FFT RMS or Peak?

Hi,
I've got a sine wave signal that oscillates about the zero axis in the negitave and positive. It is measuring wave height about the static water level in an water wave tank. I am calculating the FFT but the amplitude is lower than the wave height in the time domain. Does this mean that I'm calculating it wrong? I don't know wheather I should be using the FFT Peak or RMS? I assumed that the FFT amplitude should give me the same amplitude as the time domain.
Any advice would be greatly appreciated.
Thank you,
Donners
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Donners,
It can get complicated but I suspect that what you are seeing is a result of the frequency of the waves not matching exactly the frequency bins of the FFT. For example suppose that df = 1 Hz. Then each element of the FFT array represents the energy in a "bin" one hertz wide centered on integer frequencies. For a wave with most of its energy at 7.3 Hz the FFT will likely have non-zero values for both the 7 and 8 Hz bins. In real systems with noise and slight frequency variations it is not uncomon for the data to be spread over 10 or 20 bins. To get the fundamental amplitude of the signal you would need to combine the energy in all those bins. Since the FFT produces a complex output, the phase component also needs to be considered. Non-sinusoidal waveshapes mean that some of the energy is in harmonics. A narrow square pulse can have a quite large amplitude yet still have no large spectral components.
The short answer is that generally there is no easy way to get precise correlation between the peak amplitude in the time domain and the amplitude of the spectral components in the freqeuency domain.
Lynn

Similar Messages

SVT 3.1 FFT Spectrum Magnitude VI output is RMS or Peak (db OFF)

Please help with my SVT problem:
1. With db OFF, what magnitude EU value does the SVT 3.1 FFT Spectrum VI output? More to the point, is the EU an rms value or peak value? I thought it was rms but now I'm not sure.
2. When I double integrate acceleration values to get displacement then feed the displacement waveform to this FFT Spectrum VI, I seem to get high displacement values compared to other portable analyzers in our shop. If the output of the FFT Spectrum VI is for example, mils rms (single sided) I have to convert this to peak-to-peak values. So I multiply the each spectrum value by 1.414 (to get peak) then by 2 for peak-to-peak. Is there a more accurate way to get peak-to-peak displacement values from an acceleration waveform?
thanks,
Glen

Hi Glen,
There are several FFT-based VIs in LabVIEW and the output units of each function is different.
The following table will give you a better idea of what each function does. In your case you might be better off using the FFT VI that comes with the Signal Processing toolkit since you are looking for Volts output instead of the Vrms given by the FFT Spectrum functions.
If you have doubts about the units returned on a specific function I recommend using the Context Help in LabVIEW. As shown on the image below this gives you more information about the function and the variables used/returned by it. You can open Context Help by going to Help>>Context Help in LabVIEW.
Hope this helps. Regards,
Tica T
Applications Engineer
Message Edited by Tica T on 04-04-2006 10:38 AM
Attachments:
units.JPG ‏50 KB
functions.JPG ‏85 KB

Fft rms vs pk scaling

I have a program that is taking waveform data from an accelerometer. I eventually want the units in g's pk, but in the fft itself I get a cleaner signal by doing the RMS. It is the same thing taking the g's RMS and then converting it to g's pk without the initial averaging?
Solved!
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KJohnson42, the NI fft vi I am using is SVFA FFT Spectrum (Mag-Phase).vi.
Let me walk though the entire process. I have an accelerometer which has a 100 mv/g output. I am using the rio waveform acquistion vi's as you can see in the picture of my code in the first post. I have the input scaled in the channel info there. That waveform is then passed into the fft vi where I use the magnitude output to get graph that has vibration magnitude on the "y" axis and frequency on the "x" axis. In vibration you have three types of measurement but for the most part only two of them are used, velocity (typically inches/sec) and acceleration (g's).
Now, you can look at those measurements in three methods, peak, peak to peak, and RMS. In the attached picture is Bently Nevada's explination of those measurements:
So my question is what exactly is coming out of the SVFA FFT Spectrium vi, pk, pk-pk or rms? The SVFA Unit Conversion vi gives me the choice of pk, pk-pk or rms values to convert the units to.
Attachments:
New Bitmap Image (2).jpg ‏329 KB

RMS - Filtros - Energias de los datos por DIADEM

Holaaa, queria saber si en DIADEM, se puede hacer procesamiento de senales, sacarle RMS, ENERGIA, TRANSFORMADA DE FURIER, FILTROS, me pueden enviar unos ejemplos por favor.
Muchas gracias.

Hello Gared,
My Spanish is a little rusty, but here are some examples of analysis reports including FFT, RMS, digital filters, etc.:
DIAdem example gallery
A detailed list of all the math functions is DIAdem can be found here:
DIAdem analysis functions
I hope this helps answer your questions,
Otmar
Otmar D. Foehner
Business Development Manager
DIAdem and Test Data Management
National Instruments
Austin, TX - USA
"For an optimist the glass is half full, for a pessimist it's half empty, and for an engineer is twice bigger than necessary."

RMS terminology and power comparison standards

I am doing my research in order to find the best of the best 2.1 speaker system for my PC. I already have an AUDIGY 2 SE soundcard and I would like to support it with an excellent sound output set. While comparing different speaker systems I have came up with the following two best of the best products: Creative I-Trigue L3800 (Price 99GBP) and Logitech Z-2300 (Price 91GBP). Even if these are the best of the best it seems that they have a serious difference of power between them. Here is the RMS speaker power specifications:
1.Creative I-Trigue L3800: 9 Watts RMS per channel (2 channels) 30 Watts RMS subwoofer 2.Logitech Z-2300: Total RMS power: 200 watts RMS Satellites: 80 watts RMS (40 watts x 2)
Subwoofer: 20 watts RMS Total peak power: 400 watts
My conclusion from the above data is that Logitech Speakers are 4 times more powerful from those of Creative (200 watts compared to 48 watts). Is my above conclusion correct? What else do I have to compare in order to buy the best system?

Clearly the " RMS " is the standard.
They do clarify their goofy ratings
in small print on some pages.
Those LT's you mention are great deal.
Very similar but in a revered brand - KLIPSCH -
a similar system is available for about $30. more.
That 'THX' certification is supposedly a big deal.
Big plus with the L3800 is remote controls functions
on Zen Vision M.
good luck

Conditionally capturing certain data while continuously acquiring data

In case my question has totally confused you, let me describe the situation, a general one which I'm sure some of you have already tackled.
Suppose you're acquiring data from several channels (let's say two waveforms). And you only want to capture their amplitudes (Y1, Y2,etc.) when either a button is pushed (i.e., things are steady state, now grab that data) or when something else has changed a certain amount (say their frequency has changed 2 Hz).
When I say "capture", I simply mean grabbing the data into a separate array of X,Y values for later processing or on-line plotting while still acquiring data.
Hopefully, this makes sense. Would greatly appreciate any thoughts on this, as it's a
situation that I seem to run into often.
All the best, Hunter

Your use of the terms "Waveforms" and "amplitude" implies a measurement of many samples into one number (for each channel).
For example, if your signal was pure 100 Hz, 1.000 Volts peak, and you captured a 0.1 second block of that signal, then you would have 10 cycles in the capture buffer, varying from +1V to -1V. You can use the RMS function to turn that block of samples into a number of 0.707, meaning the RMS value is 0.707V (= 1.0 V peak).
The basic philosophy would be:
1... Start a continuous acquisition. By specifying NUMBER oF SAMPLES = 0 in the AI-START vi, you will acquire data forever (until the buffer overflows, or until stopped).
2... Periodically (based on a timer), you ask the AI READ function to read zero samples. This does not remove data, but it does tell you how many samples are in the buffer.
3... You immediately read that many samples out of the buffer, and convert from volts to engineering units (if appropriate).
4... If you are measuring their frequency, you need to perform an FFT now. The highest peak in the magnitude of the FFT output is the dominant frequency in the signal. (Freq. of peak = Index of peak * Sample Rate / NPoints in block.
5... If you are doing FFTs, you get a big speed advantage if your block size is a power of two (128, 256, 512, 1024, etc.). If so, then replace step 3 with *-IF-* backlog >= N (power of 2) then read N points from buffer.
6... Apply your criteria - *-IF-* the button has been pressed, *-OR-* if the frequency of channel 2 is different, then compute the RMS or peak values and append to your final results array. Otherwise do nothing.
If you want to detect 2-Hz changes, you need to know the rules of how FFTs work - the frequency resolution is the BIN WIDTH, which equals the sample rate divided by the number of points in the block. If you have a 1000 Hz sample rate and a 1024 point block, your resolution is 1000 / 1024 or just less than 1 Hz. But if you have a 1000 Hz sample rate and a 32 point block, your resolution is 1000 / 32 or about 30 Hz.
Steve Bird
Culverson Software - Elegant software that is a pleasure to use.
Culverson.com
Blog for (mostly LabVIEW) programmers: Tips And Tricks

Auswertung von Schalldruckpegeln mit DIAdem

Hallo
Mein Problem ist, dass ich ein Schalldruckpegel messen und das Ergebnis mit DIAdem auswerten möchte.Ich habe im Anhang eine Datei in dem mein Aufgab zusehen ist. Ich bin soweit, dass ich den Schalldruckpegel schon ausgebe (siehe Block dB). Nun will ich aber noch das Signal aus dem Messgerät in einem Schalldruckpegel/ Frequenz Diagramm ausgeben. Dazu ist der zweite Strang (Packen, FFT, Formel Terz_Oktav und Oszi).
Meine Frage ist, was muss ich im FFT-Block, Im Formel-Block, im Terz_Oktav_Analy und im Oszi eingeben?
Bis jetzt habe ich im FFT-Block einen Peak und im Formel-Block die angebene Formel angegeben.Ist das so richtig?
Könnte mir wer weiterhelfen? Ich bin Student und kenne mich kaum aus mit dem Programm.
Ein nettes dankeschön schon mal im voraus.
Attachments:
Schalldruckpegelmessung.JPG ‏41 KB

Hallo Sebastian,
Der Schalldruckpegel in dB basiert auf dem RMS-Wert eines gemessenen Signals. Normalerweise bekommt man die passenden FFT-Ergebnisse deshalb auch dann, wenn man in der FFT RMS einstellt. Leider kann man an dem Bild nicht erkennen, was sonst noch eingestellt ist. Ich fürcht aber, dass hier grundsätzlich etwas nicht funktionieren kann.
In dem Schaltplan wird ein Signal erfasst und nach zwei Formeln in dB ausgegeben. Ich vermute, dass hier bereits RMS-Werte aus dem Gerät kommen und die Abtastrate dürfte relativ niedrig sein. Die dB-Werte, die in der numerischen Anzeige ankommen liegen bei Schallmessungen typischerweise irgendwo bei 40-120 dB, je nach dem was man da so misst. Die FFT liefert bei solchen Eingangswerten unsinnige Werte. Bei Schallmessungen sollten die Eingangswerte für die FFT Schalldrücke in Pascal sein und mit Abtastraten wie 44,100 oder 50 kHz erfasst werden. Das Beispiel zeigt also entweder unsinnig dB-Werte an oder berechnet eine unsinnige FFT.
Viele Schallmessgeräte haben zwei analoge Ausgänge - DC und AC.
DC liefert gemittelte RMS-Werte, die den Schallpegel in Pascal angeben und der Trägheit der Anzeige folgen. Bei den meisten Geräte kann die Trägheit zwischen Slow und Fast umgestellt werden. Das DC Signal kann mit Abtastraten von 10 Hz oder weniger abgetastet werden um eine vernünftige Anzeige zu erhalten. Wie die RMS-Werte skaliert sind, muss man dem Handbuch des Messgerätes entnehmen. Das scheint im Schaltplan aber funktioniert zu haben.
AC liefer die Schallschwingung die bei einem Sinus mit 94 dB z.B. zwischen -1,41 und 1,41 Paskal liegt und für die FFT geeignet ist. Der RMS-Wert dieses Signals ist genau 1 Pascal. Das AC-Signal muss unbedingt mit hohen Abtastraten von z.B. 50 kHz erfasst werden. Aus dem AC-Signal kann man durch RMS-Berechnungen über Zeitfenster auch nachträglich wieder ein DC-Signal machen. Da Schallsignale je nach Situation in völlig unterschiedlichen Größenordnungen liegen, haben die Messgeräte unterschiedliche Verstärkungen, die man im Schaltplan beim Umrechnen der gemessenen Volt in Pascal oder dB berücksichtigen muss.
Für eine genauere Antwort oder ein Beispiel müsste ich aber genauer wissen was da aus dem Gerät raus kommt.
mit freundlichen Grüßen
Ulrich Bierwisch

How can I convert a 0-50mA signal to 0-5 VDC for use by a PCI-6014 DAQ?

I have a PCI-6014 DAQ card. I am trying to measure the output of a closed-loop current sensor, which has an output of 0-50mA. When I tried using a 100 ohm resistor as a load (to measure across the resistor to get a voltage input for the DAQ) I get an AC voltage reading! I do not know how to convert the AC Voltage to a varying DC voltage. Most AC-DC converters output a fixed DC voltage (3.3V, 12V, 15V, etc.). Can the PCI-6014 measure the current directly OR is there a method to measure the VARYING 0-5 volt AC voltage? Please be specific, if possible. Thanks in advance!

Hello,
PCI-6014 can measure an AC voltage directly. The Test Panels in MAX will show the peak voltage level of a signal, and not the root mean square (RMS) voltage. For example, if you measure the 120 VAC 60 Hz sinusoidal waveform that is available from a standard North American wall outlet, you will see a wave with a peak of about 170 V.
The conversion between VRMS and Vo is VRMS = Vo/√2.
You will need to determine whether you expect your AC signal to be in units RMS or peak-voltage and make the conversion, if necessary, using the formula above.
-Arun V

What is the MultiMeter telling me?

I have a project with midi tracks and audio tracks recorded at very conservative levels. I always record and mix at low levels. My Out 1-2 fader is set to -14db with a peak of -9.6db. My channel strip peak in my mixer view is -8db. As I said none of my audio tracks were recorded "hot" in the slightest.
But yet my MultiMeter peak is 4db above 0db. I noticed that regardless of what my out 1-2 fader or my master fader are set at the MultiMeter still reads the same overs. But adjusting my channel strip faders does effect the level of the MultiMeter.
I realize that in some instances that while no channel strip could have overs there could still be an over in the master fader through cumulative audio signal. But why is the MultiMeter not affiliated with my out 1-2 or the master fader?
So what does this mean? Before I bounce a mix to disk for further editing does it matter what the MultiMeter reads? Should I rely on my out 1-2 fader for the levels that are actually going to be bounced to disk?

Hi Franz, your piano music kills my hands...
Just a couple of things:
The Multimeter reads prefader since it is an insert. If you have your master fader at anything lower than 0dB (unity) then you can subtract that from what the channel meter peak values will be. So if your multimeter reads a top peak level of -2dBfs and your fader is at -10dB, then your channel meter will read -12dBfs for its top peak value of the same source.
You don't say if you have anything else (like an EQ, compressor,etc) after the multimeter, you shouldn't for clarity sake.
You should probably set your output (Output1+2) to unity gain along with the master fader. If your multimeter is over 0dBfs, then turn down your tracks. It goes beyond the scope of your question why this is, but while Logic can handle high channel levels, other things related (like plug-ins and summing issues) can't and the sound will suffer. Your mixes will thank you.
If you are in 24bit, realize that you would have to reduce your level 40dB to bring it down to fullscale 16bit so you have headroom to burn. If your monitors are even semi-calibrated to around 83dBspl at at -20dBfs signal, you will have a healthy sound coming out of the speakers with plenty of digital headroom left over. If you want to spend that later on (mastering, etc) to make your mix louder, you have a plenty of options.
Back to your question. The multimeter is also showing you the RMS and Peak values at the same time. The dark blue will agree with your channel meter unless you have the fader set below (or above) the 0dB mark (unity). The light blue is RMS and is showing you an average level - this is important as this is more aligned to how we hear loudness issues.
Again, you can take this or leave it but here is what I would do:
1) Put the Output1+2 and master fader at 0dB and leave them there.
2) Get your monitoring system at least close to -20dBfs equating to 83dBspl - if you have to turn down your master faders because you are clipping but it isn't making your ears hurt, then turn UP your monitoring system. Mix to an RMS level of -20dBfs and then 'master' or finalize your mix wherever you feel the need to. With this method you have options and you can control the trade off between pure transients and loudness.
3) Make sure that the Multimeter is the last plug-in on your chain or it is worthless as a meter.
4) After you have #2 above, the light blue segment will be the thing you look at the most.
hope this answers your question and helps
Paul

Amplitude and level

Dear sir
What is the difference between amplitude & level measurement with vibration level measurement? I found both has measurement such as RMS and Peak value, but it gave a quite different value when I try it simultaneously.
Can I used the amplitude & level measurement to measure vibration level?

Product differences:
Amplitude and Levels Step is activated by SignalExpress
Vibration Level Step is activated by Sound and Vibration
Functional differences:
Amplitude and Levels Step gives a new result for every block (Step will hold peaks if you check that option)
Amplitude and Levels Step supports measurement of DC
Vibration Level Step always returns the result from the last reset
Vibration Level Step does not measure nor remove DC component; instead this step assumes vibration around equilibrium and/or that the data was acquired with AC coupling
Vibration Level Step supports time-domain integration (to convert acceleration --> velocity --> displacement) prior to the level computation.
Vibration Level Step supports industry-standard (fast, slow, impulse) and custom time constants for exponential level measurements.
What types of vibration are you trying to measure?
Vibration during shock event --> Amplitude and Levels
RMS level of an arbitrary signal --> Amplitude and Levels
Machine or structural vibration --> Vibration Level
In many cases, both steps can be used simultaneously Amplitude and Levels Step outputs can be used for trigger conditions. Vibration Level output (such as Exponential Level and Max-Min) can be used for logging and tracking of industry-standard levels.
Doug
NI Sound and Vibration

I need help in writing siginal in file

Hi ,
I have faced problem in writting to measurement File. I have received an error state (( These cannot be wired together because their data types (numeric, string, array, cluster, etc.) do not match. Show the Context Help window to see what data type is required.
The type of the source is cluster of 3 elements.
The type of the sink is Dynamic Data.))
How I slove this problem?
I have attached the labview file. i would be so grateful if you help to figure out the problem.
Khalid
Solved!
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Attachments:
Test (6) (11).vi ‏128 KB

Hi engkhalid,
-Yes, the Output of FFT-(RMS) is a cluster of three elements, which will not be accepted by Write to Measurement File.
-May be you can try Unbundling the cluster and build the array and then send the data, But this will alter how the data is saved in the file i guess.
-Please find the attachment for the modification i suggested.
Thanks
uday,
Please Mark the solution as accepted if your problem is solved and help author by clicking on kudoes
Certified LabVIEW Associate Developer (CLAD) Using LV13
Attachments:
Test (6).vi ‏160 KB

'Match Volume' of songs for radio show

Hi,
In the next little while my radio show will be going from 1 local station to 6 others accross Canada, which requires me to track and produce the show entirely in Audition and FTP it rather than just in RCS at the station.
There's really only one thing that I'm stuck on and I'd love some help...
Obviously not every digital music file is the same volume, so I need to match the volume for all of them. Within the 'match volume' panel there are several options (ITU-R BS.1770-2 Loudness, Total RMS, File, Peak Amplitude, etc.).
Which one should I go with? Should I use limiting? What should I set the loudness to?
I didn't go to school for any of this, just another YouTube education.
Oh, the music style is rock if that means anything.
Thanks for the help!

Your ears are really the best instrument for judging the relative loudness of music tracks. However the ITU-R BS 1770 is the one that a lot of broadcasters specify I believe. Most broadcasters also provide programme delivery specs as to how they want the audio treated.
As to compression and limiting. This will depend on what the stations themselves apply to the audio on it's way to the transmitter. Many use Optimod or similar systems to make their content sound as loud and punchy as they like to compete with other stations. So any dynamic compression or limiting that you apply will be totally wiped out by their transmission processing. So don't do too much, just enough to match your music tracks to one another. Then let the broadcaster spoil the whole thing with their processing.

Breadboard to DAQ Converters for prototyping

I recently designed something for my own testing but I thought it might be
extremely valuable to anyone who connects breadboarded circuits to data
acquisition boards. It is a small and inexpensive converter board that
allows you to connect 50-pin or 68-pin SCSI cables directly into a
breadboard. All pins are labeled on the board and the board is about 0.8 x
3.5 inches in size. If you want to learn more about the converter see
http://www.vicsys.com/pcbprods/bbdaqcvt.htm
Best Regards,
Neal Pederson, VI Control Systems
1923 Mendius Lane, Los Alamos, NM 87544
TEL: (505) 662-1461, FAX: (505) 662-7724
[email protected], www.vicsys.com

Hello,
PCI-6014 can measure an AC voltage directly. The Test Panels in MAX will show the peak voltage level of a signal, and not the root mean square (RMS) voltage. For example, if you measure the 120 VAC 60 Hz sinusoidal waveform that is available from a standard North American wall outlet, you will see a wave with a peak of about 170 V.
The conversion between VRMS and Vo is VRMS = Vo/√2.
You will need to determine whether you expect your AC signal to be in units RMS or peak-voltage and make the conversion, if necessary, using the formula above.
-Arun V

IMovie-to-GarageBand: Dialogue Levels?

GarageBand and Audio newbie with a somewhat advanced question:
How do I use GarageBand to properly balance audio levels from various video clips from various sources (iMovie doesn't offer enough precise control)? Some of the clips have normal dialogue, some have whispering, some have shouting and some have all three. I would like to pick the most dynamic clip and use it as the standard, but if it ranges from whispering to shouting, to what level do I match a clip that is all whispering?
Also, is there such a thing as broadcast audio level standard? I want the master level set so no one watching the finished DVD will have to readjust their television to hear or keep from being knocked out of their chairs. If such a standard exists, what is it?
Thank you so much for your help.

Okay, ghankstef, I've toyed with RMS buddy a bit and have a few more questions. I'll number them to save you the trouble of having to type the question you're responding to:
1. RMS Buddy has four readout categories, each with a left and right channel: Average RMS, Continual RMS, Absolute Peak and Continual Peak. Which of these am I to meets to -17.5db RMS?
2. Beneath the readouts there is an adjustment for Analysis Window Size ranging from 30ms to 1000ms (ms = milliseconds, I assume). What sample size do I want to set this to?
3. From the numerous samples RMS Buddy will take from each clip, which do I want to use as my standard? In other words, if my lowest sample is -30db and my highest is -20db, which of these do I want to bring to 17.5 RMS?
4. The Track can be increased +6db and the Master can be increased +6db, but I'm assuming from your wording that I only want to increase the level on the Track -- not the Master. Is that correct? (By the way, each clip only has one track, so tweaking the Master would not affect anything but that track.)
5. Just to confirm my understanding: Run RMS Buddy on the Track, and run AU PeakLimiter on the Master. Is that correct? Or is that only the case if there more than one track exists?
Dude, thank you so much. I hope numbering the questions makes it easier to respond. Feel free to answer simply (i.e., 1. Average RMS, 2. 1000ms, 3. -20db, etc.). Any additional explanation is just bonus.

SB Live! input impedance

I would like to record my acoustic piano.
I have a Realistic (Radio Shack) electret condensor PZM (boundary) microphone. It has DC power. It s impedance is 600 Ohms, the sensitivity is -74 dB (0.2 mV/Pa). If I assume that its max SPL is 20 dB, then my microphone max output level is 26 dBu.
I want to use that microphone to record into my Sound Blaster SB Li've! version 2.00.32 sound card. It has a EMU0K audio processor. The sound card is an internal sound card and, by looking at my computer, it looks like the number 8900/0 has been stamped into the metal.
What is the impedance of the sound card input
Should I record using MIC IN or LINE IN
Which pre-amplifier specs should I look for if I intend to purchase a pre-amplifier
Thanks
JJW

Re: SB Li've! input impedance? The model/serial number on your SB Li've would be useful in identifying your particular card. If you have a mic input jack, use that for your microphone. You don't need a preamp for that except for a low impedance "studio mic." Here are the specs for input/output of Creative Soundcards of that vintage:
MIC-IN
Sensitivity : 0mv to 200mv
Impedance 500-600 Ohms
Stereo input jack:
TIP Signal
RING Bias (if available)
SLEEVE Ground
If it's a condenser mic the TIP and RING should be tied together.
Studio mics, like low 25 Ohms, do not work with PC sound cards.
Suggestion: For cleaner recordings use a powered/pre-amped mic
into the line-in.
Line-In:
Line level inputs.
Sensitivity: 0v to 2v Most new PCI cards are RMS, most or all
the old ISA cards are peak to peak.
Impedance 47K-50K Ohms.
Normal input should be about V.
Line-Out:
0 to 2V RMS or peak to peak.
Normal output level about V.
Impedance assumed 0K Ohms, some cards 600 Ohms.
Described as: Gi'ven a full Sine wave at Khz tone
and a 0K Ohm > Load should produce about V RMS
output.

FFT RMS or Peak?

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