A novel way to massively improve the SQ of computer audio streaming

[afrancois]

On 7/13/2017 at 3:48 PM, Johnseye said:

 

Good stuff Rajiv.  Lots of changes going on.  When I hear you say "finally banished the last remnants of the slightly thin SOtM house sound" and "these just vanquished the last trace of thinness" I think to myself, what are we really trying to achieve with these devices?  If we think in painting terms, are we just painting different shades or colors of the music to achieve a sound that is more pleasing to our ears?

 

What is each reviewers ultimate goal when listening to these devices?  Are we looking to reduce noise as much as possible, thus revealing as much of the music as possible, are we looking for a sound that most closely represents live or an analog sourced recording, or are we looking for a sound that is most pleasing to our ear and does that qualification change?

 

There are certain components that contribute to a thin sound.  Solid state vs tube in the pre or amp is one.  Cable material is another.  And I wonder how you came to the conclusion that SOtM products sound thin because I've come to that same conclusion myself.  What is your point of reference in which a device sounds fatter that you are measuring against?  I found that the microRendu sounded fatter.  However the sms-200 was slightly clearer while also sounding thinner at the same time.  I think it may be hard to get one without the other, but I chose thin and clear vs. fat and clouded.  Analog music such as that from vinyl is one exception.  I think that can be clear and fat.  Almost buttery, but it's also noisy.  When I listen to music that's my reference point.  I can hear the difference between an analog sourced piece of vinyl and a digital sourced.  It's not hard, one sounds thinner than the other.  My end goal is to hear digital music as clearly as possible, with as little extraneous noise as possible, providing a clean, transparent, multi dimensional image that is buttery like analog.  That's my point of reference and critical in understanding why I think a device, a musical source or a setting may sound better than another.  It's the compass that guides my audio decisions.

Very well said. What I have noticed is that when your systems gets better, it tends to get a bit thinner. Less jitter, less noise gives a more precise sound that we are simply not used to hearing from recordings. You could of course argue that this is not the case with live performances, but listen to the background noise during these live concerts! Good studio recordings have a black background we almost never encounter in our daily lives. My conclusion is that noise attributes in a perception of a fatter sound. Once you get used to the precise and perhaps a bit thinner sound you want go back. This also is true for real digital amplifiers. You need to get used to them. Btw. I only have a full digital pre-amplifier. My mono-blocks are analog.

[afrancois]

14 minutes ago, kennyb123 said:

 

I typically think of "thin" as the absence of weight or tonal density.

 

I totally agree about the more precise sound, but at the same time I tend to hear an increase in tonal density as a result.

 

The terms I'd probably use instead of "thin" to describe what I hear as a system gets better are "more focused" or "less smeared".  

I agree it would be better to talk about focus instead of thin. Thin however would be used be people less exposed to high end. Of course those are not the ones visiting these forums. 

[afrancois]

10 minutes ago, Johnseye said:

 

Words that come to mind specific to more focused or less smeared are "defined" and "clear or clarity".  As in, I can hear an instrument clearly now, it is more defined when with the other component it fell into the background, or was hidden.

 

Thin to me at least is different than this.  However it often accompanies clarity.  It's what I first heard when I changed amps or speaker wire.  The interesting thing is that that thinness tends to disappear as I become accustom to the sound and just the clarity remains.

I believe than burning in of equipment is important, however I also know that I have to get used to a new/better sound. I think at a subconscious level our brain needs some burning in as well. 

[afrancois]

7 hours ago, austinpop said:

 

Hi Alex,

 

Thanks so much for that very definitive and informational post! I feel so much better. I like the heatsink idea. Got any recommendations that'll fit?

 

The fact that I single handedly accounted for 2 out of your 3 field failures makes me feel so special. 

 

But in all seriousness, I want to reiterate that Alex has been absolutely great about correcting any issues, turning around and sending me replacements within a day of receipt of the faulty unit. I couldn't ask for better support.

 

http://www.audiophonics.fr/en/kits-modules-diy-radiateurs/heat-sink-aluminium-black-100x50x40mm-p-9260.html

 

I use two of these on each LPS-1.

[afrancois]

3 hours ago, R1200CL said:

 

How about trying Gophert ?

http://www.dx.com/no/p/gophert-cps-3205-5a-32v-160w-portable-adjustable-dc-power-supply-black-eu-plug-396460#.Waflkno8yhD

 

@Cornan is a big fan of those, and claims very good results. 

 

 

 

For a review see https://www.youtube.com/watch?time_continue=1642&v=yIx_VWWgKUI

 

I've just ordered one (the new CPS-3205C)

[afrancois]

42 minutes ago, Cornan said:

 

A tip to maximize the (minor) investment with the cps-3205II or 3205C (C is almost exactly the same except for the active FPC found in the C version) is to get yourself (or build yourself) Canare 4S6 starquad DC cables and use starquad ac mains cable for it. It makes quite a difference!  If you have a isolation transformer with floating secondary close by connect the Gophert to it. I promise you're in for a surprice! 

 

 

 

All my computer equipment and Hifi rig is connected to an APC UPS http://www.apc.com/shop/au/en/products/APC-Smart-UPS-RT-1000VA-230V/P-SURT1000XLI.  So my AC is completely regenerated from batteries.

Would I benefit from a Topaz Ultra Isolator?

[afrancois]

9 hours ago, auricgoldfinger said:

 

Thanks for the suggestion.  I have been using the Mean Well plugged into my BPT BP-3.5 Signature balanced power transformer.  I tried using the JS-2 set at 12V, and it brought the LPS-1 back up to a more comparable level with the sPS-500 + Pangea cable (plugged into the same transformer).  However, I still find the sPS-500 bests the LPS-1.  Sorry to be the bearer of that information.

 

Given the improvement, I am now planning to use the JS-2 to power my 2 LPS-1's unless I end up needing the JS-2 port to power other devices that need a linear power supply. 

Have you tried powering the LPS-1 with the sPS-500? I'm getting very good results doing so.

 

My sPS-500 with standard DC cable was clearly not as good as the LPS-1 with star quad. 

 

Now I'm waiting for the arrival of a star quad Y-cable for the sPS-500.

[afrancois]

6 hours ago, auricgoldfinger said:

 

Thanks for the suggestion, but honestly, I'm not interested in buying expensive linear power supplies to power other expensive linear power supplies.  IMO, they ought to demonstrate value on their own or there is no point in buying them.  I already have a JS-2 (which is the lowest in the performance hierarchy), so I am willing to use it to improve the LPS-1 ONLY if I have the spare capacity.  I would never consider using the sPS-500 (highest in the performance hierarchy) to improve the LPS-1.  I have much better uses for the sPS-500.

 

I see what you mean. I power the LPS-1 with the sPS-500 because of two reasons. First, the sPS-500 doesn't cut it for me with the standard cable. Second, when I use the sPS-500 to power the LPS-1 that in turn powers the sMS-200 ultra, I no longer have these very faint clicks. Something like a very small impurity in a vinyl record? I guess that my standard LPS, that I used to power two LPS-1 had some difficulties doing so and kicked back something in my chain.

 

[afrancois]

Just now, auricgoldfinger said:

 

You may want to consider an aftermarket power cable for your sPS-500.  See my comments here: 

and here:

 

 

Ghent star quad for the sPS-500 is underway :-)

[afrancois]

1 hour ago, mozes said:

This is very interesting stuff, a radical shift from all amps in the market! I hope Chord launches this new amp at CES 2018

This has been done before by Lyngdorf. Look at the TDAI2170 and the SDA2400 and even before that. Truly amazing gear. These are true digital amplifiers without the drawback of other so-called class D amplifiers. No coloration, at all and with built-in room correction. 

[afrancois]

5 hours ago, romaz said:

 

Much has been said about how differences among servers are perhaps greater than difference among DACs.  In my own experience, I both agree and disagree with this observation and I will attempt to explain.  In my view, the DAC is clearly the more important component and why some DACs sell for >$100k.  Because this thread was never meant to discuss DACs, I have shied away from commenting but given Moussa's post, I figured I should comment although this will represent my last post on DACs on this thread.  In fact, this post will represent the beginning of my exit from posting in forums in general.  Life has become too busy.

 

When putting together an audio system, people will have their priorities.  I have already stated mine and they are simply (1) resolution and (2) transparency.  My reference isn't vinyl or tape or the million dollar setups one can hear at RMAF, Axpona or Munich, my reference is the live music I am often exposed to.  Most of what I listen to is unamplified acoustical music, whether it be large orchestral, small ensemble, choral music, or solo instrumental (especially organ but also piano and guitar).  When I am at an acoustical performance, whether it be classical or jazz, the first thing I notice is the acoustics of a venue and the resonances that venue provides.  The natural reverb and decay of instruments and voices are quite evident and from the stalls to the balcony or from one venue to another, they will vary.  

 

It has been stated that the reverberation time in a large venue like Carnegie Hall measures between 1.8 to 2 seconds.  At the Alice Tully Hall in the Lincoln Center just a mile away, this more intimate arena has a shorter reverberation time of 1.4-1.5 seconds.  Which is preferable depends on whether I am listening to a solo guitar, four string quartet or a full orchestra but regardless, I very much enjoy hearing the acoustics of a great building and never would I prefer to hear music in an anechoic chamber.  This is where most DACs stumble and where I find the DAVE excels.  This is also where I find PCM superior to DSD.  DSD provides you an expansive and a soft "tube-like" sound but this softness, which can be a wonderful way of masking the harshness of many chip DACs also results in a diffuse and imprecise presentation with respect to depth and timing and my careful A/B of my own recordings has convinced me of this.  As someone who values the accurate spatial portrayal of a live musical performance, I have found that a good music server can provide much but a good DAC can provide more.

 

When talking about resolution, as we look at our PCM files, we are provided 2 types of information:  (1) bit-depth and (2) sampling rate.  For Redbook, this means 16/44 which translates to 16-bits of dynamic range and a sampling rate of 44 kHz.  While DR is important, I contend that sampling rate is much more important with respect to a DAC's abilities.

 

When people talk about dynamic range, most people think about how loud and dynamically a DAC can play when really, it's about how quietly a DAC can perform that is important.  With regards to DAC performance, Rob Watts equates DR to the "hiss level" of the DAC and the greater the DR, the less likely you are going to hear "hiss" when no music is playing.  There is a DAC (that I will not name) that sells for >$100k and boasts a DR of 173dB (or 28.8 bits of dynamic resolution) as if we should be impressed by this.  For those that know better, this performance metric is useless since most believe most humans are incapable of hearing beyond 21 bits of dynamic resolution.  Just as important, most ADCs are also limited to about 21-bits of DR and so when people talk about 24-bit recordings, they often don't contain a true 24-bits of dynamic range.  Even at 24-bits (or 144dB) of dynamic resolution, for those who choose to look at DR in the traditional way of how loudly something can play, listening to any sound at 144dB SPL would be considered lethal. Now this is what people fail to realize -- as soon as you connect DAVE (or any DAC) to an outboard headphone or speaker amp, you now have thrown away the DR capabilities of your DAC because now, you've buried the DR performance of your DAC into the much higher noise floor of your amplifier.  For those who use an outboard amplifier with their headphones or speakers (this means most everybody who do not own a Chord DAC), you're basically listening to the much more limited dynamic range of your amplifier which is typically between 16-18-bits.  

 

With regards to sampling rate, I will explain why I consider this to be the more important spec with regards to DACs and this is why most DACs cannot match the performance of the very best turntables.  Sampling rate gives you a measure of timing resolution and this provides you not just spatial information such as depth but also timbre accuracy and the layering of fine detail.  With analog sources, you are hearing a continuous waveform and SQ is limited only by the quality of the gear that transmits this waveform.  As such, it is generally easier to get great sound from an analog setup such as a turntable.  With digital, an ADC is responsible for sampling the analog waveform a specific number of times per second and the larger the number of samples that are taken, the fewer the gaps of missing information there are and the more fluid or "analog" the recording sounds.  In theory, a waveform that is sampled 176,000 times per second (hi-res PCM) will sound better than a waveform sampled only 44,000 times per second (Redbook).  If that waveform is sampled an infinite number of times, then from a mathematical standpoint, your digital file becomes equivalent to your original analog waveform but as we know, infinite sampling is not possible based on the technology we have today and so this would suggest that digital can never truly equal analog.

 

However, there is the practical matter of the limitations of human hearing that potentially make it possible for digital to equal analog.  Most scientists agree that the human brain/ear has the ability to discern 2 separate sounds if they occur at least 5-7µs (microseconds) apart and so this represents the limits of a human's auditory time resolution abilities.  This means that when 2 sounds occur 10µs apart, as an example, we can hear 2 discrete sounds but when these 2 sounds only occur 4µs apart, instead of hearing 2 discrete sounds, we hear only one blended sound.  This is the rationale for why digital sounds "discrete" and why analog sounds "continuous."  With Redbook, as previously stated, sampling occurs 44,000 times per second and this equates to a time resolution of 20.8µs.  Anyone comparing a CD to vinyl in a resolving setup should easily be able to discern that with a CD, information is clearly missing.  As you sample more often, let's say 96,000 times per second, time resolution improves to 10.4µs and while this represents a significant improvement, most ears will likely still be able to detect that an analog source provides more information.  When you use an ADC to sample a file 192,000 times per second, time resolution now improves to 5.2µs.  In theory, at this sampling rate, a digital file should sound virtually indistinguishable from the original analog wave form and so this is the basis for why hi-res files were created.  This would suggest a 24/192 hi-res PCM file should sound equivalent to the original analog waveform.

 

For those who have done careful listening, however, with most DACs, 24/192 does not equal analog and even DXD or DSD256 files still can't match the resolution of the very best analog setups.  At most audio shows you attend, when you ask a certain exhibitor to give you their very best presentation, if they have a turntable or a reel-to-reel present, quite often they will switch to their analog source and, in fact, I have witnessed this many times.  As a further example, having visited the Magico factory in Hayward, CA recently, they have arguably the finest listening room assembled in the world today.  This room cost them $250k to build and has the equivalent of a floating floor and no parallel walls to avoid standing waves.  Short of an anechoic chamber, it perhaps has the lowest noise floor of any listening room and they use this room as their lab.  In fact, it is how they voice their speakers including their $600k Magico Ultimates and their soon to be released $175k M6.  Here is a photo of that room:

 

 

Because Berkeley DACs are the local favorite, they use a Berkeley Reference 2 DAC (Berkeley is headquartered nearby) fronted by a Baetis Reference server.  However, when they wish to present their very best, they revert to their turntable.

 

The reason is not so much because this sampling theory is faulty but because ADCs have limitations.  It is the reason why such technologies like MQA were created and why many DACs oversample.  Those in the NOS (non-oversampling) camp suggest that NOS DACs sound more natural but NOS strives only to reproduce the best that the ADCs can offer, warts and all.  Oversampling is much more ambitious and strives to overcome the limitations of the ADC by interpolating the missing bits of information through the use of sophisticated mathematic filters.  If the oversampling is done perfectly, a 16-bit Redbook file originally sampled at 44kHz per second should be audibly indistinguishable from the original analog waveform and this is the basis for the long tap-length filters that Rob Watts has been championing for decades but also the basis for what HQPlayer tries to accomplish.  As to who does it better, I will leave it for others to decide for themselves but having listened to both approaches, I much prefer Rob's approach.  As to the benefits of oversampling to DSD vs PCM, people will have their preferences, I have already stated mine.  

 

Regarding why some people fail to recognize great differences between DACs, I hear this all the time and I believe there are several reasons.  As both a headphone and a speaker listener, I have found both types of listening to have their advantages.  Headphones have the ability to portray fine detail better while speakers can image and soundstage better.  

 

DAVE is unique because its headphone output doesn't utilize a separate headphone amp.  When you plug a headphone into DAVE's headphone jack, you are actually listening to the DAC itself.  This means your headphone is tapped to DAVE's full bandwidth, ultra low noise floor (-180dB), dynamic range, and time resolution.  Moreover, what is unique about DAVE is it has no noise floor modulation and so whether you listen to music at low levels or at DAVE's peak levels, noise floor remains at the same ultra low levels.  There is simply no cleaner, clearer, more transparent way of listening to music than this.  The problem with headphone listening is that headphones do not portray depth well, certainly not as well as speakers and so to this degree, a lot of DAVE's performance cannot be fully realized through headphones alone.

 

The problem with listening to speakers with DAVE (or any DAC) is that DAVE's performance is largely buried in the amplifier you use to drive your speakers.  While DAVE's performance still shines through, its performance is blunted as you end up inheriting many of the limitations of even the finest speaker amplifiers.   Just like with outboard headphone amps, no speaker amp can match the performance characteristics of your DAC and so what you get with even the finest amps is a diminished photocopy of the original.  

 

Throw in a preamp, no matter how good, and this further adds to a loss of transparency.  That is just the nature of adding components to your analog chain.  Unless you are using a preamp for sound tuning (ie tube linestages), or you have an amp that demands a certain preamp to function optimally or unless you have multiple sources you need to switch among including a turntable, with DAVE, the very best preamp is no preamp at all.  Just like with amplifiers, no preamp can match DAVE's performance with respect to distortion characteristics, noise floor, speed, dynamics, or time resolution.  Even more, Rob programmed into DAVE the ability to attenuate down to whisper levels with absolutely no loss in resolution.  That means that as you attenuate DAVE to its lowest level (-75dB), DAVE is still outputting full resolution, something that no preamp can match.  

 

In the photo below is VAC's very highly regarded Master preamp (about $30k):

 

 

Kevin Hayes, VAC's designer, was kind enough to allow me to compare my DAVE driving his wonderful VAC tube amplifiers both with and without his Master preamp:

 

 

It was the forgone conclusion of most people in the room that the sound through the attached Harbeth speakers would be vastly better with the Master preamp in the chain.  They were surprised when this was not the case.

 

Here is another example of a dealer's DAVE driving an $11k Constellation Inspiration Stereo Amplifier both with and without Constellation's $9k preamp.  To both the dealer's and my ears, SQ was better without the preamp and so when this dealer sells a DAVE, he no longer tries to promote the sale of a preamp:

 

 

And so what Moussa and ElviaCaprice are hearing is something that is very unique.  Through their high-efficiency speakers, they are hearing the full potential of their Chord DACs limited only by their choice of cabling and speakers.  With either the Omegas or the Voxativs I am using, I am hearing every bit of detail that my best headphones can provide while also the imaging and soundstage that only speakers can provide without the resolution and transparency robbing  impact of an outboard preamp or amplifier.  At the present time, I am trying out a pair of $25k Martin Logan Renaisssance Hybrid Electrostatic speakers in my large listening room, which I find to be very resolving and transparent.  These speakers are currently being driven by a pair of Pass Labs XA60.8 class A monoblocks ($13.5k for the pair).  While I cannot deny how wonderful this sounds when fronted by my DAVE, compared to DAVE directly driving my more modest pair of Omegas, this latter setup still sounds more resolute and more transparent.  This is possible only with Chord DACs because only Chord DACs (as far as I'm aware) have output impedances that are low enough to directly drive speakers.  In the case of DAVE, which has an output impedance of 0.055 ohms, this equates to a damping factor of 145, which is stellar.  Soon, Rob Watts will be introducing amplifiers that will connect to his DACs via digital interconnects (not analog ones) and will have the same resolution and transparency characteristics as DAVE directly driving speakers.  Essentially, these amplifiers will be "invisible" meaning they will have no character of their own.  They will have class A output and the first amplifiers will output either 20 watts stereo or 70 watts in monoblock form.  This technology is supposed to be scalable where 200 watts of amplification will be possible.  

 

Furthermore, as I have alluded in other posts, I have added Rob's new M-scaler to my DAVE.  This is incorporated into Chord''s new Blu Mk 2, which is a CD transport that also includes a USB and BNC SPDIF input.  This increases DAVE's TAP resolution to just over a million TAPS.  This is a milestone that suggests Redbook is now completely indistinguishable from the original analog waveform and Rob didn't believe it would ever be achieved when he first conceptualized it back in the 80s but because of the rapid advancement of FPGA technology, this indeed has been achieved.  Practically speaking, this results in a massive improvement in DAVE's resolution, so massive that the collective impact of my server mods which includes 8 clocks being replaced pales in comparison to what Blu Mk2 provides.  For those of you who own a Chord DAVE, I would suggest you prioritize getting a Blu Mk2 beyond anything else discussed on this thread.  Combined with Chord's upcoming "digital" amplifiers, there will be no more resolute or transparent way of listening to a digital file.  Despite all of this, I am finding, however, that the quality of the music server still matters.

Interesting post, however, what many people tend to forget is that most new albums that are pressed on vinyl, have been through the digital machine. Very few pure analog recording studio's still exist today. Even quite old recordings have been digitally remastered before they are pressed onto vinyl. Something to think about. Honestly, I never heard a top class analog system in my life. Something I should do in the near future, with the risk of being somewhat disappointed for the rest of my life. This already happened to me in the digital domain, the day I heard a Steinway-Lyngdorf model D. Sorry this is my second post mentioning Lyngdorf. Guess what I have at home :-)

[afrancois]

1 hour ago, Johnseye said:

 

It is easy to hear when a vinyl album comes from a digital source.  I typically don't bother buying these because you might as well be listening to the digital version.  There are exceptions but I won't digress.  There are some companies like Music Matters, Analog Productions and Mobile Fidelity who do an incredible job of reissuing analog material.  Otherwise finding the original recording in the best condition is your best bet.  Also, recording studios began mastering on digital media around 1975 growing over the years with CDs becoming mainstream in about 1982.

I find this is quite a bold claim, saying that you can hear that a vinyl record comes from a digital source. When you know how digital has evolved since 1984 and even then it was almost impossible to reliably detect the presence of an ADC/DAC. See http://www.bostonaudiosociety.org/bas_speaker/abx_testing2.htm

Don't want to start a discussion on digital vs analog :-) Both have their advantages and disadvantages.

 

[afrancois]

1 hour ago, romaz said:

 

Sure enough.  However, THD + Noise ratings provided for these amplifiers by the manufacturer are nearly 700x more than DAVE.  

Have ever heard a Lyngdorf system? Their room correction is unrivaled. Much more important than squeezing out the last impedance or THD if you ask me. The most important factor in music reproduction is the room you are in, period. Any idea how much THD your speakers add? Those are full percentages instead of... That's why I would love to have an active speaker system. Unfortunately this is beyond my reach.

[afrancois]

1 hour ago, romaz said:

 

You make very good points, especially about the THD of speakers.  I have not heard a Lyngdorf although now, I would very much like to.

I'm using the poor man's Lyngdorf equipment :-). However, I'm quite pleased with this. When you have the change, you should audition the Steinway Lyngdorf Model D. http://www.steinwaylyngdorf.com/en/products/speaker-series/model-d It's the combination of room correction and an active speaker system that makes this system uncanny. Strangely they don't offer a streaming solution, as far as I know. Thank you for being open-minded.

[afrancois]

Hans Beekhuyzen goes berserk over the sMS-200 ultra. I too was really surprised when I heard the improvements over my standard sMS-200.

 

https://www.youtube.com/watch?v=nCoFIdPLcUk

[afrancois]

7 hours ago, austinpop said:

Over on the Lush thread, someone mentioned how they knew their system was sounding good when they could listen to Adele's 21 without wincing. I'm paraphrasing.

 

That got me thinking. After all,  the clock improvements in trifecta medley have so dramatically removed that digital haze and fatigue. It's been years since I listened to 21. Would my system be up to the Adele test? 

 

I'm happy to report - YES! I ended up listening to the whole album. On my system. With HD800 headphones, that do not suffer bright recordings gladly. The whole album.

 

If that doesn't epitomize what we mean by smoothness and lack of digital glare, I can't imagine what does.

What I have noticed as my system got better (less jitter) is that highly compressed music became listenable. Adele’s CD's are highly compressed and her voice has been digitally manipulated to emphasize the typical Adele sound and this at a point that it has become ridiculous on Adele 25.

[afrancois]

2 hours ago, AmusedToD said:

 

Well, I think you should listen to some true high end DACs that will smash any DSD upsampling voodoo with simple redbook playback. In fact - there is no comparison at all. As a matter of fact, the Ifi DAC will look and sound like a toy next to those machines, but we are talking very big bucks. 

 

My computer (i7 iMac) can also upsample to DSD512, and while not bad, it’s not necessarily better than regular non upsampled PCM, it’s just different in presentation (and often “softer” than PCM, lacking in energy and slam). Not my cup of tea.

I agree. If a DAC sounds better playing DSD, it is because PCM playback is badly implemented on that DAC. Sadly most DAC designers have been concentrating on DSD, because it is hyped as the next big thing. Of course people may prefer DSD, the same way others prefer vinyl because the way it sounds.

[afrancois]

38 minutes ago, lmitche said:

I certainly don't want to get into another tedious dsd vs. pcm debate. My post was about where to do upsampling, not about pcm vs. dsd upsampling. Plenty of people upsample pcm with Hqplayer and Audivana.

 

BTW, it would be unacceptable if energy was lacking in the presentation here.  It is not.

 

While I appreciate the concept that a dedicated purpose built audio computer sounds better, my experience with various purpose built streaming Linux boxes was not convincing.

 

And yes the Hugo is not 5x the price of the microIDSD, but I was talking about extremes, not the massive grey area between the two points. Sorry if I didn't make that clear.

Don't know what Linux streaming boxes you are referring to, but it can't definitely be the sMS-200 ultra.

[afrancois]

26 minutes ago, Johnseye said:

 

The clock replacements, software tweaks, USB cables, gadgets and high quality external PSUs are in order to reduce noise and in some cases improve timing.  A completely separate effort from upsampling.

 

My DAC sounds amazing with PCM 192.  It's only capable of DSD64.  It doesn't really do DSD imo and what I've heard of DSD is like others have commented, a bit too soft for my taste.  This is personal preference and if I had a DAC that excelled at DSD512 maybe I'd like that.  I don't know right now and can't have it until I buy a new DAC.  One thing to also consider is how you design the sound of your components.  If all you do is listen to DSD you're selecting components that fit that sound signature because you like it.  Or, if you find it may be too soft you select components to brighten it up.

 

 

 

In my limited understanding upsampling is going to get you to that higher frequency.  It's the filtering that HQPlayer includes which makes the biggest difference and why so many flock to it.  There can be filters built into the DAC as well.  Perhaps the upsample to a higher frequency aside from the filtering can be done on one device better than the other, and given the high resource requirements of DSD512 on a PC I suspect the more horsepower you give it the better.

 

The downside of that is you introduce noise.  Here I think is the crux of Larry's point.  He prefers the benefit he receives by upsampling to DSD512 over all the things one can do to reduce noise as much as possible.  To each their own.  I suspect if I liked the sound of DSD better I would make sacrifices myself.  To your point Elvia, he would still benefit from noise reduction where possible without sacrificing his upsampling performance.  No matter what, noise elimination is a benefit.  I just slapped in some of that 3M EMI-RF absorber in my PC and DAC and I can hear a clear significant improvement.

What exact type of 3M paper did you use? I myself was thinking of AB5100SHF. So many different types to choose from!

[afrancois]

50 minutes ago, Johnseye said:

 

Yes, AB5100SHF.  I bought 2 sheets which was just enough for my PC and DAC.  I'll have to buy more if I choose to do this with other equipment.

Thank you for confirming.

I just did a small test with a portable radio set to LW 144 Khz and I can confirm that my audio equipment is subject to much noise. The closer I get to the electric distribution box, 24 port router, security camera boxes and the cable modem the more noise I get. Almost all my audio equipment is in the garage. There are clearly quieter places in my house. So I hope that the 3M paper will help.

[afrancois]

7 minutes ago, Johnseye said:

 

It will help.  I was surprised how dark the background is by comparison.  The instruments stand out more as a result.  My equipment is side by side and stacked and surrounded by 3 speakers and a sub.  A lot of cabling and equipment in the area.

 

I'm also tempted to try ferrite on some cables.

 

My only question and concern is can there be too much shielding?  That was something Roy mentioned Lee stating; not to overdo it with the EMI paper.  How could it be overdone and to what effect?

Where did you put the paper? On the inside of the cover of the DAC? I've just ordered 3 sheets :-)

[afrancois]

5 minutes ago, BigGuy said:

Not sure how much "shedding" there might be from the ABxxxx sheets but I know there is a cautionary statement on using ERS paper that it contains conductive fibers which potentially cause problems with circuitry.  I actually put mine into clear sheet protectors before placing on top of components.  The ABxxxx do contain metallic particles  which may have the same issue.  There are some of these sheets which are a foil which I would expect to have less of a problem in this regards.

The datasheet clearly says non-conductive. But you are right, it contains metal flakes.

[afrancois]

4 hours ago, Johnseye said:

 

I specifically liked this comment.  Looks like one can't overdo it based on this person's experience.  Although there is the risk from increased heat.  Something I'll have to monitor in my DAC.

 

It's also worth noting that when you cut the sheet the cut edges are conductive but the face of the sheet is not.  This guy kyrill recommends taping the edges in certain situations.

 

"Crucially, though, computer transports aren't producing analog output. We don't need to worry about tailoring a frequency response or spoiling a pre-set balance. We're looking only to tame and contain the white noise melée of RF radiated by the board and its power supply that will otherwise bounce around the case and feedback into signal paths. The more absorption you use, the quieter the electrical environment of the case and the closer we reach the holy grail of the computer delivering 'only noughts and ones'. It's hard to see (and hear) how you can overdo absorption in a digital device."

Does this mean you will be inserting some 3M in your sMS-200?

[afrancois]

I received my 3 sheets of 3M AB5100SHF-210X297 today and used them around the sMS-200 ultra, MC3+ USB and even under some of my LPS-1’s with very good results.

I’m so satisfied that I immediately ordered extra 5 sheets. I still got a lot of equipment that could potentially benefit from this miracle paper. DAC’s, power amplifiers and headphone amplifier.

Perhaps not everybody will benefit as much as I do. Most of my equipment is in the garage and it seems a rather noisy environment.
 

[afrancois]

27 minutes ago, BigGuy said:

From what I see, 3M sells sheets for specific applications.  Looks like AB5100 is for EMI rather than RFI.  Not questioning your observations but would not a garage be more likely to be RFI environ?

I know a bit confusing, but this is what Wiki says:

 

Electromagnetic interference (EMI), also called radio-frequency interference (RFI) when in the radio frequency spectrum, 
is a disturbance generated by an external source that affects an electrical circuit by electromagnetic induction, 
electrostatic coupling, or conduction.

 

And another explanation:

 

EMI (Electromagnetic Interference) is also called RFI (Radio Frequency Interference). Although the terms EMI and RFI are often used interchangeably, EMI is actually any frequency of electrical noise, whereas RFI is a specific subset of electrical noise on the EMI spectrum. RFI is a disturbance that affects an electrical circuit due to either electromagnetic conduction or electromagnetic radiation emitted from an external source. The disturbance may interrupt, obstruct, degrade or limit the effective performance of the circuit. The source may be any object, artificial or natural, that carries rapidly changing electrical currents, such as an electrical circuit or the Sun. There are two types of RFI. Conducted RFI is unwanted high frequencies that ride on the AC wave form. Radiated RFI is emitted through the air. There are many pieces of equipment that can generate RFI, variable frequency drives included.