Modding the DirectStream DAC MKI

[Turbotk]

Ok got it, thanks!

I have too much gain in it currently (oops) and will adjust the feedback R down until it just stops driving the output rail to rail.

Update- happy day The model must be getting better too. I changed the gain (feedback resistor) and the bit amplitude is close to the model. I want it as high as possible but not into OA output saturation. The bits are no longer slamming the OA rail to rail, which was also causing lots of odd visible noise on the output. Could also even hear a faint hiss from the tweeter. All that fixed just by keeping the OA output out of sat…

If I had tried this 10 years ago I’d probably have been chasing this problem the next 6 months lol Previous hardship/suffering pays off, I guess…

I plan to update (improve) the output filter next. There’s still some noise to remove from the output waveform, but much better already. And it sounds even better as well. Absolutely gorgeous in fact, its playing now. Can finally now hear the concept (without the extra noise).

I’ll work on the output LPF and tune the OA input side LPF a bit as well until we get 'er just right.

Update^2 Sorry, I just cant contain myself. The sound is soo 3D spacious and airy now. Freed from the speakers; the blanket is indeed off. Rendering with ~ no distortion. Incredible. And this is still with just 1 OA/channel so far. Thank G-d!

Turbo

To me, the bit buffer is working bee-yoo-tee-ful now. This is the OA output to the LPF now. PS- I see a 5MHz bit rate:

[Serhan]

Hi Ted,
EDIT: What are the chances of having two pairs of XS4400 fail? That is my case. I removed the newer EDCOR transformers and put the originals back in! So far so good. No pops crackels or dropouts! I lost the XS4400 sound quality, but I got my DAC back! Looking forward to the next gen. DAC!
Post 2900
Best wishes

[Baldy]

That is dedication to a cause with proof of sweat equity.

[Turbotk]

Ted, I’m trying to track down a bit of static noise from the tweeters.

I see lots of bits driving when I would expect quiet. The output seems to waver with low level noise accordingly. For example, on mute. Or when another input is selected (with no source attached).

The static sound never varies. It’s there during music. It’s there with no music (source paused). It’s there with mute, or even another (non) source input selected.

At first I thought it was noise pickup… but now after investigating and some changes am strongly suspecting the noise is actually being driven.

Is this possibly because of 1 OA currently? Some scheme you have that requires OA pairs?

PS- if this static sound issue can be solved (please!) the DAC is going out on local tour asap The sound is truly amazing me… I’m ready to roll back fw or move it to 2x OA/channel.

PSS- I’m still adjusting the output LPF. It’s now a very simple 3x RC cascade. Output impedance is about 100 Ohms. A few more tweaks coming too.

Thanks, Turbo

Shows bits driven when quiet expected, and output (CH3) waving around accordingly:

[tedsmith]

“Zero” has the most bit transitions of any value in DSD. The purpose of the low pass filter is to nuke the frequencies above, say, 40 to 80k and then there isn’t any noise. There’s a tradeoff, if you use a very steep filter at, say, 50kHz you will have no noise at all, but things sound better with a shallower filter and a little (white) noise. You just need to make sure that you don’t blow anyone’s amp or speakers with too much ultrasonic noise. The SACD spec specs that the ultrasonic noise is always less than -40dBFS and that doesn’t seem to blow up too much equipment

Having two copies of unrelated noise will lower the noise by 3dB, having four opamps lowers the uncorrelated noise by 6dB. This uncorrelated noise isn’t the noise from the digital bits, but instead is noise from the opamps, their differing power, etc.

[tedsmith]

And yes, as I’ve been saying, with recent releases of the software using only one opamp will cause a little noise. If you want to use only one opamp you need to go back a few releases (I’ve mentioned the releases a few times above.)

[Turbotk]

Ok will do.

I hate to ask but is there a link handy for where to get it and how to do it?

Thanks, Turbo

[tedsmith]

Under the support menu above are “Downloads” and also “How To”. Under them you can find most of the DS software releases and notes on how to do a software change.

[Turbotk]

Thank you for your kindness and patience

BTW my unit currently has Sunlight 5.0.0.

Oops, this got deleted somehow. Update- I put on Snowmass 3.06 and the issue is solved. No more static noise from the tweeter; the noise was being driven. Thanks for your help. Now can proceed with the development

It’s hard to do major circuit changes like this (especially with 5 MHz digital right next to analog) and know where the noise is coming from. But now I can’t hear any unwanted noise anymore with my ear next to the tweeter… Hurray!

Turbo

[tedsmith]

Don’t forget to try two opamps and go back to Sunlight at some point, it does sound much better than Snowmass on most systems.

[Turbotk]

YES I was just about to post that… It seems the sound took an obvious step back with Snowmass. It flattened a bit in space.

[Turbotk]

BTW I recalled another passive LPF that we might look into here. It might simplify things a bit while achieving a deep rejection null around the DSD fundamental. I’m talking about a passive RC feedback filter. As in, how Naim did their preamp input buffer/filter after CD came out in the 80’s- the mysterious “time aligned” buffer/filter. Turns out its a linear phase passive feedback filter with a deep null at 44K1. But it uses large value R so its also buffered out. For reference the image is below (the R/C in the middle). Wonder if we could maybe do similar with low value R’s (low Z).

We could look to scale something similar basically; I do have a lot of experience with Naim amps and preamps by now. Or, I have some filter design sw that they may grant a trial license again. You can add a zero to the function and that can do similar I think. I don’t know, maybe it wouldn’t work well without a buffer/isolator to drive the feedback. Maybe it cant be done passively after all.

Waiting for some parts from Mouser (tomorrow) so we can nail down rd 1 here with 2 OA/channel and take it on tour to the audio gang fun fun

Turbo

Naim time aligned buffer:

Update- couldn’t resist a few minutes with the sw. Look at whats possible with just 1 inductor and 3 small caps for example Now that’s a null at DSD2 freq (red circle)!

[lufubu]

You really turbo to the mod title

[ccyun051]

Does upgrading the transformers in the analog board void the warranty?

[tedsmith]

PS Audio doesn’t care about mods as long as they didn’t cause the warranty issue, and don’t get in the way of fixing the warranty issue. If a mod is improperly installed, they probably won’t help fix it.

[ccyun051]

Thanks Ted! I will ask Wayne at WctAudio to install the transformers for me.

[Turbotk]

Ted, a couple of fw question please if I may:
-Is it possible to convert PCM->DSD without upsampling the PCM first?
-Might it be possible to get a test fw that brings the sonic goodness of Sunlight w/o the staticy tweeter noise (on 1 OA/chan)?

I plan to try straight DSD file over USB and see how it compares to the DAC converted PCM->DSD. My usual streaming setup is USB->Gaia DDC->I2S DAC

Updates: I found the sound lost the amazing spatiality and a decrease in the super high resolution, relatively when I gave ea OA (2 OA/chan finally) its own primary coil. Bad idea. Joined the coils together again (in parallel) and the amazing spatiality is back I attribute this to an increased noise floor, which covers up these low level spatial cues and information. Kind of like what jitter does (YMMV)

This amazing spatial quality is turning out to be the prime characteristic that sets this apart now. On top of the ~ perfect rendering and such. And we cannot go back, once heard. Anything less is disappointing at best (and immediately noticeable).

The room pressurizing low bass is also enjoyable now

Speaking of, I was also planning to roll back briefly to 1 OA/chan again, as that’s when it got amazing last night after updating to the latest 4th order simple passive filter, before duplicating another pair of OA. Even with Snomass fw. But no, I can even hear it as I type, there’s a whole lot of 3D going on and high resolution even around the corner where its playing. The music becomes freed from the speaker in a way I have not heard in my system

However to be complete I should probably follow up on that today just to make sure. Then button 'er up in best config with the shielding and get ready for tour.

Another note, I was really churning on trying to figure out why the bits amplitude was slightly higher coming out of R4 OA… Where was the mistake? Churn, churn…

Eventually I realized the bits amplitude coming in had to be slightly higher somehow… hmm… Ahhhh… the nearby R3 OA, unused except left with input termination network… One of those input R was not completely soldered (very hard to see); thus that bit network was not properly end terminated. Fixed that and joila, R4 OA is same amplitude as the others and looking great IMO (see below)

PS- I’ve found I like the “Out There” mix on Tidal quite a bit. Maybe we have a strange kinship lol Maybe some others might like it as well.

Turbo

OA R4 bits output; tweaked the input filtering a bit more as well:

[tedsmith]

@Turbotk

Ted, a couple of fw question please if I may:
-Is it possible to convert PCM->DSD without upsampling the PCM first?
-Might it be possible to get a test fw that brings the sonic goodness of Sunlight w/o the staticy tweeter noise (on 1 OA/chan)?

What problem are you trying to solve by not upsampling? Upsampling is the way to get clean antialiasing (well really a clean reconstruction filter) of PCM. Without upsampling you get gobs of aliasing or you need an impossibly steep analog reconstruction filter for the PCM.

A part of the goodness of all of the software after Snowmass is the feature that requires opamps in pairs. In general, you can’t pick and choose features from one DS software release and other features from another DS software release. The best-case scenario is that even if the software for those features is compatible (tho the big reason to leave one software release is to do a better job on the next one) we need to build 20 (or more) compiles and listen thru them for the good sounding ones vs. the bad sounding ones.

[Turbotk]

Well I was just thinking about issues with upsampling… Not a benign process. And all upsamplers are not equal. Chord for example have spent a career developing their long tap filters.

And yeah, I guess it also removes the beats that are apparent in the 44K1 or 48K data.

Anyway was just wondering out loud

Turbo

[tedsmith]

Bad upsampling isn’t benign. Good upsampling is much more benign. Upsampling is necessary for accurate reproduction of PCM (unless it’s sample rate already really high so you can afford rolling things off drastically. There are a lot of bad upsampling implementations out there, tho things are getting better. I (like Rob) have spent a lot of time implementing better upsampling than most. FPGAs (which Rob also use) allow more accurate real-time math to be done.