Xonar ST/STX mods...

Hi, I registered to the forums because this seems to be the most resourceful place for modded Xonar's. Please let me address the problem I recently encountered with my STX after 5 months of doing the Clock Upgrade (Audio-gd one @12v from computer PSU).

Suddenly after reseating the card after a motherboard replacement all the sounds coming from the system are dramatically slowed down I would say a third of the normal speed. The clarity of the sound is good, no distortion, hissing or whatsoever, just slowed down speed overall.

Thanks for your help, I am stressed and no where to find help! :)
 
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This is a clock issue. Wrong clock signal: either bad level or wrong frequency. You should come back to the standard clock approach to address this problem. Or you may contact your "upgrade" provider. At least a clock "upgrade" which it may use the 12v computer rail is only a joke.
My opinion is that you chosen however the wrong upgrade...
 
Hi, this is the upgrade I attemped, as it has been seen and reviewed in some forums.

The unit is Output 24.576MHz +8V to +20V DC hence i used the 12v rail molex adapter.

So the unit could be dead? How can I go back to a simpler resonator if I removed the small smd components across the old clock.

Cheers, thanks for your help! :)

An externally hosted image should be here but it was not working when we last tested it.
 
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Your unit it may not necessary be dead, but it may work wrong somehow. The clock signal is altered by some reasons. You may just revert the operations for installation of this "upgrade", uninstall it. I suggest you find a 24.576MHz standard oscillator (not resonator), 3,3v, and mount it so as some posts/pictures here it shows. You can power it from the 3,3v on the board.
BTW, your picture is not visible...
 
I searched this thread for this issue but didn't see anything directly related, so maybe someone here can help. I use a stock Xonar Essence STI ii with Audacity 2.1.2 under Xubuntu 16.04, also with Adobe Audition 1.5 under Win 10. In both cases, regardless of the project sample rate I set, the resulting file is of the proper format but there is zero information above about 22KHz. It's as if the Xonar ADC always operates at 44.1 or 48KHz, then the output is resampled to fit the requested format. (see attached pic of spectral analysis.) With my previous M-audio Delta 24/96, this did not happen. Can anyone shed light on this? Is it a driver issue? Hardware limitation?
 

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I've had several Xonar Essence ST cards for a while, all with some mods. I also use the HDAV H6 card for subwoofer outputs. I haven't done anything to them for a while, but got back into it recently. I've read most of this thread as well as a few others about the mods and have not seen anything about what I've found.

I believe it was Coris that said he likes the LM6172 in the I/V position and he had to replace the 2200pF caps with 10pF caps. I found that 22pF polystyrene work well, but discovered that there is something going on, presumably at the output of the DAC. I don't know what's causing it and was wondering if anyone had seen it before and was able to eliminate it. This is the output of the I/V stage:

An externally hosted image should be here but it was not working when we last tested it.


This is the output of the buffers, also with a LM6172 and 22pF polystyrene:

An externally hosted image should be here but it was not working when we last tested it.


Nice and clean, but it should be flat since nothing is playing. I've tried playing a silent track to see if it goes away when something is playing, it does not.

Given that it gets through the differential buffer unchanged it must be caused by differential current coming out of the DAC outputs just like music would.

I have seen this on three cards and in three different PCs, so it's not an isolated issue. As the capacitor value in the I/V stage increases this gets filtered out which is why most people would not have noticed it - it wasn't there. The good news is the card sounds fantastic in this configuration, the pulses are way outside of the audio band. However, they shouldn't be there and I expect they are causing some change to the sound.

It's not oscillation - the pulses wouldn't be squared off since the op amp is not clipping and I would be able to see a steady cycle.

It's not caused by power supply noise - power supply noise causes spikes like you see at the edges of the pulses in the first shot, not pulses, and most of it would be filtered out by the PSRR of the chips and the differential buffer stage as seen in the second shot.

It seems that it's something going on inside the DAC, or the DAC is being told to do this by something upstream.

Any ideas or experience with this?
 
I was wondering if there was something in the Asus control panel that might affect what I was seeing. I had previously loaded the low latency unified drivers, so I installed the drivers with the standard Xonar Essence ST Audio Center. While there's nothing there that makes a difference, it turns out that something is going on between it and Windows.

If I open the Xonar STAC and Windows playback devices windows I can change the width of the pulses. Change the sample rate in the XSTAC to 192 kHz, then double click on the Speakers Asus Xonar in the Windows window the pulse changes to about 172ns. Change to 96 kHz in the XSTAC panel, double click on the Asus speakers in the Windows window the pulses change to 330ns. Change to 44.1 kHz and the pulses change to 360ns. There is no difference in pulse width from 48 to 96 kHz. The pulse width changes do track with the frequency changes (given my measurements are not exact):

1/(48000/44100) = 0.91875
0.91875*360=330.75

This really doesn't help me, but it does show that the pulses are caused by something on the card, I'm guessing the audio processor AV100 (CMI8788). The question remains, how can I stop it?
 
I searched this thread for this issue but didn't see anything directly related, so maybe someone here can help. I use a stock Xonar Essence STI ii with Audacity 2.1.2 under Xubuntu 16.04, also with Adobe Audition 1.5 under Win 10. In both cases, regardless of the project sample rate I set, the resulting file is of the proper format but there is zero information above about 22KHz. It's as if the Xonar ADC always operates at 44.1 or 48KHz, then the output is resampled to fit the requested format. (see attached pic of spectral analysis.) With my previous M-audio Delta 24/96, this did not happen. Can anyone shed light on this? Is it a driver issue? Hardware limitation?
A kinda OT but good info for Xonar uses: I use these boards for measurements in my bench, and is fully capable to outputting more than 90kHz full expected signal with 192kHz sampling rate. "so the mods is worth" :) Happy listening 192kHz files!
And also in S-PDIF the following DAC will be output correct expected frequencies
So I can imagine a driver issue, or a software package issue
 
I was wondering if there was something in the Asus control panel that might affect what I was seeing. I had previously loaded the low latency unified drivers, so I installed the drivers with the standard Xonar Essence ST Audio Center. While there's nothing there that makes a difference, it turns out that something is going on between it and Windows.

If I open the Xonar STAC and Windows playback devices windows I can change the width of the pulses. Change the sample rate in the XSTAC to 192 kHz, then double click on the Speakers Asus Xonar in the Windows window the pulse changes to about 172ns. Change to 96 kHz in the XSTAC panel, double click on the Asus speakers in the Windows window the pulses change to 330ns. Change to 44.1 kHz and the pulses change to 360ns. There is no difference in pulse width from 48 to 96 kHz. The pulse width changes do track with the frequency changes (given my measurements are not exact):

1/(48000/44100) = 0.91875
0.91875*360=330.75

This really doesn't help me, but it does show that the pulses are caused by something on the card, I'm guessing the audio processor AV100 (CMI8788). The question remains, how can I stop it?
Seems to be something in the card. Is difficult to a low-pass bick wall filtered incoming audio stream to make a short one-sided ("half-wave") pulse
But anyway, if you're using a Windows ambient, experiment with ASIO and WASAPI drivers, only to be at "safe side"

Forgive me about OT and late posting, but since maybe someone will be reading the thread (like me today), *maybe* some more info results helpful
 
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By the way, I'm thinking to mod my Xonar, but for a bench use (a different issue/topic, in practice). The analog inputs show some potential to display very useful low noise for FFT/etc measurements if we change the original 5532 ADC driver for something more up-to-date (these inputs are already interesting as is). Same for analog output. And results far more cheaper than a AP suite.
For listening I use the S-PDIF out, since my actual DAC have far better sound than that, and I use that DAC with several other S-PDIF sources.
 
This really doesn't help me, but it does show that the pulses are caused by something on the card, I'm guessing the audio processor AV100 (CMI8788). The question remains, how can I stop it?

Looks to me like the idle channel noise generated by the modulator in the DAC. There are three discrete levels apparent +1, 0, -1. Seems perfectly normal to me and as you say its way out of the audio band so gets filtered out downstream.
 
Looks to me like the idle channel noise generated by the modulator in the DAC. There are three discrete levels apparent +1, 0, -1. Seems perfectly normal to me and as you say its way out of the audio band so gets filtered out downstream.
Interesting to know... (to avoid blaming the DAC). I have one Xonar and I don't noted this, because I don't have a digital scope. In my ye olde analog scope, only a bunch of noise appears.
 
Does anyone know what the SMD components in the green boxes are on the STX II? They aren't there on the original STX. They are directly beside the 0.1uF ceramic bypass capacitors for the ±12V pins on the buffer/differential op-amp, so perhaps they are ceramic capacitors of larger value, since the 220uF ±12V electrolytic bulk caps are so far away...? If so, does anyone know their capacitance value? I don't know what else they would be. Thanks


KBwpo1C.jpg
 
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The value of that ferrite bead it may not be very critical for the board to work quite normal. If you run the board without that ferrite in place, then it may not work properly at all. You can try to substitute the ferrite with a piece of wire, and test it for a while. It should work. As replacement you can use a whatsoever SMD ferrite bead which it may support around 400mA or so (highest resistivity at 100Mhz, for lowest DC resistivity).
How it got damaged that ferrite bead? If it was a electrical failure which caused the ferrite to broke out, then it should be found the cause of a such failure, before soldering a wire in that place...
 
So I have a power issue with my STX. I'm running an external +15 -15V DV regulated power supply connected to the board. The board is seeing both planes where they are connected... At the opamps, each channel is only seeing 15V. The output at the RCA's is weak and distorted, obviously.
Can someone suggest which points I can meter to try to find the fault?
I don't think anything blew or popped, but maybe not...