STI measurements

Very interesting topic.

One question on the procedure: if I cannot measure the pink noise level at 1m from the source (to assess the 60dB(A)) but I am able to do the rest of the calibration, can i create a pink noise signal with the right value compared to dBFS and use it as my calibration file?

If yes how can I define the amplitude?

Something like this: 10(-6dBFS/20) = 0.5

thanks

Some part of the new standard rely on absolute SPL correction factors, hence the answer is NO. You cannot make a calibration "realtive" to full scale, you need to calibrate your source in "absolute" dB-SPL.
However, I do not see the problem.
I usually calibrate the source at the beginning of the measurement session. I just place the SLM at 1m from the source, on axis, and I adjust the knob until I read 60 dB(A). It takes perhaps 1 minute, so what's the problem doing this?
What can be done in advance is to create the pre-equalized pink noise WAV file to be played on your specific source.
While the loudpeaker is behaving linearly, the equalization is independent on the the gain (and of consequence on the absolute SPL value).
For creating the pre-equalized pink noise WAV file you start with a flat pink noise, generate with Adobe Audition, long, say, 1 minute.
You play it and record with a microphone in front of the source, at 1m.
Make an octave-band analysis using the Aurora STI module, and copy and past the spectrum to an Excel spreadsheet.
Evaluate at each ovtave band the difference between the octave-band level and the total A-weighted level, and write these number below those mandate by the standard, and shown in the table above (for male speaker).
Of course there will be discrepancies, at some frequencies the measured value will be larger and at other frequencies the measured value will be smaller than the value in the IEC table.
At each frequency, you calculate the correction (value in the table - measured value).
These are the offsets to be applied to the pink noise wav file, using the Audition's graphical equalizer effect, set in octave-band mode.
At each frequency, move the slider (or key-in the value) of the offset computed in Excel, then hit apply.
The result will be a new WAV file containing equalized pink noise.
You can repeat the playback & recording again, for checking that now the resulting octave band spectrum is compliant with the table in the standard. If it is not, compute again the residual correction to be applied, and perform again the graphical equalization in audition.
repeat until the measurement shows a spectrum which is perfectly compliant with the IEC standard.
this WAV file will be used for feeding your loudspeaker when doing measurements. after adjusting the total A-weighted level to 60 dB(A) at the beginning of each measurement session, as explained.

I agree with what you describe but I have to say it is not always possible to follow this procedure.
If you use your own source, this is easy to do, but if you have to measure the STI performance of a sound system in a big arena, it is not always easy to go up and put your SLM at 1m from the big line arrays.
Also I would question if your measurement at 1m from a line array is representative of the SPL @1m of the system.

So there is the need of an alternative to this in my view, as I happen to do quite a lot of this work.

Probably there is some misunderstanding of the procedure...
At 1m from a powerful line array, the equalized pink noise will probably be well above 100 dB(A)!
The calibration at 60 dB(A) at 1m is for the artificial mouth. When there is a PA system the artificial mouth should be placed at a few cm from the microphone employed for the announcements.
Then the gain of the PA system is adjusted until the system delivers the contractual SPL at the reference listening position. A typical value can be something as 80 dB(A) in a listening position very far from the loudspeaker.
In venues where the microphone is inside tha same room as the loudspeakers, usually the gain is limited by the feedback.
This is not an issue when the microphone is in a separate room, so there is no feedback.
In this case, it is also possible to inject directly a dummy microphone signal in the PA system, obtained recording the signal from a typical PA microphone placed in front of the calibrated artificial mouth.
When instead the microphone is inside the venue, the use of the artificial mouth in front of the microphone is mandatory, you cannot measure STI injecting an electrical signal into the system.

Hi Angelo....here's Winhay another aurora user.

I have a doubt with the STI measurements.

Specifically with impulse response for STI

I readed we can use a sweep signal with no equalization and no necessarily 62 dBA......ok, but about my source??.....

For example, can I use the impusle response used for evaluating acoustical parameters for evaluating the STI parameter?...(omnidirectional source)

or

When I measurement STI parameters I MUST obtain the IMPULSE response with a directional speaker ( emulate directionality of the male voice)


I know I'm must use a directional speaker for obtain the filter signal of the source......but If i have already the impulse response of a room
measured with a omnidirectional source it's necesary obtain the impulse response with a directional speaker?....

I hope my question can be understand

Regards

FedeleDeMarco.

thanks! for the answer....it's very useful for me reinforce this kind of concepts......regards!

Hi Pr Farina,

First of all, thanks a lot for making all your tools available for anyone, I appreciate it !

I am trying to use the STI module in Audacity, so I have my 4 required tracks (calib, background, signal and IR), I perform the calibration and once I'm in the SNR Calculation tab, after selecting the background noise and signal tracks, when I click on "Compute levels" the levels do not seem correct (around -100 dB).

I also tried amplifying the signal track to see if it changes the calculated sound levels, but there is no change.

Is that normal ?

Thanks in advance for your help

Hi,
Thanks a lot for taking the time to answer. Here are the 4 files (IR still needs to be computed from Sine Sweep recording) :

drive.google.com/drive/folders/10JPh9vIboQi20S2-awWx5_CF-_ywr-vf?usp=sharing

I also included 2 pictures in that folder showing what happens when I use your recordings to compute the STI ("STI-Background-S+N.png" and "STI-calib.png").
As you can see the computed sound levels are far below 0 dB, so I'm thinking it could be a problem with the plugin?
Thanks again for your help !
Best regards
Christophe