Amateur OB speaker builder and his ARC based digital system

[Ittaku]

People may have noticed that I sold my P700 CD transport and wondered what I use to play CDs now. Virtually all my serious playback now is just done via digital files on Roon, including ripping any CDs and then upsampling them. But sometimes I may just want to play a CD for someone else without going to the effort of ripping and upsampling. I've come up with a solution and now I use this super-high quality CD transport known as a "no-name DVD writer" that cost me $30. The data is then piped through my PC and run through the same DSP convolution filter for 44kHz that I use on Roon. The quality of the CD transport makes absolutely no difference when I do this since the data is all recreated from scratch within the PC provided I can be assured the reader is bit-perfect. The output uses 24/44 on the optical output of the PC's own sound card, thereby avoiding any chance of the PC electrical noise polluting the DAC, and since the MSB DACs buffer and reclock everything internally then source jitter is irrelevant. Of course it sounds sensational.

 

Edited July 3, 2020 by Ittaku

[Steve M]

Haha Con, you must be the only person on the Planet feeding a $30K(?) MSB uber dac with a $30 DVD writer!  Would have have to agree with you though that if it works it works ...there are a lot of myths in this hobby of ours if one is really honest about it.  Cheers, Steve.

[Ittaku]

Did some interesting experiments with SACD 11289kHz .dsf audio and converting it to PCM audio at ultra high res. The most interesting part was there only appears to be noise above 22.05kHz, much like CD audio with no oversampling. Here's one that was resampled to 176.4kHz. You can see the abrupt cut-off of obvious music data at 22.05kHz. I'm wondering if it's the resampling software responsible because the abrupt cut off at 22.05 seems to happen no matter what the target sample rate is. There sure is a heck of a lot of ultrasonic noise. It seems to me you need a standalone filter at 22.05kHz regardless of what sample rate you choose for the final. I suspect it's just the software since I've seen other people rip SACDs to 88kHz and there isn't this noise-like spectrum. The other possibility is the SACD was a faked CD to SACD conversion. Need to do more investigating.

 

Edited July 11, 2020 by Ittaku

[Ittaku]

Okay, it must be the sox patch I was using that's responsible that simply targets for a 44kHz final. Here's an SACD rip to 88kHz someone else provided. That makes much more sense than what I managed to get. Time to look more closely at the code itself.

Edited July 11, 2020 by Ittaku

[Ittaku]

It turns out my analysis software can read DSD files directly and I get exactly the same result. Looks to me like the SACD is just repackaged CD data. That is a hell of a lot of ultrasonic noise. I downloaded a 8xDSD sample from nativeDSD and got the same sort of result as well. I wonder what's going on, and why the rip example above isn't like that.

 

[Ittaku]

This is just a bit of a journey of my own to understand SACD better since I pretty much missed out on it entirely.

 

So this makes it a bit clearer. Looking at a 1xDSD file and zooming into the range to 50kHz I can see there is music information up to 50k, and possibly beyond, but there is progressively more switching noise that drowns it out. I can see why many filters default to 30kHz and optionally can be set up to 50. Looking at this now I suspect that any audible advantage of SACD over CD had nothing to do with using less bits and more sample rate. By moving the antialiasing filter to 30kHz from 22.05kHz it means the filter would have no effect on phase and amplitude in the audible range up to 20kHz without extensive processing power.

 

[lemarquis]

29 minutes ago, Ittaku said:

This is just a bit of a journey of my own to understand SACD better since I pretty much missed out on it entirely.

 

Looking at this now I suspect that any audible advantage of SACD over CD had nothing to do with using less bits and more sample rate. 

Interesting. Of course there is the additional issue that a lot of SACD (DSD) content is upsampled / converted from PCM anyway, as has been noted, so I look forward to seeing what your investigations uncover in terms of the SQ benefits (or otherwise).

Edited July 12, 2020 by lemarquis

[Ittaku]

Here's the same file (all from http://www.2l.no/hires/) at 2xDSD (DSD128). Basically double the sample rate, double where the switching noise begins. If your aim is to get more ultrasonic frequencies back during your playback, you will benefit from higher DSD rates. If, on the other hand, it's to avoid the audible effects of the filter, then single rate DSD is enough. For the record, I cannot tell these two files apart. They also offer a 4xDSD variant of the same track, and I still can't hear any difference. No need to do any ABX testing as I can't tell them apart even unblinded. Looks to me like high DSD rates are just as useless as PCM files going above 88kHz sample rates.

 

 

I might add something that people might find amusing - DSD or SACD is basically the digitally recorded equivalent of class D amplification in the way it works. So if you're a believer that SACD sounds more analogue than CD, but are opposed to class D amplification, you might want to reconsider your position, especially with the latest class D amplifiers.

[Ittaku]

Well I'm an Audio Research fan but this just isn't right. I heard the Ref6 preamp was coming out with an SE edition, with upgraded caps and wiring so having auditioned the 6 I figured I will need to audition the SE when it comes out. The 6 was about $22K last time I checked. I know our dollar has fallen a bit so I can see why the retail on the 6 went up to $25K. But I'm not really sure where just upgrading a few components for the SE model deserves a $7K or almost 30% price premium .

 

[Ittaku]

I had a few people PM me asking about creating your own 12V trigger that I mentioned earlier. Any 12V DC power supply will do, as the requirement is absolutely minuscule - the specification dictates only 12mA is required. Using more achieves nothing, it will just waste power and potentially inject more powerline noise from your power supply back into the mains.

 

The main thing you need are adapters that convert the regular barrel adapter to a mono 3.5mm jack. These are surprisingly hard to come by.

See: 3.5mm Male to 5.5mm x 2.1mm Female DC Power Adapter:

https://www.amazon.com/gp/product/B077TRBRSB/

 

This is the PSU I use with it:

https://www.amazon.com.au/gp/product/B07D14XLLX/

 

 

Here's another spare 12V SMPS PSU I can use with these adapters.

 

No, attaching this did not have any demonstrable audio effect on my system as it just drives relays in my power amplifiers in a way that is isolated enough from everything else that it doesn't affect it. Your mileage may vary.

 

I have the little SMPS power supply in a separate power board (the SPI I mentioned earlier) and when I power on this power board, it triggers the power amplifiers to turn on, which are plugged directly into the wall for their own power instead of running through a power board.

[Ittaku]

Got these little beasties in the mail today! I was surprised to see them come in these neat little boxes. They're really well made, look very neat and are very robust but compact. As my amp has extra taps I also got the stoppers for the banana outlets to cap the unused ones, and the banana inlets on my speaker terminals since I use spades there. It was very interesting to see how tight a fit they were on the power amp (ARC) but were pretty loose on the speaker terminals (Cardas.) Anyway I feel more comfortable having them all covered now.

 

More supplemental hardware coming soon. Watch this space.

[lemarquis]

What is the logic behind these? Did you notice an appreciable improvement?

[Ittaku]

7 minutes ago, lemarquis said:

What is the logic behind these? Did you notice an appreciable improvement?

Of course not. They couldn't possibly have any audible effect. They're to protect them from dust (and corrosion in the case of my speaker terminals which are tellurium copper and prone to surface oxidation.)

[lemarquis]

9 minutes ago, Ittaku said:

Of course not. They couldn't possibly have any audible effect. They're to protect them from dust (and corrosion in the case of my speaker terminals which are tellurium copper and prone to surface oxidation.)

They look fine.

[davewantsmoore]

On 12/07/2020 at 3:53 PM, Ittaku said:

Looks to me like high DSD rates are just as useless as PCM files going above 88kHz sample rates.

 

The big idea for high rates of PCM, and DSD is around not converting the audio in a way which damages it.    MQA has a similar philosophy.

 

When due care is used (not in all devices, that is for sure), I don't think there is a big problem.... but, there's obviusly a large segment of the market who are happy to "ignore reality" (to sell you the content and second and third time at a "higher res").

[Ittaku]

I started getting some crackling out of the left channel which sounded like a bad valve so I pulled out the valve tester I'd been using to see if I could diagnose where the problem was. Putting the preamp on mute the noise would go away so I could tell it wasn't in the power amp. I found one preamp valve's "RMS noise values" would fluctuate a lot so I removed it from the preamp, but it was one of the ones going to the subwoofer, and I'd known there was an issue somewhere there for a while since very occasionally I'd get spontaneous noise out of the subwoofer which I'd largely ignored. Replacing this valve fixed the spontaneous noise issue, but did not alter the crackling out of the left channel. So, even though all the valves measured identically, something was still responsible for a crackling noise. I tried cleaning the contacts of the preamp valves and this made no difference, but I wouldn't have expected it to since it would have been noisy from the moment I first put it in rather than months later. So I turned to the "listen" facility of the preamp tester and found it incredibly useful. I went through my dozen or so valves that I could use for the preamp and found the one in the preamp that had a noisy baseline noise that would come and go. This turned out to the be the culprit for the crackling noise in the left channel and replacing it fixed it.

 

Then I went further and graded my valves according to their microphony level with a simple test of tapping on the preamp valve tester. For valves that measure identically, there were wild differences in their microphony levels, and being dual triodes that have two sections, I found that microphony varied wildly between the two sections as well. So after grading my valves according to microphony levels, I put the least microphonic ones back into the preamp. It's fair to say this did make an audible difference to the sound, with that inimitable "black background" quality being magnified, and indeed the speakers with my ear up against them are now almost dead quiet. I went further with my testing after this, as I'd left a valve in my collection I'd texta marked as "bad" a long time ago and found it had insane levels of microphony. This led to an interesting experiment to confirm whether putting dampers on the valve are actually effective or not. I've created a separate thread and video describing my experiences, but the conclusion is a resounding YES.

 

 

Edited July 23, 2020 by Ittaku

[Ittaku]

Look what I've got to test out!

 

 

As many of you will be aware, I'm not big on power related products for hi-fi so this should come as a surprise. However, my regular complaint with power products is that there is very little you can achieve with passive filtration - and that is unlikely to translate into substantial noise changes at the hifi end and may be detrimental, and power cables largely can't do anything at all. I have said in the past that active regeneration is likely the only power product which has a chance of making a difference, and will be of more benefit the cheaper the power supplies are in the hi-fi equipment. The downside is they're so expensive that you can only afford them if you can also afford very expensive electronics in the first place, and they'll benefit the least from regeneration.

 

However, power regeneration offers a few other benefits beside the filtration aspects - it has ultra low output impedance so provided it can serve enough current for your equipment, it should actually be beneficial in dynamic demand situations. It can strictly regulate the voltage  which is of most benefit to hardware that doesn't have voltage regulation - every single component I have is voltage regulated though, including the power amps (very few power amps actually are.) But regulating the voltage has another advantage - the power consumption of your devices and the longevity of the components is substantially affected by high voltages, and passive power supply components in electronics can't do anything about low voltages. The voltage in my area is almost always high, usually around 247V, but I've seen it as high as 257, and ironically today is only 234, so quite wide swings. Another advantage is creating a pure sine wave. This may not sound that important, but the fact is that power supply capacitors charge up on the peak of the waveform only, and power supplies are designed with a particular frequency in mind - they're best at utilising that frequency and then filtering it out. A waveform which isn't a sine wave is made up of multiple other frequencies as well so they're potentially harmful. Funnily enough, if you want to "beef up" a power supply, something closer to a square wave would actually be better because once rectified it's already almost DC and would have less ripple current. However as I said earlier, the power supplies in electronics are built around the assumption the power is of a particular frequency and waveform. The PS audio regenerators have a "multiwave" function which makes the sine wave "fatter", so whilst not quite getting to a square wave, they can perform this function optionally. They can also be configured either for tighter regulation or filtration, and have their phase optimised (I have no idea how this works!)

 

More to come.

[Ittaku]

You can see my system uses almost 900W. This is the same when idle and almost all loudness levels except for ear-splitting loud, thanks to the power amps being heavily class A biased. The subwoofer is the only thing that really dynamically drain power.

 

 

 

Cute power icons on the front you can turn on and off by themselves which correspond to the zones on the back. Best part for me is you can program them to delay startup or shutdown a configurable amount, which is ideal for switching the power amps and subwoofer on and off separately from the preamp and avoid potentially damaging noise.

 

This is set to high regulation and 230V out, not high filtration, but it's obviously enough to filter out the level of line noise I currently have:

 

Strangely today I seem to have low voltage rather than high, and it's looking a long way off being a sine wave. The other devices I use to confirm voltage also confirm that is the wall voltage, but I don't have anything that can show me the waveform to confirm this is real.

 

Pure sine wave mode:

 

Mutliwave mode set to 6%, you can see the "fatter" sine wave:

 

 

[buddyev]

Nice one, Con. And it's a close visual match with your ARCs.

You do realise that you're going to have to buy another to balance things out on the rack.

[Ittaku]

So one last picture. This is the one that concerned me slightly. The measured line EMI from the P20 output is unchanged in voltage from what I get when I plug it straight into the wall. However this line noise meter also amplifies any non-50Hz frequencies through the little speaker at the front so you can qualitatively assess what the noise sounds like, and the audible noise at least is significantly quieter. This weirded me out, but I guess it means the other EMI noise is all ultrasonic frequencies and beyond so perhaps doesn't matter?

 

Compare it to this active filtering Thor on another outlet, which is now showing 247V (showing how much the voltage fluctuates) and how much lower the EMI noise is. And yet the sound coming out of the EMI gauge was still louder than that from the P20. Chain them perhaps? Or maybe those frequencies just don't matter? I certainly didn't hear any sound improvement with or without this board.

Edited July 24, 2020 by Ittaku

[Ittaku]

15 minutes ago, buddyev said:

Nice one, Con. And it's a close visual match with your ARCs.

You do realise that you're going to have to buy another to balance things out on the rack.

Oh God no... This thing's supposed to have enough current for twice as much as what I'm running of it, so I hope not.

[andyr]

40 minutes ago, Ittaku said:

You can see my system uses almost 900W. This is the same when idle and almost all loudness levels except for ear-splitting loud, thanks to the power amps being heavily class A biased. The subwoofer is the only thing that really dynamically drain power.

 

 

 

Very cool, Con. 

 

You say you are ouputting a fraction under 230v - due to a lower-than-normal mains voltage today.  So when the input mains voltage goes back to 247v, say ... does this mean the output voltage will rise - or is it locked at 230v?

 

40 minutes ago, Ittaku said:

Strangely today I seem to have low voltage rather than high, and it's looking a long way off being a sine wave. The other devices I use to confirm voltage also confirm that is the wall voltage, but I don't have anything that can show me the waveform to confirm this is real.

 

 

That, surely is a sine wave which is clipping?

 

40 minutes ago, Ittaku said:

 

Mutliwave mode set to 6%, you can see the "fatter" sine wave:

 

 

Fascinating.  What benefit does "multiwave mode" give you and how do you decided whether you should select 6% ... or 10%?

 

Andy

 

 

[Ittaku]

Okay so you're all probably waiting with bated breath to hear what my thoughts are on what kind of audible improvement - if any - it provides. I needed to consolidate my thoughts before I can word exactly what's going on and to accept what I'm hearing, but it's a definite yes. It's hard to describe exactly what it does to the sound but it reminds me of the change I had when I shifted from a stereo power amp to more powerful monoblocks. It's not so much about that "blackness" everyone is chasing, so much as the dynamics of presentation. It seems to have breathed life into the music and made it render in physical space more effortlessly and have a more rock solid presentation in the stage, along with just more in-the-room realism, and more dissociation from the speakers. People have described passive power filtration devices as "choking" the power and dynamics, but that was never my experience, even with the top Isotek device - I simply didn't hear them do anything at all. That may have something to do with having such heavily class A biased amps that current draw is absolutely constant under virtually all circumstances. However, the reason I bring it up is this regenerator appears to have done exactly the opposite, breathing more life and effortless dynamics into the music. I've tried comparing the pure sine wave to the multiwave and I can't really hear any obvious difference so I don't know what's a win. Perhaps if I squint hard I think it sounds cleaner in the sine wave mode but I'd have trouble telling them apart.

 

It's a much bigger improvement than I got from auditioning the Ref160M amplifiers and comparing them to my 250SEs. It's a clear win and it definitely stays!

[Ittaku]

8 minutes ago, andyr said:

You say you are ouputting a fraction under 230v - due to a lower-than-normal mains voltage today.  So when the input mains voltage goes back to 247v, say ... does this mean the output voltage will rise - or is it locked at 230v?

That's the thing, I can choose whether to prioritise regulation or noise filtration. I have it set to regulation which means it will keep it at 230V and can handle anything from 200-285V

 

8 minutes ago, andyr said:

That, surely is a sine wave which is clipping?

Yep that's exactly what allegedly is coming out of the wall here. I've only ever seen one website try to measure this device and they confirmed this is a common waveform!

 

8 minutes ago, andyr said:

Fascinating.  What benefit does "multiwave mode" give you and how do you decided whether you should select 6% ... or 10%?

I mentioned it earlier when I was talking about square waves. It means there's less ripple current overall, but does change the frequency of any powerline noise that's transmitted into a group of frequencies instead of just one.

Edited July 24, 2020 by Ittaku

[aussievintage]

2 hours ago, Ittaku said:

Yep that's exactly what allegedly is coming out of the wall here. I've only ever seen one website try to measure this device and they confirmed this is a common waveform!

 

This caught my eye.  I have looked at mains waveforms many times, and each time it was a nicely shaped sinewave.  Never any clipping.