With the recent acquisition of the remarkable RME ADI-2 Pro, it’s time to revisit. This time, I’m looking for a simpler way to characterise performance. The early effort involved too many screenshots that were a pain to put into HifiZine articles. Plus, there wasn’t any way to say much about correlation to audible performance.

Introducing JoTT

The test signal I’m using now I call JoTT – short for John’s Torture Test (I had to call it something). I started by combining the CCIF and SMPTE test signals for IMD, and then modified it to make it most useful (for me). It has the following sine wave components:

The peak signal level is -1 dB FS. This avoids odd results that some DACs exhibit at 0 dB output.  (If 0 dB FS is actually required in the test signal, increase A to −6 dB FS or 50%.)

To generate JoTT (and plot the results), I am using the Electroacoustics Toolbox. Here is the spectrum of the generated digital signal (at 96 kHz sample rate):

Use as a test signal

For the sake of an example, I’ll use my Vioelectric HPAV200 with 24/96 coax DAC module (the older one). I’m driving it at the SPDIF input because the measurements of the amplifier alone are better than of the DAC. (For the purposes of explaining the test signal, it’s actually more helpful to use the worst-measuring input.) The output is taken from the left channel of the headphone jack.

I set signal levels so that 0 dB in the graphs is 1V RMS. First, here is the noise floor. There’s a very low amount of mains noise at 50 Hz and multiples of it. Also, there is some high frequency noise – I don’t think this is in the V200 itself, but is coming through ground somehow as I’ve just noticed that it appears in other measurements where I used a single-ended connection.

Here is component A (110 Hz) turned on. The cursors identify its harmonics:

Note: the levels of the harmonics is almost identical if I play just a 1 kHz sine wave at -8 dB. (If I play the 1 kHz sine wave at -1 dB instead, the levels and distribution of harmonics changes. However, you can’t have everything.)

If I now turn on component B (3 kHz), there is a small “forest” of intermodulation products around it. There is also a little thicket at 6 kHz, its second harmonic:

Now I’ll turn off A and B and turn on components C and D (9 and 10 kHz). You can see the intermodulation products at 1, 8 and 11 kHz in particular:

Now I turn on B, and you can see additional intermodulation products at 6, 7, 12 and 13 khz.

Finally, I turn on A again, and you see the “grass” of intermodulation products from around 1 kHz and above. This is the all-in-one graph, which I can use to point to both THD and to IMD in a way that clearly shows when a unit has issues – the worse the IMD, the taller and thicker the “grass” becomes.

Summary. From the single graph just above, we can read:

• The levels of the harmonic distortion components (multiples of 110 Hz).
• The level of 110 Hz sidebands around 3 kHz, and also 9 and 10 kHz.
• The level of the main IMD difference sidebands at 1, 7, 8, 12, 13 kHz.
• The thickness and height of the “grass” i.e. higher-order IMD components.
• Noise level from the mains supply.

Don’t get me wrong – this is good performance! If it “looks bad” that’s because the test is intended to show up problems. Hence the name “torture test.” I have others much worse, which I will may post at a later time…

JoTT is also useful for a quick crosstalk check, because of the multiple tones spread across the spectrum. I connected the right channel from the headphone amplifier and plotted its spectrum, while the signal generator was running into the left channel. You can see that there is some crosstalk, but it’s fairly low:

Reference loopback

Wait! How do I know that I’m not measuring the performance of my audio interface (the RME ADI-2 Pro, in this case)? Well, here is the graph when the analog output of the ADI-2 Pro is connected to its input:

Note: the DAC in the ADI-2 Pro is not being run at -1 dB FS. If I did that, there would be more distortion (although still very low). So this is not an “apples to apples” comparison of the ADI-2 Pro’s DAC with the Vioelectric DAC. Because that’s not what I’m trying to do here. The point is that the ADC of the ADI-2 Pro is being fed the same level of signal, and as shown in the above graph, the amount of distortion it introduces is insignificant (no more than shown, as the plot includes DAC and ADC distortion).

And here is the crosstalk for the ADI-2 Pro loopback:

Summary

What can I say, the ADI-2 Pro is incredible. I can now measure stuff I never could before. I will post some examples of JoTT measurements of other DACs soon, and hopefully the JoTT will appear in future HifiZine articles. If you have any suggestions or comments, please fill in the box below.

I verified in Appendix A of my article The miniDSP nanoAVR – a Case Study that, while there is a measurable difference between the two orientations (pointed at the speaker with the zero-degree calibration file, vs pointed at the ceiling with the 90-degree calibration file), it is small and unlikely to have any effect on equalization.

One of the advantages of purchasing a microphone from Cross-Spectrum Labs is that you get a 90 degree calibration file included. However, you may already have a good calibrated microphone and not particularly feel like buying another. (I understand!) So, this article contains instructions on how to make a 90-degree calibration file. Your prerequisite, of course, is that your microphone already has a 0-degree calibration file. You also need to be able to use Room EQ Wizard (REW).

Background

When you make a measurement with a calibrated microphone, you are not changing anything that the microphone does. All that happens is that the software alters the measured frequency response in accordance with the calibration file. The math, in fact, is simple:

• Actual response = Measured response – calibration file response

The actual response is the “real” frequency response at a point in space (the tip of the microphone capsule). The measured response is what the measurement program will measure at that point in space without using a cal file. And the calibration file response is the magnitude column in the cal file (we’ll see an example below).

Let’s simplify the names:

• A = M – C

and rearrange:

• C = M – A

In other words, to get the 90-degree calibration file response, we only need take the measured response with the mic at 90 degrees, then we subtract the actual response with the mic at 90 degrees. But hang on, aren’t we trying to pull ourselves up by our bootstraps? – the actual response is the reason we are trying to do this in the first place and we don’t have that yet!

The thing is, though, the actual response with the mic at 90 degrees (using the 90 degree cal file) is the same as the actual response with the mic at 0 degrees (using the 0 degree cal file). So we can get A by pointing the mic at the speaker and measuring it using the 0 degree cal file.

Clear as mud? Never mind, just follow the steps below carefully.

Steps

1. Position the microphone pointed directly at a speaker and fairly close (to minimize room reflections). I used a small fullrange driver with the microphone positioned 20 cm away.
2. Load the 0 degree calibration file and take a measurement. Rename it to A (actual response).
3. Reposition the microphone so that it is pointed at 90 degrees. The tip of the mic must be in the same location as before.
4. Clear the calibration file and take a measurement. Rename it to M (measured response). [Note: leave the microphone where it is, as you will need it again later.]
5. Apply 1/3rd octave smoothing to both measurements.
6. View A, and go to File -> Export -> Measurement as text, and save the file A.txt.
7. Open A.txt in a text editor and locate the first line starting with a number greater than 1000. Delete everything before that line. Don’t leave a blank line at the start. Save A.txt.
8. View M, and go to File -> Export -> Measurement as text, and save the file M.txt.
9. Open M.txt and locate the first line starting with a number greater than 1000. Delete everything before that line. Don’t leave a blank line at the start. Save M.txt.
10. Open a spreadsheet (I used Microsoft Excel but others should work fine).
11. Position the cursor in cell A1 and import M.txt. Depending on the program the exact way you do this may vary, but you will typically use options that say it is a text file, with fields delimited by spaces or tabs. It should like a bit look this:
    
12. Position the cursor in cell D1 and import A.txt. Now it should like a bit look this:
    
13. In cell G1, enter the formula =B1-E1 :
     
14. Select all the cells from G1 down to G55, or whichever is the last row with numbers in it. Press Ctrl-D or locate the Fill Down command. The formula gets pasted in each cell.
15. Select all of Column G and do a Copy.
16. Select Column H and do a Paste Special -> Values only. This copies the result of the equation, without copying the equation itself.
17. In cell I1, enter the number 0.
18. Use Fill Down to paste the 0 all the way to the last row with numbers (I55 in my case). The result should look like this:
    
19. Select columns B through G and delete them. You should have three columns left, like this:
    
20. Use Save As.. to save the spreadsheet as a tab-delimited text file, named C.txt.
21. Make a copy of your calibration file, and name it something useful, like serial-90deg.txt.
22. Open serial-90deg.txt in a text editor and locate the first line starting with a number greater than 1000. Delete that line and everything after it.
23. Open C.txt, Select All and Copy its contents, then paste at the end of the open serial-90deg.txt file. Don’t leave a blank line. Save serial-90deg.txt.
24. You now need to verify that the calibration file works as intended. Back in REW, load serial-90deg.txt as a cal file (Preferences->Mic/Meter).
25. Run a measurement, and rename it to V.
26. In the Overlays pane, compare A and V. They should be very close. (If not, something went wrong.)
27. Optionally, you can use File -> Import Frequency Response to import serial-90deg.txt to see how it looks. (Set vertical limits to say +/- 10 dB.)

Example 1: CSL-calibrated Dayton EMM-6

I use the CSL-calibrated EMM-6 as my first example, because I already have a 90-degree cal file from Cross-Spectrum Labs for it. That way I can see how the cal file I came up with compares.

Here’s my A in blue and M in red (1/3rd octave smoothing):

Here’s my A again in blue and V in purple:

Good!!

Here is the comparison of the 90-degree cal file from Cross-Spectrum Labs in blue with the cal file that I just generated in purple:

Example 2: miniDSP UMIK-1

I don’t have a 90-degree cal file for the miniDSP UMIK-1, so I will generate one and verify it as described in steps 24-27.

Here’s A in blue and M in red (1/3rd octave smoothing):

Here’s A in blue and V in purple:

Here, out of curiosity, is the miniDSP 0 degree cal file in red and the generated 90 degree cal file in purple:

Additional notes

The procedure given above seems long and complicated, but it’s one of those things that takes longer to describe than to actually do. I tried to find a simple way of doing it, but this is the best I’ve found using readily-available tools (and no coding).

1. The procedure uses the existing data from the cal file below 1 kHz, and the new data above 1 khz. This is because the mic response due to orientation varies only at high frequencies. In addition, it’s hard (or impossible) to get repeatably accurate measurements at low frequencies because of ambient noise (traffic, planes, wind, footsteps, …)
2. A full range driver is best to use, as it avoids any directionality effect that might occur if positioning the microphone in front of a speaker with multiple drivers. If you don’t have a full-range driver, try:
   1. Moving the switchover frequency up. You may get a bit of a glitch though.
   2. Moving the mic further away (e.g. 50cm instead of 20 cm). This may give a slightly less accurate result due to room reflections but will probably be just fine.
3. I created column I for the phase data. However, most measurement programs will probably be fine without it.
4. The two times I’ve done this so far, there is a small glitch at 1 khz in the cal file. This could be avoided by matching A and M exactly at 1 kHz prior to exporting. However, the glitch I got was only around 0.1 dB so I decided it wasn’t worth the effort.
5. The impulse response of A and V does look a little different. I’m not sure at this point whether there’s ever a situation where this might matter.

Please let me know in comments below if you try this out and how you went. Thanks!

Postscript

You can, of course, come up with a method based on the above to calibrate one microphone based on another – but instead of rotating the mic 90 degrees, you measure using the second microphone. In order to get good low-frequency readings, the measurements should be done in two halves, as described here, and spliced together.

Note: these measurements are not intended to be authoritative or definitive. I took them to satisfy my own curiosity. If you know of measurements that are consistent with or contradict these measurements, please do post a link to them in a comment below.

Preliminaries: measurements in REW

I first tested three microphones with REW. These were all done in-room, approximately at a suitable listening position, since the iPad measurements will be simple 1/3rd-octave RTA. All microphones were pointed directly at the speaker and the appropriate (on-axis) calibration file used.

Firstly, a comparison of measurement sweeps using an Earthworks M30 vs my CSL-calibrated Dayton EMM-6 microphone:

REW Sweep, Earthworks M30 (red) vs CSL-calibrated EMM-6 (blue), 1/6th octave smoothing

The  two mics are very consistent across the range, with the EMM-6 reading slightly lower from 6 to 17 kHz. I’m not 100% convinced that the M30 is quite on the mark, for reasons I won’t go into here, but regardless, pretty close.

Next, a measurement sweep comparing the M30 to a miniDSP UMIK-1:

REW Sweep, Earthworks M30 (red) vs miniDSP UMIK-1 (green), 1/6th octave smoothing

The UMIK-1 reads slightly higher than the M30 below 50 Hz and above 8 kHz.

Now, I mentioned above that the iPad measurements will be an RTA (real time analysis), not a measurement sweep. Here’s how the REW RTA function (1/3rd octave) compared to the sweep, both with the UMIK-1:

Measurement sweep in REW (green) vs RTA in REW (red)

They’re pretty close, with the RTA drooping slightly at the top and bottom (the same effect was observed with the other two microphones). Note that the RTA does tend to bounce around a little, so some variation at individual 1/3 octave bands should be ignored.

iPad measurements

The measurements on the iPad were done using the RTA function of AudioTools. To compare measurements, the RTA was stopped and saved as a file, which was then moved to my laptop and imported into REW. The actual display in AudioTools is a bargraph, but they come out as smooth curves in REW. Good enough to compare though.

I was unable to use the EMM-6 and the M30 with the iPad as my MOTU Microbook II USB audio interface doesn’t work with the iPad. Bummer. I guess it saved me some time. So I used the UMIK-1 (with the Apple Lightning-USB adapter), a Dayton iMM-6, and the internal iPad mic. The UMIK-1 and iMM-6 had their respective calibration files loaded, while the internal mic used the AudioTools default (which apparently didn’t do much, see below).

First off, here’s the RTA measurement using the UMIK-1 taken on REW, compared with the RTA measurement using the UMIK-1 on the iPad:

RTA from REW (red) vs RTA from AudioTools (purple), both using UMIK-1

Pretty close, although there seems to be some drop-off in the AudioTools measurement below 90 Hz and I don’t think I’d trust the two lowest bands (i.e. below 30 Hz).

How does the AudioTools measurement compared in absolute terms? In other words, let’s assume that the sweep is the most accurate measurement. Here’s the sweep taken in REW using the UMIK-1, compared to the RTA measurement in AudioTools:

Measurement sweep in REW (green) vs RTA in AudioTools (purple), both using UMIK-1

Oooo… the AudioTools RTA reads about 3 dB lower from 70 Hz downward. Above 90 Hz, it’s as close as you could want. I can’t really explain this – if I can get the other mics working with the iPad at some point, I’ll see if I can verify whether this effect is consistent or just an anomalous measurement.

How does the internal mic compare? Here is the RTA from AudioTools using the UMIK-1 compared to the RTA using the internal mic:

Audiotools RTA using UMIK-1 (purple) vs the internal iPad mic (green)

As you can see, a fair bit of a drop below 200 Hz and above 6 kHz.

The little Dayton iMM-6, though, comes with a calibration file. Here it is compared to the UMIK-1:

Audiotools RTA using UMIK-1 (purple) vs the Dayton iMM-6 (blue)

It wobbles around a bit but is generally a decent match, with the exception of the 100-400 Hz range where it varies more than I’d be comfortable with for serious measurements or EQ. (I repeated the measurement and got a similar result.)

Cal file conundrum

After I took the RTA with the iMM-6, it occurred to me to check the calibration file. Loaded into REW, it looks like this:

Dayton iMM-6 calibration file

See all those wobbles above 4 kHz? They really shouldn’t be there. Dayton (or rather, their supplier) are doing something funny with the calibration. Still, for 1/3 octave RTAs, it makes no practical difference, but I wouldn’t use these cal files for measurement sweeps without smoothing. Here’s the Dayton cal file for my EMM-6 compared to the cal file from Cross-Spectrum Labs:

EMM-6 calibration files: from Dayton (green) and from Cross-Spectrum Labs(blue)

(In case it’s not obvious, the Dayton cal file simply can’t be correct.)

Conclusions/Recommendations

Based on the above measurements, here is what I think…

1. The internal microphone in the iPad isn’t really suitable for measurement work. It is probably possible to generate a calibration file that would make it work much better for this kind of RTA, at least above 40 Hz, but the default setting in AudioTools wasn’t it.
2. The little iMM-6 performed OK. If you’re just looking at the general trend and not trying to EQ something to the last dB, this is great value at $17. Or, it would be a great tool for learning the ropes without spending very much. If it weren’t for the funny response in the 100-400 Hz region, it would be a complete no-brainer at that price.
3. The external USB mic, the UMIK-1, performed consistently with the measurements from REW albeit with some droop below around 90 Hz. I can’t explain that but I’d assume that it’s the software not the mic itself. In either case, something to keep in mind if doing EQ. The Dayton UMM-6 would I assume be a good alternative, except for…
4. The Dayton cal files. If spending a bit more for a USB measurement mic (and the needed USB adapter for the iPad), then I would recommend getting the mic from Cross-Spectrum Labs. (Actually, I’d recommend that for the UMIK-1 as well as the UMM-6 – for a slight increase in cost over the stock mic you also get a 90-degree calibration file, which is more than a little handy if you’re trying to EQ an HT system i.e. with speakers all around you).
5. I don’t have a USB audio interface (i.e. with mic preamps) compatible with the iPad. For some reason I had assumed that the Microbook II would work, until I tried it and remembered that it does require a driver even on the Mac. Any recommendations on an iPad-compatible interface? Please post a link below if so.
6. iPad software is more limited than measurement programs running on a regular computer. It wins on convenience and portability but not on features or analysis capabilities. Nothing wrong with that, just be realistic.

Having now ordered and received some waveguides from Dave Pellegrene, I wanted to revisit the design of the Convertible. Based on the measurements of the SB29RDC in the waveguide and some other considerations, I’ve switched to a 6.5″ woofer, the Seas U18RNX/P. Here’s the concept design for a “convertible” monitor using the U18 and the Pellegrene waveguide:

Earlier, I had always assumed that the ported base of the “convertible” speaker would only add air volume (and a port). However, I recently had one of those “Aha!” moments, and with a little further investigation realized that a much better approach would be to add an extra woofer as well, thus turning it into a “2.5 way” ported floorstanding speaker. Here’s the concept diagram of that configuration:

Why the switch from 8″ to 6.5″? First, take a look at the SB29 response in the waveguide (0 30, 60, 90 degrees):

And here is Seas’ published U18 response:

If you look at the dispersion shown by the 60-degree curves (orange in the top graph, bottom curve in the lower graph), the directivity of the woofer gradually increases above a few hundred Hz and then matches the waveguided tweeter around 2 kHz. It’s almost as if these two were designed for each other!

The other thing about the U18 is that it has a nice little bit of extra Xmax, at 6 mm (peak). In fact, it has almost the same volume displacement as the U22 that I wanted to use in the original proposal. The net result is that I think two U18’s, while a bit more expensive, are a more effective solution than a single U22, for this speaker. And why not use two 8’s? Because the ported box would be too large.

I’m quite excited by this design (at this concept stage anyway). Everything – size, output, plate amp power, etc – just all seems to fit together really well. But first, I’ll try to finish off the Mini Convertible series of articles, translating this same concept to the “mini” format.

To mount the waveguide, you remove the three screws from the tweeter faceplate, insert the provided studs into the tweeter, and use the provided nuts and washers to bolt on the waveguide.

With a very quick and rough measurement, with the tweeter held by hand about 15cm away from the microphone, here’s the tweeter with the standard faceplate, at 0, 30, 60 and 90 degrees (ignore vertical scale except for relative reference):

And here it is mounted in the waveguide:

So that looks very promising!

Mini Convertible at Bathurst

As you can see, I made the second prototype with the tweeter offset in the baffle, so I can measure the effect it has on baffle diffraction. (Haven’t done the measurements yet.)

Some of the attendees:

Attendees at Bathurst GtG 2013

Terry, by the way, has an awesome DEQX-based system, which I’ve been meaning to try doing some sort of write up on for HifiZine. One day.

Photo Credits: Russ Tunny.

I never thought I’d get excited about a plate amp.

But, while waiting for the drivers for the Mini Convertible active speaker project, I was wondering how best to add a subwoofer to the sealed/monitor version of the speaker. After working through a couple of options, the completely obvious hit me: the PWR-ICE125 plate amp provides a buffered copy of its digital input signal, so there’s no reason that you can’t chain a subwoofer built with a third PWR-ICE125 amp, like this:

In this setup, the amp in the subwoofer is set to sum both left and right channels, and to run in BTL (bridged) mode. A couple of observations here:

• There’s no reason to stop at one subwoofer – as far as I can tell, you can chain as many more as you like. It’s not hard to envisage a distributed multi-sub system built using these, with additional subs being added whenever funds allowed or the need arose.
• The digital source needs to be volume-controlled. While each PWR-ICE125 does have a volume control, it applies only to the amp that it’s on and doesn’t affect amps further down the chain. (This is as it should be: the digital link out is a buffered copy of the input, no reclocking, sample-rate conversion, or other processing.)

What would be a good driver for this? One good-looking option is the Dayton RSS315HFA-8 12″ subwoofer driver. At the rated 250W (into 8 ohms) of the amp in BTL mode, this is its simulated output (half space, anechoic) in an 84 liter (3 cu ft) sealed box:

Maximum output is 98 dB at 20 Hz (excursion limited) and 106 dB at 40 Hz (power limited), with an f3 around 36 Hz. Why an 8 ohm driver? While the amp is rated to operate into a 4 ohm load in BTL mode, the continuous power rating into this load is much less. Personally, I just feel more comfortable with a more conservative 8 ohm driver in this situation.

So far so good. What if you wanted a smaller box? Here’s another option: use a smaller driver, but use two boxes. Eh what? Let’s look at this option: build the plate amp into one subwoofer box, and also build a second “passive” subwoofer box. Then we run the plate amp in “stereo” mode instead of BTL mode. It looks like this:

I mentioned above that the digital source has to have a volume control, but if it hasn’t, this can be done by putting a miniDSP OpenDRC-DI at the front of the chain, which I’ve shown in the above diagram. Once you have one of those in the chain, you can use it to linearize the phase of the whole system, with the aid of rePhase.

What about output levels – do we lose or gain anything? Here’s an example, a pair of Dayton RSS265HF-4, each in its own 28 liter (1 cu ft) box:

Maximum power output is 98 dB at 20 Hz, and 106 dB at 40 Hz, both power-limited. So, the same as the single 12″… f3 is about 46 Hz… but you can easily alter this with a Linkwitz transform. We have incurred the cost of two 10″ drivers now instead of a single 12″, so what did we gain?

• Each box is much smaller – 28 vs 84 litres (1 vs 3 cu ft) – and therefore likely to be easier to place.
• With two subs, we increase the number of bass sources, which can (if done carefully) help produce a smoother (pre-EQ) response.

While it certainly wouldn’t hurt if the amp had more power – say twice as much – for an extra 3 dB maximum output above around 30 Hz, this is still enough for most home systems, and likely to be suited for situations where space is at a premium. Of course, you can play numbers games like these all day, but in the end the builder has to choose what suits them best. Horses for courses, as they say… but still, this is all pretty cool.

Of course, smaller speakers have less output and less bass. That is unavoidable. But, they may be easier for many people to build and/or more likely to find domestic acceptance. The goal here is to come up with a basic, documented starting point, with room for individual builders to experiment and share their construction notes and ideas. The ability to store multiple DSP configurations within the amps will I hope encourage experimentation and sharing of DSP configurations between builders.

I’ll start with the cabinet concept. I wanted to come up with something that can be used in either a sealed or ported configuration. The sealed version is a compact monitor, with the plate amp mounted on the back. The base plate of the monitor can be removed and replaced with a stand. This stand is hollow – thus making the cabinet larger – and has a port near the floor. Hey presto! A ported floor-standing loudspeaker.

The diagram below illustrates the concept with a cross-section through the speaker. The sealed monitor is on the left and the ported floor-stander is on the right. The plate amp has its own sub-enclosure to seal it from vibration within the cabinet.

What of the drivers? For the woofer, I’ve decided to use the Seas U16RCY/P, which has a frame diameter about that of a typical 5.5″ driver but a cone area about 25% higher (99 cm²):

In a sealed box, f3 is in the eighties, making it ideal for crossing to a subwoofer or for use in a small HT system. A Linkwitz transform in the amp can of course be used to drop the f3, subject to power and excursion limitations – this should work well in smaller rooms or on a bookshelf, even without a subwoofer. This driver could even be used in a box as small as 4 liters, although larger would be preferable.

With the ported base/stand attached, the low-frequency response depends on the total internal volume and the box/port tuning. For example, a volume of 14 litres gives an f3 of 45 Hz and an f10 of 33 Hz, while 22 litres with an EBS alignment gives an f3 of 35 Hz and an f10 around 28 Hz – albeit with slightly lower output capability.

For the tweeter, I’ve decided to use the Seas 27TBCD/GB-DXT, which has an unusual faceplate that helps control dispersion so that it should match well with the chosen woofer:

With these drivers, the smallest possible baffle size (using 18mm material to build) would be around 160×280 mm. Leaving a little room to breath and to allow for a roundover or chamfer, let’s call it 190×300 mm (7.5×11.8 in) for the baffle and around 300 mm deep including the sub-enclosure for the plate amp. The ported version would be set up so that the speaker is about 1 m (39″) tall.

Pricing works out as follows (driver prices as listed on Madisound):

• miniDSP PWR-ICE125 plate amp x 2: $550
• Seas U16RCY/P woofer x 2: $190
• Seas 27TBCD/GB-DXT tweeter x 2: $124
• Wood and finishing: $50+

Realistically, total build cost should work out around $1100 (USD, a bit more in Australia and other countries) including all parts and supplies, shipping, and a reasonable level of finish. Because of its size, it won’t be a “bass monster” or suitable for high output levels, but other than these natural limitations I’m expecting that this will be a neutral, easy-to-place, and very flexible/customizable little speaker.

• The Mini Convertible, a 6″ two-way active loudspeaker project
• The Convertible active loudspeaker – revisited

Introduction

I’ve wondered for a while if it might make sense to publish some “HifiZine projects” – fully-documented DIY projects that supplement the many tutorials in the Technical and DIY section here on HifiZine. For me, any such project would most likely be in the area of DSP-based active loudspeakers, with a documented construction guide and a provided DSP configuration file. The idea would be that those tackling their first project would be able to build a DSP-based active speaker with low risk. Then, once they had learned the basics of DSP and speaker design, they could proceed to refine and tweak the project as they wished.

This idea hadn’t solidified into anything concrete until recently. There were many questions. For example, the speaker configuration: should it be a 2-way? A 3-way? Integrated subs? MTM? Open-baffle, boxed, or a hybrid? And which drivers? The choices are almost endless, so some firm selection criteria would need to be laid down before even starting. And the electronics…. with the need to buy multiple boxes or cards from different sources, I didn’t see a clear way to crystallize an easily-obtainable, “universal” solution. Finally, the whole lot had to be reasonably affordable but still offer great performance for the money.

The catalyst to resolve the dilemma came somewhat unexpectedly. miniDSP recently announced a two-way plate amp that incorporates their eponymous DSP functionality together with a B&O ICEPower module. While I generally prefer separate amps because of the flexibility to “reuse and recombine” into different projects, this new miniDSP plate amp does solve all of the problems of the electronics for this type of project: easy to build (nothing to build at all, in fact), easy to source, and affordable.

The plate amp would constrain the speaker to be a two-way. Constraints are good: they reduce the size of the decision space and enable forward progress to a solution. The two-way requirement also counts out open baffle speakers – while it’s certainly possible to make a two-way OB (and I’m working on one with the Involve/ER Audio mini panels), it’s trickier to get a good result and any such speaker will be of interest to a smaller niche of listeners/DIYers.

If it’s going to be a boxed speaker, though, I’d prefer a “uniform directivity” design – that is, a smoothly-varying off-axis response without the off-axis tweeter “flare” that is common to two-way designs. That immediately suggests a waveguide or horn on the tweeter. While waveguides used with compression drivers have gotten a lot of attention in recent years in the Econowave project, I didn’t feel that a digital Econowave clone would be suitable as a HifiZine project. (If you’re interested, though, check out “gainphile”s nicely-documented DSP Econowave.)

Tweeter

I decided that my preference would be an efficient dome or planar tweeter with a waveguide. The Vapor Audio Aurora is an example of such a speaker. The waveguide serves to limit dispersion of the tweeter at the lower end of its range and increase efficiency – or conversely, reduce distortion for the same output level.

But, designing my own waveguide – never mind machining it into the baffle like that – is not something I’m prepared to tackle. It would also work against the idea of a project based on readily-sourced components. I needed an off-the-shelf solution! Fortunately, one came to light, the Monacor DT-300 tweeter and matching WG-300 waveguide. I found a couple of favorable comments online, detailed measurements by Kimmo Saunista here, and use in a project by Troels Gravesen here (although modified, as he often does). Kimmo’s measurements indicate good distortion performance and a possible crossover point somewhere around 1.7 kHz.

Other tweeter/waveguide options

Since writing this proposal, I’ve been keeping an eye out for other tweeter options for this project. Here are some:

• Pellegrene Acoustics has waveguides for a number of tweeters, including the Vifa XT25, SB Acoustics SB29, and Transducer Labs NC29.
• Creative Sound Solutions has the Planar2 horn-loaded ribbon. There are no published measurements, but it does look a lot like the driver used in the Soundfield Audio Monitor 2 (click link for measured off-axis response).
• John Krutke reports a good result with the SB29RDCN-C000-4 in a modified PE 8″ waveguide (search for “February 5, 2010”). The downside here of course is the need to modify the waveguide.

Woofer

An 8″ woofer seems most likely to match the directivity of the tweeter/waveguide combination at the crossover frequency. I worked through the online catalogs of Madisound and Parts Express to find a readily-available 8″ woofer that meets the following criteria, roughly in decreasing order of priority:

1. Smooth response through the midrange i.e. suitable for a 2-way
2. Sufficient bass extension and output to be usable without a subwoofer
3. Reasonably high sensitivity to match the tweeter as closely as possible
4. Reasonable price
5. 4 ohm impedance to get the most power from the ICE module

In the end, I chose the Seas U22REX/P-SL, which matches all of the criteria except the last. At around $150, it’s a bit more expensive than I would have liked for this project, but relative to total project cost, not excessively so. This fairly new driver is the lower midrange in the Linkwitz LX521 but its parameters and response curve also make it well-suited as the woofer in a two-way. The (nominal) 8-ohm impedance means that the amp module will deliver only (a nominal) 60W into the woofer, but the sensitivity of the driver makes up for it: peak output levels are adequate at 108 dB (at 1 m), or around 105-6 dB once baffle step compensation is included. At these levels, all of the drivers I looked at are excursion limited below 100 Hz or so anyway.

The final piece of the puzzle is the cabinet design. The woofer is flexible enough to be used in either a sealed or ported enclosure. I decided to do both. The main speaker module will be a heavily-braced monitor of around 265mm wide, 320mm deep, and 440mm tall (10.5×12.5×17.5″). This is a sealed box with the plate amp mounted on it and an internal volume of around 20 liters. In this configuration, the speaker has an f3 of around 80 Hz, making it perfect for crossing over to a subwoofer or for use in an HT system. (And, of course, with DSP you can use a Linkwitz transform to make it any f3 and Q that you want, within power and excursion limits.)

The base of the monitor will be removable so that it can be replaced by a stand that turns the speaker into a ported enclosure. This opens up the interesting possibility of having different alignments by swapping out stands. An alignment fairly close to the standard QB3 will be best for rock music; a lower tuning frequency will give an underdamped alignment for a tighter bass; and a larger enclosure with an EBS (extended bass shelf) alignment but lower power handling will likely be favoured by classical music lovers. (Scale below goes from 20 Hz to 200 Hz).

Summary

The total cost of this project starts at around $1000 per pair, plus shipping costs, plus whatever you want to spend on cabinet materials and finishing. So realistically let’s say around $1200-$1400 per pair:

• miniDSP plate amp x 2: $550
• Seas U22REX/P-SL x 2: $296
• Monacor DT-300 and WG-300 x2: $100-$150
• Wood, wire, and finishing: $50+

The result should be a speaker with good dynamics, the characteristic clarity of active speakers, and very good top-to-bottom performance without the typical compromises of a two-way.

The conventional method of “calibrating” the SPL (sound pressure level) in Room EQ Wizard is to use an external sound level meter (SLM). This procedure, documented in the REW online manual, basically means playing a signal through the speakers, measuring the result with a separate SLM, and telling REW what the reading is, so it can “match it up” with the electrical signal being received from your measurement microphone via your soundcard / audio interface.

The accuracy of this method depends largely on the accuracy of the external SLM, and to some extent on your ability to put a measurement microphone and an SLM at the same point in space. With a cheap SLM, in addition to the basic level being a bit off, the varying frequency response (see last graph of linked article) may lead to a different reading depending on the nature of the measurement signal. But in any case, it’s good enough to get a “ball-park” reading that is adequate for most in-room and speaker measurement purposes.

If, however, you have a microphone with a calibrated sensitivity reading, you can do the calibration without needing an external SLM. You will also need an inexpensive digital voltmeter (DVM).

What is a “calibrated sensitivity reading”? Just a conversion factor between sound pressure level and the voltage generated by a measurement microphone as a result. A commonly-used SPL for this purpose is 94 dB, which is equal to 1 Pascal (Pa) of pressure variation (super-imposed on the “DC” atmospheric pressure). Here is the conversion factor that came with my “Premium+”  calibrated EMM-6 microphone from Cross-Spectrum Labs:

In other words, at an SPL of 94 dB, the microphone produces 11 mV of output. Great (if you have one)! But, how do you use it? Actually, it’s simple. (And to be honest, it took me nearly a year to realise how simple it is, so I felt it’s worth writing down for others as well.) In addition to the calibrated microphone, you will need:

• A loopback cable. If you’ve followed the REW documentation on how to calibrate your soundcard (for frequency response), you should already have one of these. For most common soundcards, the two ends will look like this (XLR on left, TRS on right):
• A digital voltmeter (DVM). You don’t need an expensive (never mind calibrated$$!) one here, just a fairly basic one will do fine, as long as it has an AC voltage range of 2 V. Even cheap DVMs have an accuracy of at worst a few percent – bear in mind that even if the DVM is 10% out, your SPL readings will only be off by just over a dB.

So here’s the step-by-step. This procedure is designed to gave reasonable readings with inexpensive equipment, and should work with most hardware. All of this assumes that you are already able to use REW to make measurements, that you have a microphone calibrated for frequency response, and that you have loaded the mic calibration file; if not, consult the REW online help and the REW forum before attempting the following.

1. Plug the TRS end of the loopback cable into your soundcard output. Set the DVM range to its 2 VAC range, and connect it to pins 2 and 3 of the XLR end of the cable. Clipleads are very handy here:
2. Open the REW signal generator, set it to generate a 1 kHz (see Note 1) sinewave at -3.0 dB (full output), and turn it on:
   
3. Adjust the output level of your soundcard until your DVM reads exactly (as close as you can get) 100 times the 94 dB sensitivity reading. So with the 11 mV/Pa sensitivity figure shown above for my EMM-6 (each individual mic may be different), the DVM must read 100 x 11 mv = 1100 mV = 1.1 V.
   
4. Now, we set REW to generate 1/100 of that voltage, by changing the RMS level in the REW signal generator to –43.0 dB. The voltage on the soundcard output will now be the same as the voltage generated by your microphone when the SPL on it is 94 dB.:
5. Remove the clipleads, turn off phantom power (see Note 2), and plug the XLR end of the loopback cable into your soundcard’s mic input.
6. Open the REW SPL meter, and look at the level meter labeled “dB FS” in the bottom half of the window. Adjust the input gain of your soundcard until the level reads fairly high, but not higher than -3 dB. Let’s say -5 dB (see Note 3). Like this:
   
7. Now click on the Calibrate button. The default value that comes up should be 94, but if not enter it, then click on Finished. You will get a notice that the maximum level that can be measured is 99 dB (with the level set to -5 dB).
   
8. Turn off the signal generator. Unplug the loopback cable. Turn on phantom power, and connect your microphone to the mic input. Connect your audio or HT system to the soundcard output as you normally do for measurements, and then turn your system on.

You should now be able to set your output levels and make a measurement. Note that you can change the output level at any time to suit: in REW itself, with your soundcard’s output gain control, or with your system’s master volume control. However, you must not change the gain of the mic input of your soundcard, or your SPL readings will no longer be accurate (and you will have to redo the above procedure).

Some notes, referenced from the above steps:

1. I’ve read that cheap DVMs are most accurate at mains frequency i.e. 50 or 60 Hz. I’m not sure how relevant this is, as mine is only a fraction of a percent different between 50 Hz and 1 kHz. At any rate, you can easily check if your meter reads differently at 50 (or 60) Hz and 1000 Hz by changing the frequency in the REW signal generator.
2. Your soundcard is probably safe with 48 VDC phantom power on its output, and I have inadvertently left phantom on while making a loopback connection and not damaged anything. However, it’s much safer  to make sure that phantom power is turned off, as it is possible that 48 VDC on the outputs of a solid state circuit will severely damage it.
3. This setting is not critical, as long as it’s no higher than -3 dB. It determines the amount of headroom in your measurement. If you measure at the recommended level of 75 dB SPL, the setting in step 6 should generally give enough headroom. If REW warns that you are clipping a measurement, then you will need to either reduce your SPL or redo this step with the level set lower – like say -10 or even -20 dB.

A note on accuracy

Some caution is needed with regard to accuracy of measurements like this. It would not be reasonable to expect that the above technique will give you fractions of a dB accuracy in your SPL measurements. I suspect that many hobbyists are unrealistic in these regards, and don’t realize how expensive and difficult it is to get SPL measurement accuracy to sub-dB tolerance levels.

In the realm of a typical hobbyist, the sensitivity figure in a microphone calibration from Cross-Spectrum Labs is in the range of +/- 1 to 1.5 dB. You must also remember that your microphone’s output will vary with temperature and humidity, and your DVM also has a tolerance. I think it’s reasonable to state a tolerance when using this method of +/- 2 dB. Again, if you need higher tolerance, you’ll need to be prepared to pay a lot more for it.

Beware also of mic calibration files that are not 0 dB at 1 kHz. Earthworks, for example, provide a calibration that is relative to a reference microphone, rather than treating the 1 kHz reading of the mic-under-test as 0 dB. In cases like this, you’ll have to adjust accordingly.

Note also that I haven’t yet compared results obtained using this method with a calibrated SLM. There are a couple of ways in which an error of 3 dB (difference between RMS and peak) or 6 dB (difference between balanced and single-ended) could be introduced. I’m fairly sure that this method works correctly but until I am able to compare the result obtained this way with a calibrated SLM, I am only going to say that it “probably works.”

Please report in a comment below if you have used this method and compared it to a calibrated SLM. Thank you!