one place and 14 in another. Datasheets for more recent chips say 16 values can be written but only 14 are used.

 

IOW, the second stage must be a linear phase filter... Which is a sensible choice.

[Miska]

1 hour ago, barrows said:

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?

 

Yes, if you look at AKM data sheets for chips that support DSD (at least most of those) have two reference filter designs. One for PCM use and another for DSD use.

 

For DACs that are not very cost-limited it would make sense to have two relay-switched filters, but I rarely see such. Apart from things like DAC8 DSD.

[asdf1000]

3 hours ago, barrows said:

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).

 

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

 

Do you have an idea/guess with your 9038 Pro? If so, how did you come to this number?

 

[mansr]

20 minutes ago, Em2016 said:

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

Most likely some fixed fraction of the master clock rate, which due to the ASRC doesn't need to be related to the audio sample rate.

[barrows]

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).

[barrows]

16 minutes ago, mansr said:

Most likely some fixed fraction of the master clock rate, which due to the ASRC doesn't need to be related to the audio sample rate.

Yes, I think this is correct.  But, the ESS chips, IMO, do sound better when run synchronously (as long as the source/masterclock is low jitter) where the masterclcok and bit clock are derived form the same clock.  For example, I run the ESS 9038 with a 45.1584 masterclock, and this same clock is the clock input to the Amanero interface, and re-clocks the data lines just before the DAC.  Synchronous operation like this drops out (or one can turn it off) the DPLL as it has no need to make changes as the masterclock and bit clock are already synchronous.

[Always.Learning]

It is ironic that, in this thread devoted to Chord's M-Scaler, there are only a handful of reviews/impressions from actual users. Here is a review I just posted on Head-Fi:

 

I have now had the M-Scaler in my system for about ten days. Here are some observations.

 

Overall Gestalt

 

The single biggest takeaway for me is that music is more composed and calm. Music does not feel as hyped as before; there are not as many moments that show some hint of harshness; there are fewer sharp edges. In a word, music is more natural. Some would say this is a darker sound and I wouldn’t disagree. But it is not darker at the expense of detail or resolution. Frankly, this greater sense of composure ran counter to my expectations, which centered on things like clarity, detail, and soundstage. I didn't know quite what to make of the M-Scaler during my first hour or two of listening because in some ways it seemed like some drama was missing. The more I listened, the more I realized that I was more easily drawn into the music and a greater emotional connection was formed. There was a subjective sense that music had slowed down. 

 

This is pure speculation on my part, but my guess is that HMS is easing the difficulty my ear/brain may have processing music; listening is both more relaxing and more involving.

 

It is important to put this in context:  my current system (see below) is the least forgiving and most transparent that I have had over the years (though I have some friends with more resolving and less forgiving systems). The speakers are not laid back; rather, they are dynamic, nimble, and fully capable of revealing harshness, glare, and poor recording quality. So this newfound sense of composure and calm is particularly beneficial in the context of my system. 

 

Separation of Musical Lines

 

This is probably the second biggest revelation for me:  the ability of the M-Scaler to unravel dense musical passages and allow one to follow disparate musical lines. There is a very evenhanded treatment of different strands in the music that allows one to hear and focus on multiple strands of music simultaneously rather than forcing the focus on a particular instrument or voice. This enhanced ability to follow stacked elements of the music or criss-crossing musical lines is a chief virtue of the HMS. 

 

In his excellent review of the HMS on audiophilelifestyle.com, @austinpop captured this well when he said:

 

"The best way I can describe it is temporal coherence. There is just something “right” about the sound in terms of its timing. Take dense orchestral music like the Mahler. Normally, with music like that when different instruments come to the foreground in terms of volume, one’s ear is drawn to them, and the others recede from attention. It’s akin to surfing - riding on the crest of each wave. Forgive the mangled metaphor - I don’t surf! What I found with the HMS is that as an instrument receded from its crest, it remained perfectly easy to follow. As a result, I found myself able to keep my attention on what instruments were doing off their peaks. On dense music, this is a revelation! It draws you so much deeper into the piece, both intellectually and emotionally."

 

See https://audiophilestyle.com/ca/reviews/my-quest-for-a-new-dac-part-5-chord-electronics-hugo-m-scaler-hugo-tt-2-r781/

 

This ability to follow musical lines not only draws you into the music, it allows you to appreciate and understand how it is actually put together.  

 

High and Low Volumes

 

On most tunes I am now able to turn up the volume by several decibels with no hint of strain, compression, or harshness. Conversely, the system sounds very good at low volumes, allowing one to hear a coherent whole without missing pieces. Since my listening room is our living room, this flexibility is important to me. 

 

Human Voice

 

Wow. Very dimensional, very natural. Little inflections are heard and understood. 

 

Bass

 

The M-Scaler does not produce more (or less) bass than I was hearing before. Rather, it is cleaner, more intelligible, tuneful, and easy-to-follow bass. Bass also seems more locked in with the rest of the music. If bass is important to a given tune and recorded well, you are in for a treat.

 

Soundstage/Depth

 

One of the major strengths of my current system is a large soundstage with excellent imaging. I wouldn't necessarily say that the HMS has produced a bigger soundstage. I do hear better layering front to back. I also "see" images that are more defined and that seem to have greater cushions of air around them, without fuzz on the edges. For example, on a recording of two pianos, it is easier to hear the two pianos as distinct instruments played by distinct musicians, each occupying their own space. On a musical level, this allows one to appreciate the interplay between the musicians and grasp more of their intent. Related to the issues of soundstage and depth, there is significantly greater recovery of ambience and air in recordings that have it.

 

Impact on Redbook Files

 

I've never been a big believer in the importance of format; it’s music first, then recording quality; with format last. I am now even less of an advocate for the importance of (or differences among) hi-rez formats. HMS tends to level the field, unlocking hidden beauty in your Redbook files. Really good recordings become stupendously good. As others have noted, however, it is not a miracle worker. The impact of HMS on Redbook varies widely depending on recording quality. 

 

Setup Issues

 

Chord should really insert some additional pages in the HMS owner’s manual that spell out exactly how one should connect the M-Scaler to DAVE, Hugo2, and Qutest. Currently, the manual only deals with Hugo TT2. I was up and running quickly, thanks to some helpful comments by folks in this thread.

 

An Observation Regarding Power and the HMS

 

I am unlikely to try different power supplies on the HMS, as I have read enough comments, from enough experienced people, to conclude that it is difficult to better the supplied power supply without going to a great deal of expense. But I did try one little experiment.

 

[Side note:  I replaced the stock power cord that comes with the HMS with a Shunyata Venom power cord. Pretty cheap upgrade at $95. I have not compared the stock power cord with the Shunyata.] 

 

Initially, I plugged the Shunyata power cord into a fairly cheap Wireworld power strip. This is where I have plugged in my other non-analog power supplies that power various components in my digital chain (Uptone Audio LPS 1s). This power strip is plugged into a different circuit than my dedicated audio circuits. This practice follows the advice of Shunyata, which is to separate purely digital components (like the HMS) from anything that contains an analog circuit, both physically (to the extent possible) and in terms of power circuits. 

Sometimes the received wisdom needs to be tested, however, and so I plugged the Shunyata cord into an available outlet on my Audience aR6 TSSOX power conditioner. This particular outlet was designed for digital components (like a DAC), although I prefer to plug DAVE into one of the outlets normally reserved for amps (and I have my amp plugged into the wall). 

There was a clear and unambiguous change for the better using the power conditioner to get power to the HMS. What stood out was that bass was more intelligible and propulsive, and backgrounds seemed blacker. There was another step forward in reducing edginess and a step up in musical engagement. The Audience conditioner, combined with its expensive Audience power cord to the wall, and my dedicated 20 amp circuits using 10 gauge wire all combined to produce this result.

 

BNC Cables

 

Note that my experience so far has been with the stock cables. Eventually, I plan to audition two or three brands of BNC cables and see how much of a difference they make. 

 

Comparisons with Vinyl

 

I have a good vinyl setup and it's been an interesting exercise to compare good vinyl records with the same tunes played through HMS. Not surprisingly, the results vary widely. For those who believe HMS has made vinyl obsolete, I'm sorry to report that is not the case. Vinyl is still king with select recordings. Example:  any tune on Joni Mitchell's Blue album (the reissue on Rhino). This record has that "breath of life" that is immediately recognizable within 10 seconds. It's mostly about tone and texture. The dulcimer on "California" and "A Case of You" has much more texture and woody resonance than with any digital version (that includes a Tidal 24/192 MQA version (unfolded by Roon to 24/96); a Qobuz 24/96 version; and my own rip of a Redbook CD). By comparison, the dulcimer through my digital chain sounds a little plasticky, lifeless, and grey. Joni's voice is simply more human, more tangible, and more real on vinyl, and my analog system throws a considerably bigger image. The comparison is not close at all. 

 

On the other hand, yesterday a friend brought over Radiohead's "In Rainbows" album and the Reference Recordings album "Exotic Dances from the Opera." Neither vinyl album could hold a candle to digital playback of the same tunes through HMS. 

 

I will say that the variation in the quality of vinyl playback is much greater -- from really bad to really great -- than what I typically hear through digital with HMS, which tends to vary from OK to really great. 

 

My System

 

To help put my observations in context, my system consists of:

 

Devore Gibbon X loudspeakers

HiFiMan HE1000 headphones

 

Chord DAVE

Chord Hugo M-Scaler

Roon/Tidal/Qobuz running off SD card

Mac Mini modified by Uptone Audio in bridged mode serving as Roon Core server, powered by Uptone Audio JS-2 power supply

Paul Pang ethernet switch with TXCO clock powered by Jameco LPS

SOtM dCBL-Cat7 Ethernet cables

Blue Jeans Cable 6a ethernet

SOtM sMS-200 Ultra powered by Uptone Audio LPS-1 power supply with custom Zenwave UPOCC copper cable 

Uptone Audio Iso Regen powered by Uptone Audio LPS-1 power supply and Ghent Audio cable 

SOtM txUSB Ultra powered by Uptone Audio LPS-1 power supply with custom Zenwave UPOCC copper cable 

 

TW Acustic Raven turntable and integral phono cable

TW Acustic Raven 10.5 tonearm

Miyajima Shilabe cart

Zanden/TW Acustic record mat

 

Crayon CFA 1.2 integrated amp with Synergistic Research Blue Fuses

Crayon CPH-P phono stage with dedicated Paul Hynes SR5 DR48  power supply with Synergistic Research Blue Fuse

 

Daedalus/WyWires speaker cable

High Fidelity CT-1 Enhanced interconnect

WyWires Gold interconnect

Phasure Lush USB cable

Phasure Lush^2 USB cable

Audience AU24 SE USB cable

 

 

Audience aR6 TSSOX power conditioner

Shunyata Alpha HC power cord powering amplifier direct-to-wall

Shunyata Alpha HC power cord powering Chord DAVE

selection of Audience AU24 SE and Audience PowerChord power cords 

Two 20 amp dedicated circuits with 10-gauge wire and FIM 888 AC outlets

 

Stillpoints Ultra SS and Ultra Mini under most components including speakers

Finite Elemente rack with Cerabase footers

Equipment Vibration Protectors from A/V Roomservice under other components

Finite Elemente Ceraballs under other components

Custom wall-mounted TT rack

Acoustic Revive Schumann resonator powered by Jameco LPS

Stillpoints Aperture acoustic room treatment (2)

A/V Roomservice acoustic room treatment (4)

Acustica Applicata Diffusion Absorption Audio Device

ZSleeves

DIY ultrasonic record cleaner/VPI 16.5 record cleaner

 

High Fidelity Cables MC 0.5 Wave Guide

High Fidelity Cables Magnetic Adapters