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

[auricgoldfinger]

On 9/4/2017 at 1:50 PM, zoltan said:

Then I'll wait before placing an order for the sps-500

 

I did a quick A/B comparison of the sPS-500 and JS-2 using the Pangea power cable.  To me, the sPS-500 is still the clear winner.  Richer sound, more texture, better micro-dynamics, and sparkle are the thoughts that immediately come to mind.  The JS-2 sounds somewhat flat and less detailed in comparison.

[auricgoldfinger]

4 hours ago, JohnSwenson said:

What are you powering the LPS-1 from? Have you tried powering it from the JS-2? If you use the Mean Well you can get high frequency noise getting into your system back through the AC mains. Some systems are sensitive to this and others are not.

 

John S.

 

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. 

[auricgoldfinger]

4 hours ago, afrancois said:

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.

 

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.

[auricgoldfinger]

3 minutes ago, afrancois said:

 

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.

 

 

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

and here:

 

 

[auricgoldfinger]

I would like to share my impressions from adding an isolating transformer with a floating secondary along with a DC voltage regulator/power supply to my system.  Thanks go to @Cornan for posting the concept and recommending the Gophert power supply (I am using a CPS-1610 16V 10A model) and @austinpop for recommending the BK Precision 1604A isolating transformer.

My objective was to improve the power supply to my Synology DS213+ NAS, where all of my music files reside.  Frankly, my expectations were low, but since the financial outlay was relatively small, I decided to give it a try.

I received the isolating transformer first.  There was a perceptible improvement, but I was not especially excited about the result.  However, after installing the Gophert, I heard a very dramatic improvement in sound quality.  Music became much more detailed and dynamic, images sharpened, and the soundstage deepened.  I heard the magic to which @Cornan has previously alluded.  (I am still only using the stock cable for the isolating transformer.  @Cornan says to expect further improvement from a star-quad ac power cable.)

I have now ordered a Tripp Lite IS500HG (with 4 outlets) and another Gophert for my router.  Before setting it up with the router, I plan to experiment using it to power my microRendu, ISO REGEN, and tX-USBultra.  The mR and IR are currently powered by separate LPS-1’s, and the tX-USBultra is powered by an sPS-500 with Pangea ac cable.  I am anticipating some interesting results.

[auricgoldfinger]

13 hours ago, austinpop said:

BTW @auricgoldfinger - what is the path from your NAS to your mR?

 

What components and cables?

 

You forced me to finally complete my profile.  Hopefully, you'll find the answers you seek there. 

 

I also have in my possession a Linksys switch with SOtM mods that will be clocked by the tX-USBultra as well as a NIC for the Windows 7 PC that will be used for a network bridge.  I plan to install these items in the very near future when my schedule permits.

 

Hopefully, I will complete the SOtM trifecta in 2018 with an sMS-200Ultra (silver wire mod).

 

[auricgoldfinger]

3 hours ago, str-1 said:

This is good to hear.  I have a Synology DS216j, which is still being powered by the supplied adapter.  I think I will buy this Gophert model, which as you say is relatively inexpensive.  What dc lead are you using to connect it to the Synology?  

 

I don't think BK supply isolation transformers with UK voltage so I will have to look for something similar.

 

I bought my cable from Ghent Audio.  You have to email Ghent at [email protected] to make the arrangements as this cable is not listed on his website.  Tell him that you want a custom 4S6 DC cable with DC-2.1G to 2 x banana plugs (4mm) and let him know the length you need.

 

Somewhere on CA I saw mentioned a UK site for isolating transformers.  Maybe @Cornan will chime in with a recommendation for you.  I think he knows the company name.

[auricgoldfinger]

9 minutes ago, jaspal kallar said:

 

 

Is This the Ghent DC cable  i.e 4S6 DC cable with DC-2.1G to 2 x banana plugs (4mm)  ?

 

Yes, that's the one.  You need the DC-2.1G plug.

[auricgoldfinger]

Keep an eye out for an official announcement from SOtM regarding their sCLK-OCX10 reference clock.  The specs should be released in the next several hours.

[auricgoldfinger]

6 hours ago, austinpop said:

 

I've seen the specs. It looks very promising. I hope May releases the details soon.

 

Here they are:  http://www.sotm-audio.com/sotmwp/english/portfolio-item/sclk-ocx10/

 

Seems like there are 3 options.

[auricgoldfinger]

8 hours ago, Boomboy said:

Hey there first post .. always enjoyed reading this thread . 

Wondering if this new sclk-0cx10 will be an update for a existing sotm product or a stand alone unit ? If anyone knows ? 

I own a sms 200 and about to purchase their d-link switch mod without clk output. Thnx 

 

It's a stand alone unit meant to provide a reference clock for devices that have a master clock connector.  There are different possibilities for adding a master clock to your system.  You could have SOtM add a clock card with a master clock connector to either your sMS-200 or the switch.  I'd recommend discussing the options with May.  The best solution should also reflect any planned future acquisitions (such as the sMS-USBultra) that you may be considering.

[auricgoldfinger]

1 minute ago, austinpop said:

 

And here is the press release:

http://www.sotm-audio.com/sotmwp/english/press-release-rmaf-2017/

 

It looks like you can configure your choice of quality/performance upon purchase to Standard, Advanced, or Reference. And - it will have 4 outputs!

 

On first glance, the Reference phase noise values match or exceed the Ref 10, but as always, how it sounds is what matters.

 

I asked May about pricing, but she asked us to wait a bit longer as they were not ready to reveal that yet.

 

I'm really looking to the demo in RMAF.

 

I'll be very interested in your impressions!

[auricgoldfinger]

9 minutes ago, Bruce Orr said:

Where would the new sclk-0cx10 clock fit in with a Mac system with Ethernet, router,  uRendu and ISO Regen using LSP-1 power and a DAC?

 

Thanks for the help. 

 

Based on your description, none of your devices are able to accept a master clock input.

[auricgoldfinger]

For me, the QX-5 is not a viable option because it is limited to DSD 128 as DoP.  I only use native DSD 256 for my numerous DSD albums.

[auricgoldfinger]

4 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.

 

Truly an epic post!   Thanks so much for taking the time to compose it.

[auricgoldfinger]

20 minutes ago, lmitche said:

Thanks Johnseye,

 

I was curious to learn if the Paul Hynes LPSes are powerful enough to support an Hqplayer upsampling machine.  Looks like the answer is no.

 

Your quick, clear and comprehensive response is appreciated.

 

How much power do you require for HQP upsampling?

[auricgoldfinger]

Seems like a good opportunity for SOtM to introduce a bespoke power supply. 

[auricgoldfinger]

1 hour ago, austinpop said:

Darn it, @lmitche and @unbalanced output, now you've got me thinking.

 

The last time I tried HQPlayer was before the clock chains, when I was on the sMS-200. I should try it again, to confirm for myself what I'm hearing now. But before I do, a couple of questions.

 

What percent of your listening is Redbook content? In my case, I may be atypical. While my library is about 70% Redbook, my actual listening tends to be <50% Redbook. I gather the benefits are most pronounced with lower-res tracks?

 

Also, Larry, regarding this:

 

 

Last time I tried it, I remember 4 parameters (none, none, none, none) that you had to manually tweak via pull-downs to vary the settings. Can you tell me how (maybe via PM) one can set up HQP to auto-detect PCM or DSD tracks, and conditionally upsample PCM tracks to 384, and DSD tracks to 128, without intervention?

 

I'm not sure I can download another trial version of HQP, without having to buy it.

 

I've attached my HQP settings to help you get started.  I am setup for auto upsampling to PCM384 and bit perfect DSD256 which is the maximum my DAC supports.  I agree that you should use it with Roon, but there is the annoying problem of the HQP process often stopping when file formats change.  You'll have to force close HQP when that happens.

 

@Miskawill probably give you another trial since your first one was so long ago.

 

[auricgoldfinger]

8 hours ago, feelingears said:

Akin to what @austinpop said about assembling hardware (vs. software), I'm leery of HQP's apparent complexity and user unfriendliness despite (or because of) my work in software. But this is audio so one is always exploring or tempted to try more. So, begging everyone's pardon, but would someone please point me/others to a good "HQP for Dummies" post or set of instructions, anywhere? Something user-friendly and more "forgiving,"  if such a thing exists?

 

(Also, HQP is a one-man show is it not? I wish there were more "there, there" as a going concern.)

 

Thanks in advance.

@seeteeyou You just beat me to the punch, but I'll have to see if any of those speaks to HQP (and speaks in "Dummies" dialect). 

 

Start here, and then follow up by reading the manual starting on p. 8 to get more detail on filter and sampling settings.

 

 

[auricgoldfinger]

9 hours ago, austinpop said:

I discussed it further with him at RMAF, in the context of where I go next from my trifecta medley.

 

What were Roy's suggestions for your next step?

[auricgoldfinger]

15 hours ago, austinpop said:

I am hoping some local folks, like @limniscate or @atxkyle go down these paths, so we can do easy comparisons! Or maybe you, @auricgoldfinger? You're not that far from Austin!

 

I'm still trying to get caught up to you by first completing my trifecta. 

[auricgoldfinger]

9 minutes ago, austinpop said:

Should I ground the output of the Meanwell, the LPS-1, or both?

 

@JohnSwensonMy question, exactly.  Plus, how does one ground the output of the LPS-1?

[auricgoldfinger]

17 minutes ago, JohnSwenson said:

I hope this makes things clear, I'm running out of ways to say this.

 

I find having everything on this topic in one post to be very helpful.  It seems definitive to me.  Thank you!

[auricgoldfinger]

Changing the subject...

 

What kind of IEC ac cables are people using for their Paul Hynes SR5's and SR7's?

[auricgoldfinger]

Just now, Johnseye said:

 

I used a standard 3 prong.  I'll upgrade it to something better in the future, but I'm not sure there will be much of a benefit.  Paul doesn't sell boutique IEC AC cables which may indicate they don't have a significant impact if any.

 

I have some Shunyata Venom HC cables I can try with it.

 

Paul told me that he doesn't sell them because most of his customers already have their own preferred high end leads.  Interestingly, he uses DC10FS woven power leads himself, but as they are solid core, they will not pass the electrical regulations for the construction of flexible IEC leads in place in Europe and other countries around the world, so he can't sell them.