Help troubleshooting buffer circuit for VA's ZRLC meter

[spaceludwig]

Good day to all,

I need some help troubleshooting what seems like a simple buffer circuit for Visual Analyser's (oscilloscope freeware) ZRLC meter.

I've made both a breadboard and PCB version of the circuit - a few iterations of which can be found online - and both exhibit the same problem. Namely, even though I am succesfully capturing (see pic below) the signal the software is generating to do it's thing (i.e. putting out a 1Khz signal that is used to measure the DUT's ZRL or C after self-calibration) at the input of my sound card the software never attains the point at which it tells me to connect the DUT.



I have managed to get to this point with the built in sound card of a laptop which belongs to a friend but want to be able to make use of the software with my desktop PC via the USB sound cards I have.

Below is a pic of the breadboard version:




The circuit to the left is a voltage splitter using a 5534 to provide dual rails to the buffer circuit's op-amp (JRC2114) to its right. I am using them simply because of their output current capabilities since I have several laying about. The only departure on the breadboard circuit from that referenced above on the software's site is the use of 100uF decoupling caps at the supply pins (i've noticed without them opamps will often start oscillating on a breadboard) and as a last resort resort I added some 10uF bipolar blocking caps at the output pins as I thought the DC offset might be the source of the problems I've been having. Wrong, apparently.

Because my USB sound card operates on +4dBu line level vs the laptop's -10dBv, I am boosting the software's output signal (~2.54 VRMS) with a 70V line matching transformer hooked up backwards. Without it my signal is around -25dbfs. There is no option for me to add gain from within the PC as the levels are at 100%.

Some observations: I noticed when using the circuit with the laptops audio input device that the ZRLC meter does not work as it's supposed to (i.e. self calibration after pressing MEASURE button, then a prompt to ATTACH DUT, then readout) unless the signal is close to peaking. But I have a hard time believing that with a +4dBu line level sound card I need to go higher than the -3dbfs I am currently getting for it to function properly. The oscilloscope function of the software seems to work fine after calibration.

Anyone on this forum using VA successfully with a USB sound card? Any tips/advice on how to troubleshoot this would be massively appreciated. I'm happy to post pics/videos of any solution(s) that resolves these issues along with PCB layout/gerbers/etc for future reference.

Cheers!

 

[spaceludwig]

I've noticed the 10k input resistor to ground on the breadboard is on the wrong non-inverting input. This has been corrected.

 

[abbey road d enfer]

I don't know why, you post passed under my radar.
There are two points of discussion here:
It seems the software does not do what you wish it to do; I would suggest you contact Signore Accattatis about it. His e-mail address is on his site.
Now, regarding your set up, I see two major issues.
You power the buffer from a zero volt and two positive rails. It's a common system, these rails are usually referred to zero-volt or "ground", +V and V/2. The result is that all input and output voltages are shifted up by V/2. In particular, the inputs see signals that are referenced to zero-volt, which makes them operating outside their normal operating condition, and unable to amplify/buffer negative voltages. Since you are interested only in the AC signals, you could use coupling capacitor between the signals and the inputs; if you do that, the outputs will be centered at V/2, which the soundcard imput may not handle properly, so you would also need coupling capacitors. These capacitors should be of large enough value to present no significant LF attenuation. For that, their reactance at the lowest frequency of interest should be significantly smaller (10x) than the impedance they are connected to.
Then you mention using a transformer; I don't understand why... My understanding is that you use the USB soundcard to generate signals and the internal for measurement. VA should work with the USB soundcard or the internal soundcard, but not with both, unless you use a wrapper to aggregate both soundcards.

 

[spaceludwig]

Hello Abbey Road, thanks for taking the time to respond

I need to clarify a few things it appears I didn't explain correctly.

You power the buffer from a zero volt and two positive rails.

No, I am powering from a +/- dual rail (4.5, 9, 13.5V, i.e. depending on the number of 9V batteries I am using) power supply. I added a link to the schematic in the opening post.

My understanding is that you use the USB soundcard to generate signals and the internal for measurement.

Not quite. After having a hard time using a professional line level sound card I tried the circuit using the built-in (Realtek) consumer line-level sound card of a laptop, with moderate success. The inputs of the former will clip at around 24V p-p whereas the latter will peak around 2V p-p.

Then you mention using a transformer; I don't understand why...

I am using a transformer to step-up the output signal the software, VA, is producing as part of its ZRLC-meter measuring process. The sinewave it generates maxes out at 2.54V RMS which, in my observations, is way too low for the program to do what it needs to do. In other words, when using a consumer level sound card the software is capable of putting out a line-level signal high enough to clip it's own input, whereas a USB sound card operating at professional line-levels cannot, i.e. Consumer line-level = 1V RMS out, 0.7V RMS in; prof. line-level = 2.5V RMS Out, 12V RMS in. These are observations I made with 2 USB sound cards, a Tascam UH-7000 and a Steinberg UR22 mkII. As explained above, it seems the software will only get to the CONNECT DUT stage once the input signal reaches close to 0dbfs during calibration.

What I am trying to determine with this thread is the source of the problem: is it the circuit I built/use, or the program? I believe the first public version was written in 2007 when consumer line-level sound cards were standard so maybe never optimized for USB sound cards that operate at studio line-levels?Maybe both. Perhaps something else entirely...

Hope this clarifies my current situation, apologies if my initial post wasn't clear.

 

[abbey road d enfer]

No, I am powering from a +/- dual rail (4.5, 9, 13.5V, i.e. depending on the number of 9V batteries I am using) power supply. I added a link to the schematic in the opening post.

Then why do you use a voltage splitter?

Not quite. After having a hard time using a professional line level sound card I tried the circuit using the built-in (Realtek) consumer line-level sound card of a laptop, with moderate success. The inputs of the former will clip at around 24V p-p whereas the latter will peak around 2V p-p.

OK, I believe I got this right.

I am using a transformer to step-up the output signal the software, VA, is producing as part of its ZRLC-meter measuring process.

My comment is still valid. The soundcard does not have the oomph to drive properly a spekar transformer.

The sinewave it generates maxes out at 2.54V RMS which, in my observations, is way too low for the program to do what it needs to do.

Is it? What are the measurements that need such a high level?

In other words, when using a consumer level sound card the software is capable of putting out a line-level signal high enough to clip it's own input, whereas a USB sound card operating at professional line-levels cannot, i.e. Consumer line-level = 1V RMS out, 0.7V RMS in; prof. line-level = 2.5V RMS Out, 12V RMS in. These are observations I made with 2 USB sound cards, a Tascam UH-7000 and a Steinberg UR22 mkII. As explained above, it seems the software will only get to the CONNECT DUT stage once the input signal reaches close to 0dbfs during calibration.

Both these soundcards have gain controls that have enough range to detect very low signals. I don't see any reason why the buffer would be necessary.
Maybe it's just simply that you are trying to measure very low impedance components with too high a series resistor or too low frequency.

What I am trying to determine with this thread is the source of the problem: is it the circuit I built/use, or the program? I believe the first public version was written in 2007 when consumer line-level sound cards were standard so maybe never optimized for USB sound cards that operate at studio line-levels?Maybe both. Perhaps something else entirely...

I rather believe it's something you do or expect that isn't right.
I have installed VA for evaluation, and so far I couldn't get it to do anything of interest, since the frequency response tests are "not implemented yet".