# How to make quasi-anechoic speaker measurements/spinoramas with REW and VituixCAD

#### morpheusX

##### Active Member
I'm not sure why you think the 1.2K is an error, a dip right after the baffle diffraction peak is quite normal in sharp edged cabinets.

I'm sorry if i wasn't clear, but i wasn't referring to the 1.2kHz dip, but instead, to the 1.2kHz peak.

RED: Far field, 1.7m from floor; Green: Ground Plane; Purple: Nearfield with diffraction:

You might need to do some experimentation to see where the best place is to get the longest gate.

I've measured the attic, and the exact (rounded) measures are 7920 mm x 5800 mm.
So if i use a circumference with a 2900 radius, and given the speaker is 1000 x 400 x 400 mm, it would give me 2.7m free of reflections:

If you want to avoid using the platform then experiment some more with the ground plane placement. You can look to try and undo the extra baffle added by the ground plane by comparing the difference in Vituix and seeing if it is significant enough to worry about accounting for.

I'll look into it, thank you for your detailed post!

#### fluid

##### Addicted to Fun and Learning
I'm sorry if i wasn't clear, but i wasn't referring to the 1.2kHz dip, but instead, to the 1.2kHz peak.
Right, that does look to be due to the smoothing effect of a short gate. Change the gate time and see what happens to the response.

I've measured the attic, and the exact (rounded) measures are 7920 mm x 5800 mm.
So if i use a circumference with a 2900 radius, and given the speaker is 1000 x 400 x 400 mm, it would give me 2.7m free of reflections:
I don't know how clear this diagram will be but here goes, if you put the speaker and mic one metre away from the centre point of the space along the longest diagonal you should get the longest reflection free path length. In this example if I try and trick Vituix to calculate it for me it gives nearly 16ms reflection free time.

Inside or outside normally the floor or ceiling set the limit but with ground plane you can leverage the angles to extend the window. Try it and see if the theory holds

#### morpheusX

##### Active Member
If i understood everything, the distances introduced on VituixCAD, 1) are from:

1) From the center of the speaker, in a 90º angle with the diagonal = 2862 mm
2) From the tip of the mic, in a 90º angle with the diagonal = 4520 mm

Isn't this valid just for the 0º measure?
Or, given that the axis of rotation is always the same, the distance to the boundaries is always the one used on 1) ?

#### OWC

##### Active Member
Inside or outside normally the floor or ceiling set the limit but with ground plane you can leverage the angles to extend the window.
Don't forget the back wall as well, that counts for the speaker as well as the mic.

For this reason it sometimes can be useful to measure anti-parallel to the walls of the room, ideally at 45 degrees.
As a rule of thumb, the distance behind the speaker (as well as the microphone) should be at least the measuring distance.
Although this can be calculated on a way that is very similar to the floor/ceiling reflection.
This is just basic geometry math.

Btw, depending on the height of the ceiling, but in most houses it's not floor OR ceiling, but floor AND ceiling.

#### OWC

##### Active Member
edit:
Here a quick example

Let's assume we measure at 1m distance and 1 meter height, there is no ceiling, the back-wall is at 1 meter behind the front of the speaker, there is no wall behind the microphone (to make things a little easier).

The first floor bounce will be at 344/((2*sqrt(0.5²+1²))-1) = 278Hz
The reflection from the back-wall will be 344/(3-1) = 172Hz

If this back-wall would be at 0.5m we get, 344/(2-1) = 344Hz

With a Hann 25% or Cosine window, one could go a tiny bit lower

#### gabo4au

##### Member
edit:
Here a quick example

Let's assume we measure at 1m distance and 1 meter height, there is no ceiling, the back-wall is at 1 meter behind the front of the speaker, there is no wall behind the microphone (to make things a little easier).

The first floor bounce will be at 344/((2*sqrt(0.5²+1²))-1) = 278Hz
The reflection from the back-wall will be 344/(3-1) = 172Hz

If this back-wall would be at 0.5m we get, 344/(2-1) = 344Hz

With a Hann 25% or Cosine window, one could go a tiny bit lower

So what is that base formula?

#### OWC

##### Active Member
So what is that base formula?
For which one? Although both are just very standard geometry mathematics.
With the floor/ceiling one uses the Pythagoras rule for the total distance.

When this is known it's just a matter of subtracting the direct sound from the indirect sound.

After that it's just using dx = v * t = v/f
resulting in; f = v/dx

Where;
dx = distance in meters
v = velocity of sound (m/s)
t = time (s) -> 1 over f = frequency (Hz)

#### gabo4au

##### Member
For which one? Although both are just very standard geometry mathematics.
With the floor/ceiling one uses the Pythagoras rule for the total distance.

When this is known it's just a matter of subtracting the direct sound from the indirect sound.

After that it's just using dx = v * t = v/f
resulting in; f = v/dx

Where;
dx = distance in meters
v = velocity of sound (m/s)
t = time (s) -> 1 over f = frequency (Hz)

Thanks, sorry for being such an idiot! haha. I understand the math, just didn't know how it related to the frequency. Thanks for helping me understand.

#### fluid

##### Addicted to Fun and Learning
Isn't this valid just for the 0º measure?
Or, given that the axis of rotation is always the same, the distance to the boundaries is always the one used on 1) ?
When the mic and rotation axis are the same it should be valid. What will change is the strength of the reflection based on how the speaker is turned and it's directivity. Try it either way and see what gives you the better measurement to work with.

#### nathan

##### Active Member
This is a guide to taking simple quasi-anechoic measurements for the least hardware expense possible, using Room EQ Wizard (donate!) and VituixCAD (donate!). This way, I can also stop repeating myself every time someone asks me how I do it!

Warning: this guide will be wordy. My hope is that even a beginner can learn to do quasi-anechoic speaker measurements this way, so I apologize if I repeat myself or state some obvious things. I started measuring speakers with absolutely no engineering background and barely any knowledge of acoustics, and I wished there'd been a wordy guide for me starting out.

I'd like to start by acknowledging the late Jeff Bagby, whose white paper on quasi-anechoic measurements is basically how I got started; much of what's in this guide is essentially just a 'translation' for REW. Of course, Dr Toole's book was invaluable for the initial inspiration and teaching me how to interpret that data. I later took this Udemy course which helped clear up some questions I had about quasi anechoic measurements. And thank you to Amir for providing a platform to emphasize speaker measurements, as well as Stereophile, Soundstage Network, Erin/hardisj, and others who provide valuable sources of speaker measurements that I've often used to compare my data with.

This guide will be divided into 6 parts. How many you read depends on how thorough you want to be with your measurements:
1. Introduction to quasi-anechoic measurements
2. Setup and gear
3. On-axis measurement (sans low bass)
4. Nearfield bass measurements
5. Off-axis measurements
6. Create a full shebang spinorama
Each part will build on the previous ones, so you don't have to the full guide. If you're already familiar with the ideas behind quasi-anechoic measurements and just want to know how to do them in REW, you can skip the 'On-axis measurement' section.

Please give this a try! Even if you just perform a single on-axis measurement without bass, that's already a lot more useful than most of the speaker information available on the web.

Finally, please keep in mind this is just one way of doing things, lots of which I've learned through trial and error. I am open to feedback =]

Update 4/12/21: Fixed some typos, reworded some bits for clarity.
Update 5/11/21: Fixed some more typos, reworded more bits for clarity.
Update 5/20/21: Added a reminder to make sure sample rates for input and output device match (should be 48 kHz with Umik-1)
Update 9/7/21: Added an acknowledgment to a Udemy course I'd forgotten I'd taken which helped me learn as well.

1) Intro

The best thing you can do is send your speaker to Amir or Erin for testing with the Klippel NFS. But if you can't do that for whatever reason, creating quasi-anechoic measurements can help contribute to the pool of valuable speaker data. Even a single on-axis measurement can be extremely useful.

Quasi-anechoic measurements are basically a way to take a speaker measurement indoors or outdoors and ignore the influence of walls (including the ceiling and floor). The gist of it is to make a sine sweep in REW and truncate the impulse response such that REW only includes the data from right before the first major reflection 'hits' the microphone (it's much easier to do than it sounds!).

View attachment 120852

This is called 'gating' or 'time-windowing' the impulse response. In doing so, you lose some resolution, which is most apparent at low frequencies, and the data becomes completely invalid for the bass (usually below 100-200 Hz). The wider the gate, the higher the resolution of your data. A 5ms gate, such as is used by Stereophile, will give you a resolution and lowest valid frequency of 200Hz. My measurements are typically done at 6.5ms, which gives me a resolution of 154Hz. The resolution calculation is 1/[window in seconds], so 1/0.0065, though REW will let you know too.

To make up for the lack of resolution at lower frequencies, we can take super-nearfield measurements of the speaker's bass components (woofers, ports, and passive radiators), and simulate the far-field bass response from it. (Another common, even more reliable method for bass measurements is the ground-plane method, but that requires an ample amount of space, so I've never really used it).

With a bit of care and trial-and-error, you can get results that greatly approximate those made in an anechoic chamber or with the Klippel NFS. For some validation of the method, and an idea of what you can expect, here are some examples of my own measurements compared to anechoic sources.

JBL HDI-1600 (vs Amir's NFS):

View attachment 120854

View attachment 120855

D&D 8C (vs Erin/hardisj's NFS):

View attachment 120856

Focal Chora 806 vs Soundstage Network's at the NRC anechoic chamber:
View attachment 120857

The Spinorama/CTA-2034A standard says that a ±1.5dB measurement agreement for the same speaker is considered 'good'. You can see the above measurements are very close to that, despite measuring different test units.

Note that this does not mean the quasi-anechoic method is as accurate as an anechoic chamber or Klippel NFS. In particular, resolution in the low mids pales in comparison, which means narrow resonances may be obscured partially or entirely. But the data is still very useful for determining trends in tonality and can become effectively equivalent to anechoic ones by the upper mids.

2) Setup and gear

Here's what you'll need:
• Room EQ Wizard. This guide was written with beta Version 5.20 RC6. As of writing this guide there are a several important features in the betas not available in the 'stable' release that is currently available on the REW website (V5.19). In my experience, the betas are extremely stable for the type of work we're doing
• (If splicing nearfield bass) The Jeff Bagby Diffraction and Boundary Simulator, for adjusting nearfield bass measurements to match farfield results. This requires Excel; I've not tried it on Google Sheets or other spreadsheet software.
• (If doing full spinoramas) VituixCAD (version 2.0.65.0 was used for this guide). It will automatically create a spinorama once provided with enough horizontal and vertical off axis measurements. It can also adjust nearfield bass measurements, but I prefer the simplicity of the Bagby spreadsheet.
• A MiniDSP Umik-1 or other flat measurement microphone. If you don't already have one, I'd highly recommend getting a calibrated Umik-1 from Cross Spectrum Labs for extra accuracy. It only costs a few bucks more than ordering one directly from MiniDSP (\$110+ shipping). It's not necessary, but increases accuracy in the upper treble and lower bass and adds peace of mind.
• A microphone stand. It just needs to be thin so as to be minimally reflective. I use something like this, about 20 bucks.
• A sturdy way to elevate speakers far off the ground, preferably 5+ feet, but as far from surfaces as you can manage. I've typically simply placed my speaker stand on top of a table. The sturdiness of the speaker stand is particularly important if you want to do vertical measurements. I'm currently using this.
• (If doing off-axis measurements) You'll need some kind of turntable to place your stand on. I use this and label it with angles in 5 degree increments. For added security, especially for vertical measurements, I highly recommend getting a rachet or cam strap to secure the speaker while it's off-balance. I use one or two of these.
• Open space. If measuring indoors at 1m — sufficient distance for most bookshelf speakers, in my experience — you'll want the closest wall (including the floor and ceiling) to be about 1.5+ m (5+ feet) to match the time window and resolution I've used in most of my measurements (6.5ms). You'll also want to move all furniture out of that 1.5 foot radius — or as far as possible — but small objects shouldn't cause much of a problem. If you have low ceilings and can't measure outdoors, you might have to settle for a smaller gate or measuring at less than 1m.
The open space is key. Again, the greater the time difference between when direct sound hits the microphone and the first big reflection hits the microphone, the more resolution your data will have, and the lower the frequency your measurement will be valid to.

When setting up your speaker on the stand, it should look something like this (taken from the CTA-2034A standard):
View attachment 120858

It is important to make the edge of the speaker stand as flush as possible with the speaker's baffle, as otherwise the setup can introduce minor reflections that might look like resonances. And again, I'm using a 1-meter distance, rather than the 2m the spinorama standard technically asks for, in order to increase the available time window.

Neither 1m nor 2m are magic numbers, by the way. For horizontal measurements of small speakers, simply being 2-3x the baffle width is usually enough. For a single on-axis measurement of a small speaker, you might get away with less than 2 feet. Experiment and see how the response changes at different distances, and find the best compromise for your space. Vertical polar measurements will be the most affected by short distances, so I would try to keep at least 1 meter for those for most speakers.

Don't sweat your setup too much. It doesn't need to be too fancy. This is what I used for the JBL HDI-1600 measurements above (set up for vertical measurements):

View attachment 120861

The important thing is to simply minimize reflections enough to keep your data sufficiently clean to be useful, which you can readily assess from the resulting frequency and impulse response. If the impulse response looks messy or the frequency response looks unexpectedly 'squiggly', try to move stuff around to make it as clean as possible, then remeasure. It'll take some trial and error, but again, don't sweat it too much. Perfect is the enemy of good.

One more note: make sure that the sample rate for your input and output devices are the same (Thanks for the reminder @sweetchaos). The Umik-1 can only operate at 48kHz for example, so you'll want your audio output to be at 48kHz as well. Many devices will default to 44.1 kHz and using a different sample rate can have a slight effect on the highest frequencies in my experience. Using a higher sample rate won't improve accuracy, per REW documentation.

On Windows 10, you can do this by going to Sound Settings> Sound Control Panel, tapping on your playback device's properties, and then changing the sample rate in the 'advanced' tab.

View attachment 130931

You should also make sure any spatial audio effects and the like are turned off.

3) The On-Axis measurement (sans bass)

The most basic quasi-anechoic measurement you can do is a simple on-axis sweep.

It's way easier and faster to perform, say, a single on-axis quasi-anechoic measurement (or even a few horizontal off-axis angles), than to do a full vertical and horizontal spinorama with nearfield bass spliced in. In fact, if you can position the speaker fast enough, it only takes a few minutes to do.

As noted earlier, creating open space around the speaker is key and your setup will likely take the most time in this whole process. Before even making a quasi-anechoic measurement, simply moving your speaker away from walls and measuring from closer — thereby minimizing the 'loudness' of reflections — cleans up the data a lot.

To illustrate this effect, here is an old measurement of the Buchardt A500. This is an on-axis measurement taken as a single sweep from my listening position 3m (~10ft) away:

View attachment 120862

This doesn't tell us much about the speaker's direct sound.

Now here is another measurement taken from just 1 meter, after repositioning the speaker such that it is 5+ feet from every wall, including the floor:

View attachment 120863

The highs are much cleaner now, and we have a better idea of the speaker's sound, but this is still not terribly useful. Next, I'll show the exact same measurement file you see above, except with a gate or time window applied. Note that this was not a separate sweep, I am simply modifying how REW interprets the same file:

View attachment 120866

That's more like it! Although we lost the bass response, we have now removed the 'noise' of the room and have something that tells us something much more useful about the "true" direct sound of the speaker.

Here's how you do it.

Again, position your speaker as far away from walls as possible. Make the speaker's baffle flush with the edge of its stand. Aim your microphone at the speaker's reference axis; check the manual, but if not stated, it's usually the tweeter or midway between the tweeter and woofer. If you're using a boom microphone, try to keep the arm extended such that the microphone is far from the 'stem' to minimize reflections near the microphone.

View attachment 120868

(Ideally, the boom would be in line with the microphone, but I couldn't get it high enough in this case).

Then just take a regular sweep measurement in REW. I assume most of you know how to do this, but it can be done from the 'Measure' button on the upper left (shortcut: Ctrl+M). These are my usual settings:

View attachment 120869

(Ignore the output and input settings, as the microphone wasn't connected when I took this screenshot).

There is one important setting in the 'measure' window that you should keep in mind for doing off-axis measurements later. By default, REW sets t=0 at the IR peak, but this causes problems once you go more than 90 degrees off-axis (basically, the reflection off a wall might be louder/have a higher IR peak than the direct sound). So it's better to set it to t=0 at IR start. The resulting FR should be the same.

View attachment 120870

Now tap start (or press the spacebar), and once the sweep is complete you have all the data you need!

From here, we just need to change the way REW interprets the data to get our quasi-anechoic measurement.

Head over to 'Impulse' tab, and make sure you're in the percentage view on the upper left. The impulse response shows us the same FR data we just captured from a time perspective.

View attachment 120871

You should now see something like this:

View attachment 120872

See that blip right before 7ms, and how the data is all messy after that? That is where the first reflection hits the mic (each subsequent 'blip is another reflection). Were going to remove those blips from our data. You may need to finagle a bit with the zoom controls on the upper left and bottom right corners to get a good view of the blips:

View attachment 120873

If you're measuring outdoors, you might not see such pronounced reflections. That's fine; don't worry about it too much, as we can always adjust the gate later.

Now tap on 'IR Windows' at the top of the REW window. Set the 'right window' to a time just before your first reflection. In my case, I set it to 6.5ms. The left window is usually not very important, but if measuring outdoors, it may help to shorten it to about 2-5ms to prevent loud sounds from contaminating the data.

View attachment 120874

If you're doing off-axis measurements, it's good to leave yourself a little 'slack' between your window and the first reflection, as sometimes distances change a little bit as you're rotating the speaker. Hence me using 6.5ms even though I could stretch the window a little higher.

As noted earlier, the longer you have before the first reflection hits the microphone, the more resolution you have in your data, and the lower the frequency your data is accurate to.

REW will tell you what the frequency resolution of your measurement is, which will also be the lowest frequency the data is useful to. As shown above, 6.5 ms gives a resolution of 154 Hz. If possible, try to get at least a 5ms gate (this is what Stereophile uses, for reference, although they measure from a further distance), which has a resolution of 200Hz. Still, even a smaller gate can be useful, just know you'll have lower resolution.

And that's basically it. Once you tap on 'Apply Windows' you should now see a cleaner frequency response. You can also get a live view of the changes caused by changing the time window by dragging the green 'R' marker at the top of the Impulse tab.

Some miscellaneous notes:
• Make sure REW is set to a reasonable scaling to get a useful view of data. It's easy to obscure flaws in the frequency response with very tall scaling. If you tap on 'Limits' in the All SPL window, the SPL Top and Bottom should be a 50dB difference in most cases.
• For a better way to ensure consistent scaling when sharing your frequency response, I recommend using REW's built-in 'Capture' button on the upper left. Under 'graph aspect ratio,' select 25 dB/decade. This is technically the aspect ratio defined by the spinorama/CTA-2034A standard too, although not even Harman uses it most of the time. The good thing about using this method is that even if you set different vertical limits than the usual 50dB, your frequency response will export at the same scaling.
• 1/24 is my preferred smoothing.
• You can make your frequency response dashed or dotted by tapping 'Controls' and then 'Trace Options.'
• As we're not measuring sensitivity for this guide, SPL choice isn't terribly important. 85dB @1m is a reasonable SPL, but I used 75dB for a long time to not annoy neighbors. It matters most when using DSP speakers whose frequency response might change (compress) significantly with SPL level.
• Sometimes there are objects that cause unexpected reflections. Others matter a lot less than you'd think (like your own body, sometimes!). You can usually tell if something is amiss by how 'messy' the impulse response looks, or if the frequency response looks unexpectedly wiggly. Again, trial and error. Mess around with positioning and settings until you find something that works consistently.
Phew. I know that was a lot, but you should see it's really not all that difficult. Hopefully, this will get you started!
I just want to say "Thank you" for this whole guide but especially for this set of instructions. I went hunting for this process today in order to test out some second hand speakers. I'm mostly interested in quickly finding any issues but this guide will let me use my measurements for so much more.

#### Nuyes

##### Active Member
Forum Donor
Reviewer
Belatedly thank you. @napilopez

Thanks to your guide, I made a DIY turntable, which has become my most powerful tool and my best friend.

I have only been active in the Korean audio community so far, but from now on, I want to share various information with ASR as well.

Once again, nice to meet you and thank you.

#### dominikz

##### Addicted to Fun and Learning
Forum Donor
Belatedly thank you. @napilopez

Thanks to your guide, I made a DIY turntable, which has become my most powerful tool and my best friend.

View attachment 215321
View attachment 215322

I have only been active in the Korean audio community so far, but from now on, I want to share various information with ASR as well.

Once again, nice to meet you and thank you.
Welcome! I'm very excited to see more contributors of loudspeaker measurements here!

Your turntable looks very robust (much more so than mine ), nice work!
However it seems to me that as depicted it might cause some early reflections to interfere with the measurement - have you tried to estimate this effect, and do you have some means to mitigate (e.g. adding absorption to the stand while measuring)?
When I do my measurements I noticed that adding absorption on the loudspeaker stand (absorbers and thick blanket) and microphone clip helps to get a cleaner quasi-anechoic response.

Thanks!

Last edited:

#### abdo123

##### Master Contributor
Forum Donor
Welcome! I'm very excited to see more contributors of loudspeaker measurements here!

Your turntable looks very robust (much more so than mine ), nice work!
However it seems to me that as depicted it might cause some early reflections to interfere with the measurement - have you tried to estimate this effect, and do you have some means to mitigate (e.g. adding absorption to the stand while measuring).
When I do my measurements I noticed that adding absorption on the loudspeaker (absorbers and thick blanket) and microphone clip helps to get a cleaner quasi-anechoic response.

Thanks!
I was about to say the same thing, I suggest that they at least push the speaker so that the baffle is flush to edge of the wooden surface. For these kind of things you want the thinnest (and sturdiest) pole you can find carrying the speaker, not the current contraption of diffraction points.

#### Nuyes

##### Active Member
Forum Donor
Reviewer
Welcome! I'm very excited to see more contributors of loudspeaker measurements here!

Your turntable looks very robust (much more so than mine ), nice work!
However it seems to me that as depicted it might cause some early reflections to interfere with the measurement - have you tried to estimate this effect, and do you have some means to mitigate (e.g. adding absorption to the stand while measuring)?
When I do my measurements I noticed that adding absorption on the loudspeaker stand (absorbers and thick blanket) and microphone clip helps to get a cleaner quasi-anechoic response.

Thanks!
I was about to say the same thing, I suggest that they at least push the speaker so that the baffle is flush to edge of the wooden surface. For these kind of things you want the thinnest (and sturdiest) pole you can find carrying the speaker, not the current contraption of diffraction points.
The image I attached reflects the initial settings.

Now, the front panel, which occupies a fairly large area, has been removed, leaving only the 12mm base plate.

Also, the edge of the base plate coincides with the axis of rotation of the turntable.

Thank you.

OP
N

#### napilopez

##### Major Contributor
Forum Donor
Belatedly thank you. @napilopez

Thanks to your guide, I made a DIY turntable, which has become my most powerful tool and my best friend.

View attachment 215321
View attachment 215322

I have only been active in the Korean audio community so far, but from now on, I want to share various information with ASR as well.

Once again, nice to meet you and thank you.

Honestly I'm glad I wrote this because now that I don't measure speakers often I'm sure I'll forget some steps myself next time I do! Its due for a revision given some updates to Rew and VituixCAD but I'm glad more to have helped bring more data to the audio community

#### stoneeh

##### Member
Thank you for putting the effort into this writeup.

Since over the last 2 years I have done an extensive empirical study on the subject, comparing (combined) near field & (spliced) farfield measurements to true free field measurements on a total of so far 13 bassreflex enclosures (all kinds; from small bookshelf speaker to 18" PA subwoofer), the results of which I published in an academic article (unfortunately for this thread / forum, in German), allow me to comment.

The method works very well, when executed correctly / appropriately - but nothing about it is trivial.

The nearfield measurement method has been originally documented in Low-Frequency Loudspeaker Assessment by Nearfield Sound Pressure Measurement, D.B. Keele, 1973.

Positioning of the microphone: Keele elaborates on and recommends a position near the center (dustcap). Here's an empirical comparison of mine for a small fullrange speaker, same amplifier voltage, same distance of the mic; dustcap green, outer edge black - we can see a difference, which is mostly minor (~1 dB) over the relevant frequency range:

Distance of the microphone to the radiating sources is also very important, as has already been mentioned in this thread. We cannot just "wing it" here - a few cm distances might already produce a loss of several dB SPL.

In their empirical study / paper On Acoustic Very Near Field Measurements, the authors J.Prezelj, P.Lipar, A.Belšak, M.Čudina, state: “To be within 1 dB of the very nearfield pressure, the pressure microphone must be no farther away from the center surface of the piston than 0,11a”.

So, especially for small sound sources, it is vital to place the microphone as close as possible to the source! For bassreflex ports, the mic can be placed flush with it; for speaker membranes, you'll always have to keep at least a few milimeters distance to allow for some excursion, even at low SPLs - but no more than that.

Proper SPL adjustment of the sound sources is also extremely important, which I will go into more detail in later.

Next, let's look at a two practical comparisons vs. free field measurements, when all important variables have already been accurately / appropriately considered, to demonstrate how close of a match is achieveable.

Methodics, hardware / software, environment & further details:

- Measurements are performed with ARTA, 8k FFT @ 48 kHz, 1/24 oct. smoothing (= essentially none), with a Isemcon EMX-7150 microphone (calibration file loaded).
- All measurements are repeated at least once, and a result only accepted as valid if curves match perfectly over multiple runs.
- Measurements are done with low-level input, as to rule out potential compression effects.
- Outdoor measurements are done in a location far from civilisation, on wind-free, quiet nights. As a result, ambient noise is lower than noisefloor of the signal chain (~ -80 dB). There are no obstacles in the vicinity, meaning the measurement is completely free of reflexions.
- Outdoor free field measurement is done as GPM (ground plane measurement). As documented in the original work on the method, Ground Plane Acoustic Measurement of Loudspeaker systems, Gander, 1980, GPM is essentially a 4pi measurement, adding 6 dB SPL for the mirror sound source of the floor.
- GPM is done on a perfectly flat, hard surface.

One potential point of contention for using GPM as a reference free field measurement would be that the mirror sound source of the floor might not only double the sound pressure of the speaker, but also extend the speaker's baffle. This however can be, and in this case was, compensated for by simply adjusting the baffle step correction (doubling the baffle height) in software.

Green curve for all DUTs is the combined near field measurement, SPL adjusted via a empirically derived self-formulated calculation, bafflestep-corrected in ARTA, spliced with a gated far field response on stand / tripod. Black curve is the free field measurement.

DUT #1 - fullrange 12" PA speaker:

DUT #2 - fullrange broadband 4" DIY HiFi speaker:

This speaker was more complicated to measure - dustcap measurement didn't make sense due to its large height (too much distance to the cone), and the distance to the rear port had to be accounted for - still, when factoring those variables in, both methods again showed great correlation:

One thing to consider is that ARTA's simple bafflestep correction is only really valid in the bass area (up until 100, 150 Hz), and due to gating, the tripod measurement doesn't have much resolution there. So some disagreement in the upper bass / lower mid area is unavoidable. Other softwares with more elaborate baffle step correction will produce even better results.

Also, edge diffraction of the baffle is of course different in a true 4pi measurement vs. GPM, with one edge flush with the ground, which accounts for remaining mid-high frequency differences. VituixCADs diffraction tool might be able to compensate for that too, though, as you can specify a floor / boundary surface.

As we can see, though, bass response of the combined nearfield vs. far field / free field measurements matches almost perfectly - within tenths of a dB! And, rest assured, that goes for the other 11 enclosures / speakers that I have tested as well.

Now, let's quickly get into SPL adjustment of sound sources (port vs. membrane). The known methods are:

- D.B. Keele's original formula (see paper linked in the beginning of this post): 20 log * square root of sd/sv.
- Jeff Bagby's formula (see link in OP): 20 log * membrane diameter / port diameter (or vice versa); no source provided, but this corresponds to the DB Keele formula, taking out the square root for area and switching area for diameter.
- source unknown - method of matching SPL of the curves by hand at a fraction of the frequency of fb, assuming port and membrane to have the exact same output in this frequency region

Worringly, until recently, none of these methods had ever received an extensive empirical evaluation. Sporadic user measurements / comparisons can be observed, but far from enough to claim universal validity.

From my comparisons, I can say these methods mostly produce similar and, depending on expectation, acceptable results / accuracy - mostly within 1-2 dB of the free field measurements.

There are outliers though. The previously covered DUT #2 is such a speaker - the traditional methods for SPL matching produce less favorable results with it. Here's a comparison of in green Keele's / Bagby's formula, and blue SPL matching by hand below fb; left graph port nearfield measurement (in black membrane nearfield response), right graph bafflestep-corrected summed response (vs. black far field / free field, GPM):

Alrighty.. I hope this quick writeup was at least partially revealing. I hope to be able to translate my paper on the subject to English at some point; but, as we all, I am not made of time, so no promises. I'll for sure link it here if and when I do.

Best Regards

#### dominikz

##### Addicted to Fun and Learning
Forum Donor
Thank you for putting the effort into this writeup.

Since over the last 2 years I have done an extensive empirical study on the subject, comparing (combined) near field & (spliced) farfield measurements to true free field measurements on a total of so far 13 bassreflex enclosures (all kinds; from small bookshelf speaker to 18" PA subwoofer), the results of which I published in an academic article (unfortunately for this thread / forum, in German), allow me to comment.

The method works very well, when executed correctly / appropriately - but nothing about it is trivial.

The nearfield measurement method has been originally documented in Low-Frequency Loudspeaker Assessment by Nearfield Sound Pressure Measurement, D.B. Keele, 1973.

Positioning of the microphone: Keele elaborates on and recommends a position near the center (dustcap). Here's an empirical comparison of mine for a small fullrange speaker, same amplifier voltage, same distance of the mic; dustcap green, outer edge black - we can see a difference, which is mostly minor (~1 dB) over the relevant frequency range:

View attachment 234637

Distance of the microphone to the radiating sources is also very important, as has already been mentioned in this thread. We cannot just "wing it" here - a few cm distances might already produce a loss of several dB SPL.

In their empirical study / paper On Acoustic Very Near Field Measurements, the authors J.Prezelj, P.Lipar, A.Belšak, M.Čudina, state: “To be within 1 dB of the very nearfield pressure, the pressure microphone must be no farther away from the center surface of the piston than 0,11a”.

So, especially for small sound sources, it is vital to place the microphone as close as possible to the source! For bassreflex ports, the mic can be placed flush with it; for speaker membranes, you'll always have to keep at least a few milimeters distance to allow for some excursion, even at low SPLs - but no more than that.

Proper SPL adjustment of the sound sources is also extremely important, which I will go into more detail in later.

Next, let's look at a two practical comparisons vs. free field measurements, when all important variables have already been accurately / appropriately considered, to demonstrate how close of a match is achieveable.

Methodics, hardware / software, environment & further details:

- Measurements are performed with ARTA, 8k FFT @ 48 kHz, 1/24 oct. smoothing (= essentially none), with a Isemcon EMX-7150 microphone (calibration file loaded).
- All measurements are repeated at least once, and a result only accepted as valid if curves match perfectly over multiple runs.
- Measurements are done with low-level input, as to rule out potential compression effects.
- Outdoor measurements are done in a location far from civilisation, on wind-free, quiet nights. As a result, ambient noise is lower than noisefloor of the signal chain (~ -80 dB). There are no obstacles in the vicinity, meaning the measurement is completely free of reflexions.
- Outdoor free field measurement is done as GPM (ground plane measurement). As documented in the original work on the method, Ground Plane Acoustic Measurement of Loudspeaker systems, Gander, 1980, GPM is essentially a 4pi measurement, adding 6 dB SPL for the mirror sound source of the floor.
- GPM is done on a perfectly flat, hard surface.

One potential point of contention for using GPM as a reference free field measurement would be that the mirror sound source of the floor might not only double the sound pressure of the speaker, but also extend the speaker's baffle. This however can be, and in this case was, compensated for by simply adjusting the baffle step correction (doubling the baffle height) in software.

Green curve for all DUTs is the combined near field measurement, SPL adjusted via a empirically derived self-formulated calculation, bafflestep-corrected in ARTA, spliced with a gated far field response on stand / tripod. Black curve is the free field measurement.

DUT #1 - fullrange 12" PA speaker:

View attachment 234643 View attachment 234644 View attachment 234645

View attachment 234651

DUT #2 - fullrange broadband 4" DIY HiFi speaker:

View attachment 234653 View attachment 234655 View attachment 234654

This speaker was more complicated to measure - dustcap measurement didn't make sense due to its large height (too much distance to the cone), and the distance to the rear port had to be accounted for - still, when factoring those variables in, both methods again showed great correlation:

View attachment 234652

One thing to consider is that ARTA's simple bafflestep correction is only really valid in the bass area (up until 100, 150 Hz), and due to gating, the tripod measurement doesn't have much resolution there. So some disagreement in the upper bass / lower mid area is unavoidable. Other softwares with more elaborate baffle step correction will produce even better results.

Also, edge diffraction of the baffle is of course different in a true 4pi measurement vs. GPM, with one edge flush with the ground, which accounts for remaining mid-high frequency differences. VituixCADs diffraction tool might be able to compensate for that too, though, as you can specify a floor / boundary surface.

As we can see, though, bass response of the combined nearfield vs. far field / free field measurements matches almost perfectly - within tenths of a dB! And, rest assured, that goes for the other 11 enclosures / speakers that I have tested as well.

Now, let's quickly get into SPL adjustment of sound sources (port vs. membrane). The known methods are:

- D.B. Keele's original formula (see paper linked in the beginning of this post): 20 log * square root of sd/sv.
- Jeff Bagby's formula (see link in OP): 20 log * membrane diameter / port diameter (or vice versa); no source provided, but this corresponds to the DB Keele formula, taking out the square root for area and switching area for diameter.
- source unknown - method of matching SPL of the curves by hand at a fraction of the frequency of fb, assuming port and membrane to have the exact same output in this frequency region

Worringly, until recently, none of these methods had ever received an extensive empirical evaluation. Sporadic user measurements / comparisons can be observed, but far from enough to claim universal validity.

From my comparisons, I can say these methods mostly produce similar and, depending on expectation, acceptable results / accuracy - mostly within 1-2 dB of the free field measurements.

There are outliers though. The previously covered DUT #2 is such a speaker - the traditional methods for SPL matching produce less favorable results with it. Here's a comparison of in green Keele's / Bagby's formula, and blue SPL matching by hand below fb; left graph port nearfield measurement (in black membrane nearfield response), right graph bafflestep-corrected summed response (vs. black far field / free field, GPM):

View attachment 234677 View attachment 234678

Alrighty.. I hope this quick writeup was at least partially revealing. I hope to be able to translate my paper on the subject to English at some point; but, as we all, I am not made of time, so no promises. I'll for sure link it here if and when I do.

Best Regards
Wow, excelent effort - very interesting!
Thanks!

#### stoneeh

##### Member
Thank you Dominik.

Well, if it's explicitly asked for - here, for the German (speaking) readers: Kombinierte Nahfeldmessung – Praxistest & Neuevaluierung der etablierten Methoden

This is the current version of the publication, which includes measurements of 10 enclosures. As I said, I am currently at 13. There will be an addendum with at least the current 3 more enclosures - but probably more, since I at this point regularly test / measure speakers, and always do the near field measurements as well.

The publication references / is based on some of my earlier work / research, which I'd also recommend reading for a full understanding of every aspect.

I'll quickly comment on the method of publication, because the one I chose is certainly not an usual one: I considered various possibilities. I could directly post to a forum. I could post the .pdf on a personal website. I could join one of the various acoustical societies and engage there (I was actually strongly suggested to do so by a former AES member). Etc.

In the end, I enjoy my anonymity. I enjoy research, and exchanging knowledge with likeminded folks. I have little to no commercial ambitions. But I also in some way want to protect my work / effort. Posting the paper in video form to my already existing (and more or less flourishing) Youtube channel seemed like a logical step - it's freely available, but also not easy to copy / distribute.

I hope it's easy enough to read in this form. You might want to pause the video and manually forward through with the arrow keys of your keyboard. Enjoy, and feel free to get back to me after (here or on Youtube, doesn't matter).

Last edited:

Good job!

OP
N

#### napilopez

##### Major Contributor
Forum Donor
Thank you for putting the effort into this writeup.

Since over the last 2 years I have done an extensive empirical study on the subject, comparing (combined) near field & (spliced) farfield measurements to true free field measurements on a total of so far 13 bassreflex enclosures (all kinds; from small bookshelf speaker to 18" PA subwoofer), the results of which I published in an academic article (unfortunately for this thread / forum, in German), allow me to comment.

The method works very well, when executed correctly / appropriately - but nothing about it is trivial.

The nearfield measurement method has been originally documented in Low-Frequency Loudspeaker Assessment by Nearfield Sound Pressure Measurement, D.B. Keele, 1973.

Positioning of the microphone: Keele elaborates on and recommends a position near the center (dustcap). Here's an empirical comparison of mine for a small fullrange speaker, same amplifier voltage, same distance of the mic; dustcap green, outer edge black - we can see a difference, which is mostly minor (~1 dB) over the relevant frequency range:

View attachment 234637

Distance of the microphone to the radiating sources is also very important, as has already been mentioned in this thread. We cannot just "wing it" here - a few cm distances might already produce a loss of several dB SPL.

In their empirical study / paper On Acoustic Very Near Field Measurements, the authors J.Prezelj, P.Lipar, A.Belšak, M.Čudina, state: “To be within 1 dB of the very nearfield pressure, the pressure microphone must be no farther away from the center surface of the piston than 0,11a”.

So, especially for small sound sources, it is vital to place the microphone as close as possible to the source! For bassreflex ports, the mic can be placed flush with it; for speaker membranes, you'll always have to keep at least a few milimeters distance to allow for some excursion, even at low SPLs - but no more than that.

Proper SPL adjustment of the sound sources is also extremely important, which I will go into more detail in later.

Next, let's look at a two practical comparisons vs. free field measurements, when all important variables have already been accurately / appropriately considered, to demonstrate how close of a match is achieveable.

Methodics, hardware / software, environment & further details:

- Measurements are performed with ARTA, 8k FFT @ 48 kHz, 1/24 oct. smoothing (= essentially none), with a Isemcon EMX-7150 microphone (calibration file loaded).
- All measurements are repeated at least once, and a result only accepted as valid if curves match perfectly over multiple runs.
- Measurements are done with low-level input, as to rule out potential compression effects.
- Outdoor measurements are done in a location far from civilisation, on wind-free, quiet nights. As a result, ambient noise is lower than noisefloor of the signal chain (~ -80 dB). There are no obstacles in the vicinity, meaning the measurement is completely free of reflexions.
- Outdoor free field measurement is done as GPM (ground plane measurement). As documented in the original work on the method, Ground Plane Acoustic Measurement of Loudspeaker systems, Gander, 1980, GPM is essentially a 4pi measurement, adding 6 dB SPL for the mirror sound source of the floor.
- GPM is done on a perfectly flat, hard surface.

One potential point of contention for using GPM as a reference free field measurement would be that the mirror sound source of the floor might not only double the sound pressure of the speaker, but also extend the speaker's baffle. This however can be, and in this case was, compensated for by simply adjusting the baffle step correction (doubling the baffle height) in software.

Green curve for all DUTs is the combined near field measurement, SPL adjusted via a empirically derived self-formulated calculation, bafflestep-corrected in ARTA, spliced with a gated far field response on stand / tripod. Black curve is the free field measurement.

DUT #1 - fullrange 12" PA speaker:

View attachment 234643 View attachment 234644 View attachment 234645

View attachment 234651

DUT #2 - fullrange broadband 4" DIY HiFi speaker:

View attachment 234653 View attachment 234655 View attachment 234654

This speaker was more complicated to measure - dustcap measurement didn't make sense due to its large height (too much distance to the cone), and the distance to the rear port had to be accounted for - still, when factoring those variables in, both methods again showed great correlation:

View attachment 234652

One thing to consider is that ARTA's simple bafflestep correction is only really valid in the bass area (up until 100, 150 Hz), and due to gating, the tripod measurement doesn't have much resolution there. So some disagreement in the upper bass / lower mid area is unavoidable. Other softwares with more elaborate baffle step correction will produce even better results.

Also, edge diffraction of the baffle is of course different in a true 4pi measurement vs. GPM, with one edge flush with the ground, which accounts for remaining mid-high frequency differences. VituixCADs diffraction tool might be able to compensate for that too, though, as you can specify a floor / boundary surface.

As we can see, though, bass response of the combined nearfield vs. far field / free field measurements matches almost perfectly - within tenths of a dB! And, rest assured, that goes for the other 11 enclosures / speakers that I have tested as well.

Now, let's quickly get into SPL adjustment of sound sources (port vs. membrane). The known methods are:

- D.B. Keele's original formula (see paper linked in the beginning of this post): 20 log * square root of sd/sv.
- Jeff Bagby's formula (see link in OP): 20 log * membrane diameter / port diameter (or vice versa); no source provided, but this corresponds to the DB Keele formula, taking out the square root for area and switching area for diameter.
- source unknown - method of matching SPL of the curves by hand at a fraction of the frequency of fb, assuming port and membrane to have the exact same output in this frequency region

Worringly, until recently, none of these methods had ever received an extensive empirical evaluation. Sporadic user measurements / comparisons can be observed, but far from enough to claim universal validity.

From my comparisons, I can say these methods mostly produce similar and, depending on expectation, acceptable results / accuracy - mostly within 1-2 dB of the free field measurements.

There are outliers though. The previously covered DUT #2 is such a speaker - the traditional methods for SPL matching produce less favorable results with it. Here's a comparison of in green Keele's / Bagby's formula, and blue SPL matching by hand below fb; left graph port nearfield measurement (in black membrane nearfield response), right graph bafflestep-corrected summed response (vs. black far field / free field, GPM):

View attachment 234677 View attachment 234678

Alrighty.. I hope this quick writeup was at least partially revealing. I hope to be able to translate my paper on the subject to English at some point; but, as we all, I am not made of time, so no promises. I'll for sure link it here if and when I do.

Best Regards

Thank you so much for sharing your knowledge. I have actually myself been considering how to turn some of my "research" into something more academic and formal so it's nice to see someone has been doing just that. I have not measured many speakers recently but I have wanted to look more into what might cause discrepancies between my quasi-anechoic results and some NFS results.

As you mention in your comparison to GPM, these are generally small depending on your expectation, and arguably negligible in the context of Room modes and SBIR. Still, it would be great to get these results as close as possible.

Looking forward to reading through your paper, with the help of translation =]

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