[marcmccalmont]

"Should" we take those words seriously when the word "should" appeared twice in a row?

If breaking a ground loop were actually all we need to do, maybe we "should" simply spend 50 bucks and call it a day?

https://ifi-audio.com/portfolio-view/accessory-idefender3-0/
https://www.amazon.com/iDefender3-0-USB-Ground-Loop-Eliminator/dp/B01N3XKOLG

Actually I am using one between my Aurender n100h and Hugo2 a small improvement to my ear. I will try a usb to toslink converter too see which works better

 

[flyte3333]

"Should" we take those words seriously when the word "should" appeared twice in a row?

If breaking a ground loop were actually all we need to do, maybe we "should" simply spend 50 bucks and call it a day?

https://ifi-audio.com/portfolio-view/accessory-idefender3-0/
https://www.amazon.com/iDefender3-0-USB-Ground-Loop-Eliminator/dp/B01N3XKOLG

Ah, now we're getting deeper into the weeds.

This won't block all leakage currents (and their loops) the way optical and battery PSU's do. I've posted links to John Swenson posts on this a few pages ago too, explaining these leakage current loops and their link to RF. Do check them out if you're interested. It links directly to Rob's points above.

Battery PSU's blocks leakage current loops (there's zero connection to mains power) and so does optical.

Dave sounds fantastic on all inputs with no tinkering needed. But my own tinkering has found even better performance squeezed by blocking all leakage currents getting into/through Dave (or any DAC on the planet) - no need for ferrites either if you can block this leakage loop, for the reasons Rob mentions in the those 3 points I posted - no RF getting into Dave.

And as Rob himself has said, RF (linked closely with leakage currents) is like a fungus if it gets into the DAC.

 

[Articnoise]

Yup this is exactly why I said just a few posts earlier that this solution (or any solution) isn’t guaranteed to automatically sound good or best in everyone’s system.

I tried a little iPod Touch - by design they are power frugal.

The beauty is it’s probably free for most of us to try - we all have a laptop or mobile or tablet.

Blocking leakage currents may be the lesser of the evils and it may not be. As I’ve said many times and @elviscaprice mentioned just above, there’s no free lunch with any solution.

Just give it a crack and decide with your ears.

Even if there were some measurements post of some sort, each persons system is so different, there are so many different combinations of gear and they all interact differently.

Yep there’s no free lunch, but some meal can be culinary pleasures while others are just fast food that may cure your starvation then you are in a hurry. Therefore choose your lunch wisely – you only live once.

 

[flyte3333]

Yep there’s no free lunch, but some meal can be culinary pleasures while others are just fast food that may cure your starvation then you are in a hurry. Therefore choose your lunch wisely – you only live once.

Absolutely and Dave is one pleasure (of many) that everyone has to enjoy at some point in their life

 

[doraymon]

What we have been discussing on this thread is very interesting but also very complex for non pros like me.
Hans Beekhuyzen has a nice channel on YouTube where he talks, among other things, of what he calls "Audio Hygiene".
The idea is that you don't need to graduate in medicine to be able to learn a number of "Hygiene Rules" to preserve your health.
It would be nice to for us to have a book with "Audio Hygiene" rules that cover the largest possible issues deteriorating the sound quality of our Dave.
Any volunteers? Rob?

 

[Articnoise]

Absolutely and Dave is one pleasure (of many) that everyone has to enjoy at some point in their life

Great that you fund something that you truly enjoy and that put a smile on your face, even if it’s not exactly free but OTOH there’s no free lunch – so rock on .

 

[yellowblue]

What about balanced power conditioners to solve the problem of ground-loop current?
Like this
https://www.plixirpower.com/products/elite-bac-400
"It uses a custom single-winding balanced power transformer to reduce ground-loop current and noise to negligible levels, making the PLiXir Elite BAC the ideal pick for electrically sensitive equipment"
I own it and I like what I hear.

 

[Thenewguy007]

What about balanced power conditioners to solve the problem of ground-loop current?
Like this
https://www.plixirpower.com/products/elite-bac-400
"It uses a custom single-winding balanced power transformer to reduce ground-loop current and noise to negligible levels, making the PLiXir Elite BAC the ideal pick for electrically sensitive equipment"
I own it and I like what I hear.

I haven't used that, but have used isolators & power conditioners & they didn't really lower the noise floor or do any significant sound changes (other than maybe for the worse).

Anyone try Grounding blocks?
Entreq & a few sell criminally overpriced ones. There top range one is like 9 grand for what essentially is a plain wooden box with dirt inside it.

 

[doraymon]

I haven't used that, but have used isolators & power conditioners & they didn't really lower the noise floor or do any significant sound changes (other than maybe for the worse).

This is really interesting. I'm wondering if my Consonance PW-1 filters are doing a similar effect and I never realised it...
They've been a trusted part of my system for a long time and I never bothered removing them to check the effect.

 

[flyte3333]

For those not using Roon and looking to try streaming from computer to a mobile to Dave DAC over WiFi, I've tested the Neutron Music Player which is available for both iOS and Android...

And it turns your mobile device into a UPnP renderer/endpoint.

One thing I was concerned about is if it's capable as a bit perfect endpoint, because there's all sorts of DSP settings in the app and no simple 'DSP off' switch, the way Roon has.

While I'd personally prefer MQA go away, one very cool thing is you can use it as a bit perfect testing tool. I happened to have a used Explorer2 lying around and it finally came in useful

So playing from Audirvana (a Tidal MQA 353kHz album), to the Neutron Music Player over WiFi, if the entire chain up to the Explorer2 input was not bit perfect, then the infamous blue light wouldn't turn on.

But yes, I get the blue light so can conclude Neutron is a great bit perfect UPnP endpoint. It takes a bit of tweaking in the settings though, to turn off all the DSP settings.

The Explorer2 can't be powered by an iPhone or Android phone alone since they only supply 100mA power, but with the iPhone being charged by a 2.1A powerbank, no problems connecting to the iPhone.

So there you go, one practical use of MQA is a bit perfect playback testing tool, if you find a used Explorer2 on eBay like I did !

A DSD256 album from NativeDSD.com is another good way to test for bit perfect playback also (and test your WiFi network) played to a mobile and via DoP to Dave. If the chain isn't capable of bit perfect playback, you'll get dropouts with Dave with DoP.

 

[Gomiki]

There are actually two independent issues going on with DSD that limits the musicality - and they are interlinked problems.

The first issue is down to the resolving power of DSD. Now a DSD works by using a noise shaper, and a noise shaper is a feedback system. Indeed, you can think of an analogue amplifier as a first order noise shaper - so you have a subtraction input stage that compares the input to the output, followed by a gain stage that integrates the error. With a delta sigma noise shaper its exactly the same, but where the output stage is truncated to reduce the noise shaper output resolution so it can drive the OP - in the case of DSD its one bit, +1 or -1 op stage. But you use multiple gain stages connected together so you have n integrators - typically 5 for DSD. Now the number of integrators, together with the time constants will determine how much error correction you have within the system - and the time constants are primarily set by the over-sample rate of the noise shaper. Double the oversampling frequency and with a 5th order ideal system (i.e. one that does not employ resonators or other tricks to improve HF noise) it converges on a 30 dB improvement in distortion and noise.

So where does lack of resolution leave us? Well any signal that is below the noise floor of the noise shaper is completely lost - this is completely unlike PCM where an infinitely small signal is still encoded within the noise when using correct dithering. With DSD any signal below the noise shaper noise floor is lost for good. Now these small signals are essential for the cues that the brain uses to get the perception of sound stage depth - and depth perception is a major problem with audio - conventional high end audio is incapable of reproducing a sense of space in the same way one can perceive natural sounds. Now whilst optimising Hugo's noise shaper I noticed two things - once the noise shaper performance hit 200 dB performance (that is THD and noise being -200 dB in the audio bandwidth as measured using digital domain simulation) then it no longer got smoother. So in terms of warmth and smoothness, 200 dB is good enough. But this categorically did not apply to the perception of depth, where making further improvements improved the perception of how deep instruments were (assuming they are actually recorded with depth like a organ in a cathedral or off stage effects in Mahler 2 for example. Given the size of the FPGA and the 4e pulse array 2048FS DAC, I got the best depth I could obtain.

But with Dave, no such restriction on FPGA size applied, and I had a 20e pulse array DAC which innately has more resolution and allows smaller time constants for the integrator (so better performance). So I optimised it again, and kept on increasing the performance of the noise shaper - and the perception of depth kept on improving. After 3 months of optimising and redesigning the noise shaper I got to 360 dB performance - an extraordinary level, completely way beyond the performance of ordinary noise shapers. But what was curious was how easy it was to hear a 330 dB noise shaper against a 360 dB one - but only in terms of depth perception. My intellectual puzzle is whether this level of small signal accuracy is really needed, or whether these numbers are acting as a proxy for something else going on, perhaps within the analogue parts of the DAC - I am not sure on this point, something I will be researching. But for sure I have got the optimal performance from the noise shaper employed in Dave, and every DAC I have ever listened too shows similar behaviour.

The point I am making over this is that DSD noise shapers for DSD 64 is only capable of 120 dB performance - and that is some 10 thousand times worse than Hugo - and a trillion times worse than Dave. And every time I hear DSD I always get the same problem o perception of depth - it sounds completely flat with no real sense of depth. Now regular 16 bit red book categorically does not suffer from this problem - an infinitely small signal will be perfectly encoded in a properly dithered system - it will just be buried within the noise.

Now the second issue is timing. Now I am not talking about timing in terms of femtosecond clocks and other such nonsense - it always amuses me to see NOS DAC companies talking about femtosecond accuracy clocks when their lack of proper filtering generates hundreds of uS of timing problems on transients due to sampling reconstruction errors. What I am talking about is how accurately transients are timed against the original analogue signal in that the timing of transients is non-linear. Sometimes the transient will be at one point in time, other times delayed or advanced depending upon where the transient occurs against the sample time. In the case of PCM we have the timing errors of transients due to the lack of tap length in the FIR reconstruction filter. The mathematics is very clear cut - we need extremely long tap lengths to almost perfectly reconstruct the original timing of transients - and from listening tests I can hear a correlation between tap length and sound quality. With Dave I can still hear 100,000 taps increasing to 164,000 taps albeit I can now start to hear the law of diminishing returns. But we know for sure that increasing the tap length will mean that it would make absolutely no difference if it was sampled at 22 uS or 22 fS (assuming its a perfectly bandwidth limited signal). So red book is again limited on timing by the DAC not inherently within the format.

Unfortunately, DSD also has its timing non-linearity issues but they are different to PCM. This problem has never been talked about before, but its something I have been aware of for a long time, and its one reason I uniquely run my noise shapers at 2048FS. When a large signal transient occurs - lets say from -1 to +1 then the time delay for the signal is small as the signal gets through the integrators and OP quantizer almost immediately. But for small signals, it can't get through the quantizer, and so it takes some time for a small negative signal changing to a positive signal to work its way through the integrators. You see these effects on simulation, where the difference of a small transient to a large transient is several uS for DSD64.

Now the timing non linearity of uS is very audible and it affects the ability of the brain to perceive the starting and stopping of instruments. Indeed, the major surprise of Hugo was how well one can perceive that starting and stopping of notes - it was much better than I expected, and at the time I was perplexed where this ability was coming from. With Dave I managed to dig down into the problem, and some of the things I had done (for other reasons) had also improved the timing non-linearity. It turns out that the brain is much more sensitive that the order of 4 uS of timing errors (this number comes from the inter-aural delay resolution, its the accuracy the brain works to in measuring time from sounds hitting one ear against the other), and much smaller levels degrade the ability for the brain to perceive the starting and stopping of notes.

But timing accuracy has another important effect too - not only is it crucial to being able to perceive the starting and stopping of notes, its also used to perceive the timbre of an instrument - that is the initial transient is used by the brain to determine the timbre of an instrument and if timing of transients is non-linear, then we get compression in the perception of timbre. One of the surprising things I heard with Hugo was how easy it was to hear the starting and stopping of instruments, and how easy it was to perceive individual instruments timbre and sensation of power. And this made a profound improvement with musicality - I was enjoying music to a level I had never had before.

But the problem we have with DSD is that the timing of transients is non-linear with respect to signal level - and unlike PCM you are completely stuck as the error is on the recording and its impossible to remove. So when I hear DSD, it sounds flat in depth, and it has relatively poor ability to perceive the starting and stopping of notes (using Hugo/Dave against PCM). Acoustic guitar sounds quite pleasant, but there is a lack of focus when the string is initially struck - it sounds all unnaturally soft with an inability to properly perceive the starting and stopping. Also the timbre of the instrument is compressed, and its down to the substantial timing non-linearity with signal level.

Having emphasised the problems with delta-sigma or noise shaping you may think its better to use R2R DAC's instead. But they too have considerable timing errors too; making the timing of signals code independent is impossible. Also they have considerable low level non linearity problems too as its impossible to match the resistor values - much worse than DSD even - so again we are stuck with poor depth, perception of timing and timbre. Not only that they suffer from substantial noise floor modulation, giving a forced hard aggressive edge to them. Some listeners prefer that, and I won't argue with somebody else's taste - whatever works for you. But its not real and it not the sound I hear with live un-amplified instruments.

So to conclude; yes I agree, DSD is fundamentally flawed, and unlike PCM where the DAC is the fundamental limit, its in the format itself. And it is mostly limited by the format. Additionally, its very easy to underestimate how sensitive the brain is to extremely small errors, and these errors can have a profound effect on musicality.

Rob

Help me guys..

After listening Rob about the inherent problems in DSD that limit his musicality and that makes him flawed from the beginning, I have a doubt, I understand that any PCM file obtained from a DSD source carries this same problem implicitly, ... right? Then, the difference between these files (DSD vs PCM from DSD source) would be only the resolution..? We must then get files recorded natively in PCM to get the best out of Dave ..?

Thanks for clarifying it...

 

[SCBob]

Help me guys..

After listening Rob about the inherent problems in DSD that limit his musicality and that makes him flawed from the beginning, I have a doubt, I understand that any PCM file obtained from a DSD source carries this same problem implicitly, ... right? Then, the difference between these files (DSD vs PCM from DSD source) would be only the resolution..? We must then get files recorded natively in PCM to get the best out of Dave ..?

Thanks for clarifying it...

Will a PCM recordings at red book 44/16 sound the same as the 176/24 and other hi-res versions when played through DAVE?

 

[lovethatsound]

Will a PCM recordings at red book 44/16 sound the same as the 176/24 and other hi-res versions when played through DAVE?

No,it will sound even better.

 

[Jawed]

So, if M scaler in a Hugo TT type chassis is likely to happen then it also seems like a power pulse array in a Hugo TT type chassis (like TToby) could make for a nice pairing: 2-box M scaler that drives speakers with upto 50W per channel...

It would be amazing if you could get this 2-box system for about the same cost as DAVE. With the option to "double" the power with a second power pulse array amplifier...

Now playing: RY X - Deliverance

 

[ecwl]

Anyone try Grounding blocks?
Entreq & a few sell criminally overpriced ones. There top range one is like 9 grand for what essentially is a plain wooden box with dirt inside it.

My local dealer brought in the Nordost grounding system once and it improved his system which consisted of the Cocktail X50 streamer, Chord DAVE and an amplifier (I can’t remember whether it was Krell or Sanders or something else at the time). However, we found out later that the Cocktail X50 is not grounded so while he was making grounding cables for my router and NAS, he made a grounding wire for the X50 and the demo Nordost grounding set is long gone. But I can say the vast majority of improvements of the Nordost grounding system probably came from grounding the X50 to reduce the leakage current noise. btw, everything was plugged into the same Nordost power bar.