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Korg Zero 8 circuit design question
MUFF WIGGLER Forum Index -> Music Tech DIY  
Author Korg Zero 8 circuit design question
I've been studying the chips on the korg mixers that had the horrible hiss issues. Trying to figure out if any of the grounding planes were supposed to be decoupled that were missed. I've noticed that a lot of the values for different caps were altered from the recommendations from the manufacturers. I am assuming this is related to the levels and voltages with the rest of the circuit.

However, I have also noticed that a in a few places Korg failed to decouple stuff completely. I have attached an example of this near the outputs. The AK4384 data sheet says that decoupling of the analog outs should occur before the opamps. However, based on the circuit diagrams korg either didn't do this at all or does it after the opamp with a cap that is slightly bigger than the one thats supposed to be before the opamp based on the manufacturers recommendations.

If I am confusing you... ignoring everything I have written, my question is whether decoupling before the opamp has a greater impact on noise reduction than after, like korg has done. I would assume it does but wanted to see what the pros think. I don't think the hiss issue is anything more than an oversight in the design in these as it is only apparent on the analog lines and not the digital. This might be it.

Would changing the the red box cap to a 10uF x7r require any changes in the resistor values? I don't think that it will, based on everything that I have read. I believe the same cap should be added where the red lines are as well.

I guess I need to open it up and take a look on the oscilloscope myself to see if there are any ac voltages in the grounds around these parts... Nobody seems to agree on decoupling on the internet, DAC manufacturers all insist on it and it seems as though this is a part often overlooked by circuit designers.
dont change those caps. youll get rid of noise, but youll get rid of your signal as well. the first stage is a lowpass filter up in the 80kHz range. by putting a 10uF there, its going to knock off everything above 10Hz. same for the other positions.

when the datasheet talks about decoupling, they are talking about powersupply decoupling. these are capacitors that help reduce noise on the powersupply line, which eventually makes its way into the audio path.

the first thing you should check, is what the actual SNR you are getting out of the device is, and compare this to the specifications for the device. you can do this with a decent multimeter or good soundcard. measure RMS level with a 1kHz sine tone at max volume, and then measure RMS level with no signal.

then, you can hunt down the source of the noise by repeating this experiment at the output of each opamp along the audio pathway. my guess is that there is some contribution from a noisey powersupply, and then all of those cascaded 5532 stages just add up to make the rest. i would be suprised if the output SNR was any better than 60dB. but the manual claims 93dB, so who knows?
Thanks for the info, I will give that a try in the next few days.

Those capacitors are being connected to the ground plane, how would decoupling from the ground plane get rid of the signal completely? Is 10uF just too much? That is the size in the datasheet. At worst, wouldn't it just cause a short delay in the signal?
IC45B is acting as a 3rd order sallen-key filter. the capacitor you have boxed in red helps set the cutoff frequency and resonance of that filter. by switching it from 33pF to 10uF, you are going to move the resonance down to nothing, and the cutoff frequency down by 6 orders of magnitude. no signals can get through a lowpass filter with such a low cutoff frequency.
But it is an interesting question whether 80 kHz cutoff is higher than necessary, unless you are sampling at 192 kHz.

The conventional theory has been that most people cannot hear above 20kHz. That is based on on self-reporting of perception. However, a Japanese CAT scan study found brain activity activated more on sounds with ultrasonic components than without. So it seems that (like we 'hear' low frequency with our body) it is not true that ultrasonic components can be filtered without changing what happens in our head.

But the reason for having say a 40Hz cutoff might be just technical prudence, to reduce the chance of a small stray ultrasonic signal causing some tiny aliasing when sampled, in addition to whatever existing signals. Another reason for filtering is to match the signal with the slew rate of the op-amps, so that you get filtering rather than distortion: if the mixer is noisy, it may be a sign that it is being driven at quite low levels internally to allow summing, so the slew issue would not be as strong.

What a lot of the hi-fi people say is that every component, especially cpacitor, color the sound in some way. So for program material the simpler the path the better: it might be better to have no filtering. Contrast with single voice signals (such as voices) where quite a lot of distortion can be tolerated (or such as sax, which sounds like a mosquito crossed with an elephant no matter what you do).

The other angle is that some filters have ripple: the octave or so below the nominal cut-off is not flat. So that can be a reason to have a higher cutoff frequency than you need.

So, all in all, if you know the mixed is quite noisy and is getting sampled at say 48kHz, them 80kHz cutoff may be more than you need, and you could halve it without problem, probably. But you would not expect it to make any audible difference, and if it did there is sign that something else is seriously wrong.
Guest: thanks, I'll try the scope plan and see if I can find anything. I just need to open it back up, that unit is a pain to take apart. After finding this picture on google ( ), the power supply might need a check also!

ricko: The unit no longer samples at 192 due to the lack of any support or drivers from korg. I think I've ever only once been able to get it to run at 192 using hacked drivers on a windows machine.

Korg really screwed everyone with this thing, ~$2000 paper weight from day one for those of us who bought it early on. Tons of fake promises with firmware updates and then blaming the hiss on levels. Mine has made the hiss from day one. The digital out out is clean though, that's why I started looking around the ground planes. The analog input is equally noisy as well and the only difference is the version of the DAC used there (from the same manufacturer). Everyone blamed the DAC as being a noisy design but looking at the data sheets and other info online, I don't think it's the actual DAC but the way they have it configured. If I can find the source of the problem... I was thinking about creating a board either redesigning the circuit completely or changing the current one to make a plug and play fix for everyone. Then I am going after Korg. The bastards will pay for this.
it wouldnt suprise me in the least if the powersupply was the issue.

i dont know this mixer too well. is the "digital out" from the inputs, through the ADCs, then out via USB (or other computer IO)? is there a "digital in", such that just the output pathway can be isolated? if it was poor layout or poor powersupplies, id suspect in and out to be equally crappy, unless the ADC is differential in.

best of luck on the project, sounds like it will be a help to others out there.
Thanks! It does both digital spdif in and out and connects to the computer via firewire. I've sent the spdif out to a pioneer mixer before at a club and it had zero noise while all the other methods of connecting it analog did. I've also sent the spdif to other sound cards and had no noise on the outputs there. They sold a lot of these units and cut off service completely.
have you tried sending spdif in? it might be that the processor inside that does "mixing" is bit reducing the signal by mistake.
Not in a way that I could tell if there was noise directly from the digital in. I will try that as well. Thanks
I found the service manual on google after some digging. I'll attach that here as well if it will help.
looking at the SPDIF out, it seems like its just a replica of what is being sent to the master out DAC, so the problem must be at the DACs or after.
I repaired several Zero-8 and just want to say: Get rid of it.
The main problem is the PSU, it overheat the electrolitic capacitors and start to introduce noise. The Zero-8 is noisy from the beggining but once the electrolitics in the PSU start to get dry the noise build up until it just output garbage.
I replaced the electrolitics but I read that some people just took the PSU away from the interior of the mixer to keep it cool and avoid the overheating problem...
Why would I get rid of it now. I've held on to it for 10 years. Selling it for a couple hundred dollars wouldn't change anything. The point is that korg scammed thousands of people with something that was improperly designed and released. Might be from all the brainwashing during my law school years but I think that there needs to be some justice here. I don't even care about the money but this was nothing less than fraud on korgs part. They continued selling it even after they were put on notice about the issues from the initial users and just fed everyone lies at first that a firmware update would fix it and then started to blame incorrect user levels... then suddenly when most were out of warranty they cut support without warning and discontinued it.

Guest: that's why I started with the area I posted pictures of in my original post and the grounding as everything else looks ok. The power supply doesn't look too complicated.. I wonder if I can try testing it out with the linear power supply I recently got from altitude on here for the ttsh.. just need to figure out a way to create the 48v rail.. might not even be necessary without the phantom power setting to test the other parts of the unit.
i would start with just substituting the +5V supply. there is probably a way you can disconnect just that wire at the connector and run your own 5V in there. first measure the current on that line. the schematic says 2A, but its probably less than that. you might be able to use a 7805 if its less than 1A. be careful of heat on the 7805, though. youll need to heatsink it. you can also measure the noise on the various supply lines, to see how bad it is. the DAC is run directly from the 5V supply, so noise will just shoot through. the +/-18V just runs the opamps, and the PSRR should keep that from affecting things too much, as long as the noise on them isnt horrible.

you can also set your meter to AC mode and measure the noise on each of those powersupply lines.
Thanks, I appreciate all the help! That gives me plenty to work on for now! I have an extra power-one psu laying around also with 3 amps on the 5v that I can try out. Should be easy enough. I'll open up this week and start with the easy stuff first.
you can run the +/-18V with a +/-15V supply as well, if you need to.
So I finally got a chance to take this thing apart and take a look inside. My power supply seems fine. No caps exploding or anything like that. This unit had the hiss from day one so I was pretty sure it wasn't a over heating issue or caps exploding. I haven't had a chance to test another power supply yet but I wanted to post a few findings. First it seems that they jumped a ground on the out board. My unit is a later production, so this could indicate that they were trying to find a fix for the hiss on the main boards and not the power supply.

As typical for Korg units there was plenty of no clean flux (at least I assume it is no clean flux) everywhere. The yellowing under the power board that many claim to be burn marks seems to be just flux. I cleaned it all off with an alcohol/ acetone mixture that I use. After cleaning I put the unit back together in hopes that it would get better. The hiss is still there however the sound is a bit cleaner. I had this weird comb filtering going on before that I always assumed was the room but that now seems to have gone away. I haven't put the sides back on but it seems to be running slightly cooler as well like this. There isn't much room for a bigger linear supply in there though. I might have to look around for an alternate power supply for it. Few pics attached.

[edit]: Forgot to mention, the hiss now only starts after the device is finished booting up. So when I first start it and the light show is occurring there is zero hiss but as soon as the light show is done the unit has finished booting, the hiss starts. This should rule out and any emi or other shielding issues within the case.

HI everyone,

Well I had the Korg Zero 8 now for 9 years. the problems came quit fast for me, But since im now in the diy of modular ass wel, im looking in to fixing this bad boy ass well.

I already successfully switched the touchscreen, as this was also a problem.

Last week since the heat wave in the Netherlands, the unit just spitted out ring moudulated distortion. normally after warming up the sound would calm down and be clean.

I looked at the power supply and the Elcos where all gazed out.

I took it to a small repair store, and the made the problem even worse.

I came to pick it up, and the only took out 1 cap and 1 wire curled resistor. They say it was all fried up there, but I havent got a clue, how it was before.

I ordered at digikey new caps and changed them, it fires up again, but I have a short somewhere, because my analog vue meter all the way to 500%.

How hard is it to find a short in a switched power supply and how to find it.

Is someone here maybe nice enough to find me a replacement smps or linear one.
In case of the new power supply, I will make a big 8wire speakon plug on the back what will go to the external power supply.

Any help is really helpful, because I have a gig coming up in the end of September @ the dutch modular fest 2018

OUTPUT: (1) +5V/2.0A (2) +12V/1.0A (3) +18V/0.5A (4) -18V/0.5A (5)+48V/0.1A

And two grounds for these outputs.

How do I find a proper replacement for this.

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