[onlychild]

Not sure if this has been posted yet, but a BNC Coax optical bridge was just announced that specifically states it supports Chord's DX! This could be next level with regard to taming RF interference between the M Scaler / Blu MkII and Chord DAC!

"Audiophiles have sought to minimize noise transmission using cleaner sources, ferrite laden coax cables or myriad other tweaks. These solutions are partial or, at best, only attenuate the problem to subjectively match the noise floor. For those with the most resolving systems and highest appreciation of musical transparency, the optical galvanic isolation of OPTO•DX is objectively superior."

https://audiobacon.net/2019/06/28/audiowise-inc-introduces-opto•dx-optical-isolation-bridge/
https://audiowise-canada.myshopify.com/

Headfi review with M Scaler and Blu: https://www.head-fi.org/showcase/audiowise-opto•dx-optical-isolation-bridge-for-dual-spdif.23757/

I received my OptoDX boxes Saturday morning and had them playing in my system all weekend. These things are the real deal. They did exactly what they claimed to do, clean up noise. I have not heard my system with this low of a noise floor before. The biggest change I noticed is just how sweet and "real" the treble sounds now. I actually look forward to the next song on shuffle, instead of wincing in fear of hearing the hardness it had before in the upper region. Bass resolution and soundstage depth also increased, as well as detail resolution, but oh my the treble!

I am running an Innuos Zen Mini with LPS using optical out to the HMS which is running off a pilot pro 2 battery with High Fidelity Cables CT-2 cables from HMS to Dave. Dave is being powered by a HFC Pro series power cord. Dave is feeding the HFC Trinity Helix Headphone module which is feeding the Empyrean and HE1KSE. What is surprising is that even before the OptoDX boxes, I thought I was running a pretty noise free system, but the boxes are a real eye opener to the high frequency noise the HMS is producing. Maybe from the FPGA processing?

I tried using the BNC patch cables that come with the boxes, but I found my HFC CT-2 to sound better when used from the RX box to Dave.

Both boxes were being powered by separate pilot pro batteries and Ghent Gotham DC cables.

Great job Dan!!

 

[me2621a]

Good questions, and not so easy to answer.

1. Take a regular 48k/24b recording; that has samples every 20.8 uS. We know that the ear/brain is at least as sensitive to 4 uS (my subjective evidence suggests it's actually a great deal more sensitive). Also, we know that if we use a sinc function we will perfectly reconstruct the original bandwidth limited signal; we also know that if we do not use a sinc function the transients will not be reconstructed accurately - they will be a little too early or too late. We also know from psychoacoustics that transients are essential from a perception POV, as they are used by the brain to compute pitch, timbre, rhythm and instrument separation. The M scaler takes the 48k signal and then converts it too 768k with a guaranteed 16 bit accuracy - as the M scaler is the same as an ideal sinc function to better than 16 bits - so it will recover the bandwidth limited analogue signal to better than 16 bits. We now have a 768k recording, with an output of every 1.3 uS now, which of course is much easier for the brain to deal with.

2. But some questions remain. Every time we double the sample rate, the size of the area of the transient error gets 4 times smaller; and with a 16 times increase in sample rate we have a 256 times reduction in the area of the error - but even with a 768 k output we can still get a small transient error. Now this is still audible, and that's where the second WTA filter in my DACs takes over - and this takes the 768k signal and converts it too 12,288 kHz, or a sample every 88 nS, and this filter is still audible, even though we are now dealing with very small timing errors. So yes timing errors above 768k are subjectively important - and it's an aspect I am still researching into. We will eventually hit two barriers - there is no point in getting more accurate transients if the DAC can't accurately resolve it when the DAC converts it too analogue - and there is no point in reconstructing transients to a higher accuracy than the ear/brain can detect. It's these aspects I am looking into.

But there is something special about hitting 768k. The 768k recordings I have sound very different to 384k recordings, and have the qualities that the M scaler gives; this suggests that there is something special about hitting the 1.3us accuracy of 768k. And certainly, the WTA 2 filter currently does not sound like the M scaler - it only seems to affect the ability to perceive the starting and stopping of notes, not pitch, timbre and instrument separation, which happens when you get to 768k.

3. From a processing POV going from 384 to 768 involves exactly the same number of multiplications and processing as 48k to 768k; it's just the filter updates its input samples at a much faster rate, but the M scaler does exactly the same work. Also, the benefit of 16 bit reconstruction when using 16 bit files is obvious - but with 24 bits what do we need? Again, this is another aspect that I am researching into.

I hope that clarifies it for you. In short there are indeed more questions that need to be answered...

Hi Rob,
Thank you for your detailed answer. For the most part this makes sense to me and if I understand your response it boils down to the following:

With the M Scaler we can reconstruct the incoming signal back to the original 16 bit bandwidth limited analog signal and therefore we can resample it at a rate grater then 700khz. By doing this our samples now occur once ever ~1.3uS instead of once ever 20+ uS, and given that the brain can perceive shifts as small as 4uS and subjectively even less this results in a perfect upscaled digital stream from the M Scaler to the DAC with much much smaller transient errors. Then the DAC takes it from there, and because it is operating off of the new upscaled digital stream and there is much more information in that stream (as it has been reconstructed) and the resulting output is "better".

This does bring me to one new question which came up when I read your response about the WTA filter in your dac. If I understand correctly, your DACs (Dave, Hugo TT2, ect...) then oversample the resulting stream to 12,288 kHz, when this occurs is it also approximating a sinc function to a particular bit depth, or is it doing something else entirely?

Thank you again for the detailed response Rob, I really appreciate it.

 

[birdlandbill]

Thanks very interesting indeed to read.
My VERY limited knowledge of digital theory does not allow me to follow and understand everything in absolute detail. But I DO know what can hear via my HMS/Qutest and it has fundamentally changed my take on rbcd 16/44.1.

This morning I have just listened to an early digital BIS recording from Gothenburg with the GSO playing Sibelius's 7th symphony with great pleasure!
The recording was made with only THREE Neuman mics in the good acoustics of the Gothenburg hall with SONY's PCMF1 in 1984!
And what I have just heard sounded almost like hearing a mic-feed from that hall.
Playback via PCM F1 did not sound that good in those days.
Strings could sound a bit steely via that ADC.

I am amazed on a daily basis at how many more recordings I can now really enjoy not only for their musical and performance qualities, but also how very realistic 16/44.1 can sound via HMS.

I mentioned those early digital recordings in some posts here after my return from Asia earlier, and via the stock BNC they did NOT sound as good as I had hoped. But now via upgrade BNC cables they do.
And the difference is not subtle to me.

The coherence and realism,how this orchestra actually sounded playing live in their good hall with these early three mics only 16/44.1 recordings via HMS actually sound more realistic to me than BIS's multimic'd Sibelius symphonies recent remakes from Minnesota.
I was invited to those sessions too, but I would have had to pay for my own flight so I declined the offer.
But I have been to quite a few recordings sessions and many concerts in Gothenburg.
It truly sounds as if I am listening to the analogue signal before digitizing with many of those early cd's now.

But yes really well recorded hi res can sound even more realistic, which brings me to the question of which ADC can record at 768khz?
Didn't you mention 256M taps needed to recover 24 bits to the same level as 1M taps can with 16/44.1 two years ago?

The only PRO ADC's which I am familar with are still limited to 24/358 or DSD 256 which I have heard in use at recording sessions.
Have you already made test recordings with a prototype Davina or from which ADC did you get native 768 recordings?
Cheers CC

Where did you uncover the recording conditions, specifically the Neumann microphones? Sounds like you were actually at the recordings (how wonderful). I've been discovering with HMS/Dave that the sound quality I am getting from classical recordings has everything to do with both the sound engineering, microphone placement, etc, as well as artist performance. This information is rarely provided. Can learn a bit by googling the sound engineer and discovering the methods typically used. Thank you for the information re: GSO Sibelius 7th.

 

[Chester Rockwell]

Any of you M Scaler addicts attending CanJam London this weekend?

 

[andrewd01]

Well, personally I normally withhold my opinions until I have actually auditioned various equipment, dacs ,cables headphones amps and such.

Have you also done that?

I did not comment at all on any benefits or lack of such with these cables until I had actually auditioned them.

And if my memory serves me right, it normally does, even the designer behind DAVE, RW personally recommended trying ferrites on the BNC cables to combat RF problems when people here began hearing such problems with DAVE/BLU2.
I auditioned DAVE/BLU2 in Singapore both with and without ferrites on the BNC cables and the difference was clearly obvious.

Judging from what I can personally now hear with HMS/Qutest and ferrited insulated cabling it may not only be a simple case of Feng Shui as you seem to think?

I can firmly say that to me, the stock cable is NOT as good as either of these two and possibly several other BNC cables mentioned on this thread lately.

Maybe the Opto DX is even better?
But I won't know until I can personally compare.

Cheers CC

I was commenting on the aesthetics of the arrangement.

 

[Christer]

Where did you uncover the recording conditions, specifically the Neumann microphones? Sounds like you were actually at the recordings (how wonderful). I've been discovering with HMS/Dave that the sound quality I am getting from classical recordings has everything to do with both the sound engineering, microphone placement, etc, as well as artist performance. This information is rarely provided. Can learn a bit by googling the sound engineer and discovering the methods typically used. Thank you for the information re: GSO Sibelius 7th.

Hello,
well BIS and Robert von Bahr the CEO of BIS also responsible for many of the early analogue and digital recordings both as producer and recording engineer is quite an exception to the rule. He always used to provide the recording information in great detail and still does so until this day.
And I actually quoted the facts from the booklet this time.
I did not remember that they were Neumann mics..
But I remember that the mid 80s Sibelius recordings made in Gothenburg like many others from there by BIS were engineered by the recording engineer Mikael Bergek and he normally used a few mics for his productions for BIS and only moved to real multimic'ing with the introduction of SACD and multichannel.
I once asked him during some sessions for the BIS Tchaikovsky symphonies SACD series much later why there was suddenly such a forest of mics compared to his normal three to five mics for a large scale symphonic recording earlier.
His answer was that they were needed for balancing reasons in mch mainly and that more postproduction was also involved now.
He said he wished he could still use only a basic setup but it would have been to risky to do nowadays.
And not everybody appreciated his earlier more ambient takes he said.

But in spite of all the mics in the hall they also sound very good in stereo. But even better in mch.

Mikael himself was particularly happy with the SQ of Tchaikovsky's 5th from that series.
Cheers CC

 

[Christer]

I was commenting on the aesthetics of the arrangement.

Ok, I did not quite understand what you meant.

I am single since some years and live in a "man cave" with my HIFI,my books and music taking first place in my living and listening room.
But even while still living with a woman she had to put up with my big speakers and other not so nice looking HIFI related things in our living rooms.
One of my exes used to tell visiting girly friends of her's,"those big black coffins taking up all that space of our living room, are his speakers, and the big ugly blue metal boxes on the floor are also his."
And those speakers didn't even reach anywhere near the ceiling, my current speakers almost do.
HMS is actually quite small.
But I wish it had contained all the things needed for my musical enjoyment in only ONE even smaller box.
Cheers CC

 

[birdlandbill]

Hello,
well BIS and Robert von Bahr the CEO of BIS also responsible for many of the early analogue and digital recordings both as producer and recording engineer is quite an exception to the rule. He always used to provide the recording information in great detail and still does so until this day.
And I actually quoted the facts from the booklet this time.
I did not remember that they were Neumann mics..
But I remember that the mid 80s Sibelius recordings made in Gothenburg like many others from there by BIS were engineered by the recording engineer Mikael Bergek and he normally used a few mics for his productions for BIS and only moved to real multimic'ing with the introduction of SACD and multichannel.
I once asked him during some sessions for the BIS Tchaikovsky symphonies SACD series much later why there was suddenly such a forest of mics compared to his normal three to five mics for a large scale symphonic recording earlier.
His answer was that they were needed for balancing reasons in mch mainly and that more postproduction was also involved now.
He said he wished he could still use only a basic setup but it would have been to risky to do nowadays.
And not everybody appreciated his earlier more ambient takes he said.

But in spite of all the mics in the hall they also sound very good in stereo. But even better in mch.

Mikael himself was particularly happy with the SQ of Tchaikovsky's 5th from that series.
Cheers CC

Thank you for the great information, particularly the Tchaikovsky 5th.

Recording an entire orchestra with 3 microphones doesn't lend itself to do-overs for individual cockups and creates a distant sound for some instruments..., but I like it.

Nothing about Swedish sound engineer Bergek's engineering methods on-line.

Best from Birdland

 

[Thenewguy007]

Single ended components sound better in a domestic installation with short interconnects where noise is not an issue imho. Half the number of components in the signal path. All Chord dacs are single ended. Chord also recommend single ended digital connections. This is why Chord dacs have dual BNC connections instead of dual AES.

I've never heard any sound science on single ended being inherently better than balanced. Unless you use balanced cables on a single ended designed.

 

[Mediahound]

I've never heard any sound science on single ended being inherently better than balanced. Unless you use balanced cables on a single ended designed.

How about:

The primary reason many Dacs have a balanced internal topology is try to overcome switching noise that has been induced into the Dac chips substrate. Balanced circuitry though causes other distortions that should be avoided. Chord Dacs have no substrate switching noise because the switching elements are seperated from the FPGA and more importantly from the analogue circuitry. So because we don't use standard Dac chips that can suffer from these problems. Therefore we do not have no need to used a balanced internal topology so we don't! .

https://www.head-fi.org/threads/cho...pressions-thread.756029/page-31#post_12646676

 

[musickid]

I tried chromecast with roon expecting a step up in SQ due to the small form factor vs optical from my imac but i was wrong. The chromecast sounded very thin, brittle and lacking depth and this was immediately apparent. After more testing nothing changed so i will return it. My imac is my core but i wanted a smaller footprint and i was lucky enough to find a late 2014 mac mini which i will use as a roon bridge optical out to my mscaler. It was unused brand new boxed and i made a deal and got it for less than half the market price. This will be my front end for my system imac core and mac mini optical bridge. The mini is a lovely small very quiet endpoint and i'll set the invisible screen to shut down after 5 minutes inactivity to keep things as quiet as possible.

 

[Triode User]

I tried chromecast with roon expecting a step up in SQ due to the small form factor vs optical from my imac but i was wrong. The chromecast sounded very thin, brittle and lacking depth and this was immediately apparent. After more testing nothing changed so i will return it. My imac is my core but i wanted a smaller footprint and i was lucky enough to find a late 2014 mac mini which i will use as a roon bridge optical out to my mscaler. It was unused brand new boxed and i made a deal and got it for less than half the market price. This will be my front end for my system imac core and mac mini optical bridge. The mini is a lovely small very quiet endpoint and i'll set the invisible screen to shut down after 5 minutes inactivity to keep things as quiet as possible.

Did you have any thoughts about what was causing your experiences with CCA, after all it is digital and is connected via optical . . . . . ? (Genuine question by the way.)

 

[musickid]

It's beyond me and i was surprised. I was hoping the chromecast to improve vs optical from my imac but the difference was night and day the chromecast was thin sounding with no depth. I know my imac has a cirrus logic chip with dac/spdif out and apple have chosen a high quality 3rd party supplier here especially as a 1.5K apple computer ships with a high quality audio implementation as expected from apple. That's all i think of as an explanation here. The mac mini builds on that. In fact the specialised servers costing much more when examined don't really offer much more than what is offered by a late 2014 mac mini at a fraction of the cost.

 

[audio_1]

I've never heard any sound science on single ended being inherently better than balanced. Unless you use balanced cables on a single ended designed.

This is an interesting article in Stereophile Magazine by Martin Colloms comparing balanced and single ended connections. https://www.stereophile.com/features/335/index.html

"True, the balanced condition does result in lower noise levels and improved immunity to EMI and RFI, local ham radio, CB operators, or switching pulses from heating or refrigeration thermostats. On the debit side, stereo images often lose absolute focus, stage width, and depth. Sounding less "locked in" to the music, the balanced components often exhibit losses in dynamic resolution, dynamic contrast, and rhythm. Typically smooth-sounding and laid-back, a balanced component can be less involving, lending the music a "downbeat" impression."
Read more at https://www.stereophile.com/content/balance-benefit-or-bluff-page-3#s3e2KIQ6xkiKYr42.99

"In practice, however, the consequences of such freedom appear to be losses in absolute sound quality, particularly in the areas of "foot-tapping" involvement and dynamics. Perhaps there's a lack of rigor and critical assessment in the design of balanced components. Maybe good performance is taken for granted. The adaptation to balanced working is unwittingly used as a problem-solving crutch; the technological benefit obviously lies in easily produced, impressive specifications for signal/noise. But what about sound quality? We still cannot adequately measure that."
Read more at https://www.stereophile.com/content/balance-benefit-or-bluff-page-4#kFiEG1ctRe8a1mwv.99

 

[Rob Watts]

Could you elaborate a little why you are comparing wta2 to mscaler?
And is that wtA from a certain device or are all implementations of wta2 the same?

I actually had in mind the differences between WTA1 (48 to 768 or 1FS to 16FS) to WTA2 (768 to 12.288 MHz or 16FS to 256FS) in general rather than the M scaler. WTA1 we get big changes in bass, pitch, timbre and instrument separation - but WTA 2 only affects the perception of starting and stopping of notes. Having said that, I only have one WTA 2 filter designed, as producing an output every 88nS needs a lot of processing.

Thanks very interesting indeed to read.
My VERY limited knowledge of digital theory does not allow me to follow and understand everything in absolute detail. But I DO know what can hear via my HMS/Qutest and it has fundamentally changed my take on rbcd 16/44.1.

This morning I have just listened to an early digital BIS recording from Gothenburg with the GSO playing Sibelius's 7th symphony with great pleasure!
The recording was made with only THREE Neuman mics in the good acoustics of the Gothenburg hall with SONY's PCMF1 in 1984!
And what I have just heard sounded almost like hearing a mic-feed from that hall.
Playback via PCM F1 did not sound that good in those days.
Strings could sound a bit steely via that ADC.

I am amazed on a daily basis at how many more recordings I can now really enjoy not only for their musical and performance qualities, but also how very realistic 16/44.1 can sound via HMS.

I mentioned those early digital recordings in some posts here after my return from Asia earlier, and via the stock BNC they did NOT sound as good as I had hoped. But now via upgrade BNC cables they do.
And the difference is not subtle to me.

The coherence and realism,how this orchestra actually sounded playing live in their good hall with these early three mics only 16/44.1 recordings via HMS actually sound more realistic to me than BIS's multimic'd Sibelius symphonies recent remakes from Minnesota.
I was invited to those sessions too, but I would have had to pay for my own flight so I declined the offer.
But I have been to quite a few recordings sessions and many concerts in Gothenburg.
It truly sounds as if I am listening to the analogue signal before digitizing with many of those early cd's now.

But yes really well recorded hi res can sound even more realistic, which brings me to the question of which ADC can record at 768khz?
Didn't you mention 256M taps needed to recover 24 bits to the same level as 1M taps can with 16/44.1 two years ago?

The only PRO ADC's which I am familar with are still limited to 24/358 or DSD 256 which I have heard in use at recording sessions.
Have you already made test recordings with a prototype Davina or from which ADC did you get native 768 recordings?
Cheers CC

The RME ADIPRO2 is a 768k ADC. I think this was the one used on my test recordings. Davina isn't recording yet!

Hi Rob,
Thank you for your detailed answer. For the most part this makes sense to me and if I understand your response it boils down to the following:

With the M Scaler we can reconstruct the incoming signal back to the original 16 bit bandwidth limited analog signal and therefore we can resample it at a rate grater then 700khz. By doing this our samples now occur once ever ~1.3uS instead of once ever 20+ uS, and given that the brain can perceive shifts as small as 4uS and subjectively even less this results in a perfect upscaled digital stream from the M Scaler to the DAC with much much smaller transient errors. Then the DAC takes it from there, and because it is operating off of the new upscaled digital stream and there is much more information in that stream (as it has been reconstructed) and the resulting output is "better".

This does bring me to one new question which came up when I read your response about the WTA filter in your dac. If I understand correctly, your DACs (Dave, Hugo TT2, ect...) then oversample the resulting stream to 12,288 kHz, when this occurs is it also approximating a sinc function to a particular bit depth, or is it doing something else entirely?

Thank you again for the detailed response Rob, I really appreciate it.

Technically, the M scaler doesn't create more information from a strict mathematical sense, but preserves the information content as it changes from a sampled discontinuous state to a continuous waveform. It's hard to see that, as we are filling in the missing bits, but it's because the signal is bandwidth limited - so there is no extra information above FS/2. And WTA2 is sinc like but that's not the end of the process - a three stage filter then filters from 256FS to 2048FS, but this filter is IIR type rather than the WTA being an FIR filter.