Mikhail Naganov

LXmini Desktop Version (LXdesktop)—Part III: Binaural Tuning

This is another “technical note” about my experience of building and tuning a pair of desktop speakers with a DSP crossover, based on the original design of the LXmini by Siegfried Linkwitz. This post is about the aspect of tuning which helps to obtain the most natural presentation of the audio scene encoded as a stereo format. As Linkwitz itself explains in this talk “Accurate sound reproduction from two speakers in a living room”, a stereo representation is no more than an illusion which only appears in the brain of the listener. However, this can be a rather realistic illusion. It’s realistic when the listener is able to forget that the sound scene which he or she is hearing is created using the speakers. Ideally, the speakers themselves should not be registered by the auditory brain as the sound sources.

In the Q&A section of the talk, in particular, on this video fragment, somebody is asking Siegfried what is the recommended speaker setup for a small room. And he recommends putting the speakers wider, and to place the listening spot closer to them. That’s in fact what I’ve done in my setup (see one of the previous posts which illustrates the arrangement). The idea behind this setup is to create sort of “giant headphones”—this characteristic is attributed in to the sense of envelopment that this setup can achieve. In fact, the sound of speakers located at some distance is more natural for our auditory brain than the sound from headphones because the sound from the speakers gets filtered by our natural HRTF, thus it’s easier for the brain to decode it properly. However, our perception of sound from these “giant headphones” suffers both from a strong interaction between the speakers themselves—that’s the crosstalk affecting the center image, and between the speakers and the room—this interaction produces reflections and additional reverb not existing on the recording—that’s the “sound of the room.”

The good part is that in my unusual setup the predominant dipole radiation pattern of the speakers is supposed to reduce the crosstalk without resorting to DSP tricks. And as for reflections, they can be filtered out by the auditory brain when they are sufficiently separated from the direct sound, and—that’s another interesting point from Siegfried’s talk—has the spectral content which is similar to the direct sound. The last topic is actually complex and different people have different views on it. However, the cross-talk cancellation is something that can be easily measured.

Cross-talk Cancellation

I have made two types of measurements: one is the usual log sweep which allows recreating the impulse response and window it as necessary, and another kind is a “steady state” measurement produced by taking “infinitely” averaged RTA of a pink noise. Both measurements are made using a “dummy head” technique, so they are binaural. However, since I don’t have a proper head and torso simulator at home, I just use my own head and binaural microphones by The Sound Professionals built with the XLR outputs option so that they can be connected to the same sound card used to drive the speakers. I use REW for these captures, and I have purchased the “multi-mic input” option which is essential for this job since I want to record both the ipsi- and contra-lateral ear inputs at the same time.

The typical way to measure the effectiveness of the cross-talk cancellation (XTC) efficiency is to consider the measurement at the ipsilateral (closer to the speaker) ear and see by how much must it be attenuated in order to obtain the same result as the measurement at the contralateral (farther from the speaker, shadowed by the head) ear. The resulting frequency response curve is the profile of the effective attenuation.

So let’s see. If we look at the steady state response, the XTC from my speaker arrangement is quite modest—around -10 dB in the best case. Below is the spectrum of the attenuation for the left and for the right ear:

However, if we look at the direct sound only, by applying a frequency-dependent window (FDW) of 8 cycles to the log sweep measurement, results look much better, showing consistent attenuation values between -20 and -10 dB. It works better for one ear due to asymmetry of the setup:

Note that deep notches as well as a couple of peaks are due to comb filtering from reflections and the effects of the dipole pattern itself. I must warn that just looking at what seems to the eye as the “average value” on the graph and taking this as the suppression efficiency measure may be self-deceiving. In order to calculate the actual negative gain of the XTC I have measured the difference in RMS level of pink noise filtered via the impulse responses of these natural attenuation filters. The results are somewhat modest: -4.7 dB for the sound of the left speaker and -4.9 dB for the sound of the right speaker.

For comparison of performance with DSP XTC solutions, I have checked the Chapter 5 of the book “Immersive Sound” which talks about BACCH filters. There is a graph of a similar measurement that I have done, they have made it using a Neumann KU-100 dummy head in a real (non-anechoic) room using speakers set up at 60 degrees, 2.5 meters distance from the head, with their filters turned on. The Figure 5.12 of the book presents the measured spectrum at the ipsi- and contra-lateral ears, and similarly they measure the effectiveness of the XTC by subtracting these. I have digitized their graphs and derived the attenuation curve, it is presented on the graph below as the brown solid curve, and I have changed my curves to dashed and dotted lines for readability purposes:

We can see that the BACCH XTC does a better job, except in the region of 7–10 kHz. Also note that since I have a single subwoofer there is no attenuation below 100 Hz. The author of the chapter calculates the level of attenuation as an average of the frequency response curve values across the audible spectrum, and their result is -19.54 dB. However, since I had digitized their graph, I could build a filter which implements it and measure the resulting decrease of the RMS level of ping noise, the same method that I used for my measurements. Measured this way, the effective gain of the BACCH XTC is -8.86 dB. This is still better than my result, but only by 4 dB. So I must admit, DSP can do better job than natural attenuation due to the speaker arrangement and the radiation pattern, however as we can see from the chapter text, building a custom XTC filter tailored to the particular setup is a challenging task, and there are many caveats that need to be taken care of.

Center Image Height

As I have explained in the section “Target Curve Adjustments” of the previous post, in order to provide correct rendering of the center image, the spectrum of the sound from the speakers which are on the sides of the listener must be corrected so that the phantom center image has the spectrum that a real center source would have. The paper by Linkwitz which I cited in that post contains necessary details. One good test for the correction is to make sure that a source which is intended to be at the ears (or eyes) height is actually heard this way. For that, I use the track called “Height Test”—track 46 from the “Chesky Records Jazz Sampler & Audiophile Test Compact Disc, Vol. 2”.

Merging Perceived Results of ITD and ILD Panning

Changing the spectrum of side images in the way described in the previous section also helps to reduce attention to the speakers, because now sounds coming from them do not have the spectral characteristic of a side source.

However, while listening to old engineered recordings (from 70s or earlier) that use “naive” hard panning of instruments entirely to the left or to the right by level adjustment only, I have noticed that this spectrum change is not enough for decoupling the sound of an instrument from the speaker which is playing it. Real acoustic recordings and modern tracks produced with Dolby Atmos sounded better. This was likely because modern panning techniques use both level and delay panning. They may actually use more—to get a full idea of what is possible I used a panning plugin called “PanPot” by GoodHertz.

While playing with a plugin using dry percussion sounds from the “Sound Check” CD produced by Alan Parsons and Stephen Court I have noticed that hard panned sounds using delay panning a perceived a bit “away” from the speakers while level panned sounds are perceived coming from the speaker itself. Schematically, it was perceived like this:

I decided to combine them. In order to move hard ILD panned sounds I use the “Tone Control” plugin, also by GoodHertz. It can do Mid/Side processing, and I switched it to the “Side only” mode. Recall from my previous post on Mid/Side Equalization that M/S decomposition does not completely split out the phantom center from the sides. However, it is good enough to tune hard panned sources.

I have prepared a test track which interleaves pure ILD and ILD+ITD panning of a dry sound of a snare drum. While listening to it, I was experimenting with the settings for the corner frequency, slope, and gain of the treble shelf, as well as with overall gain of the side component. The goal was to move the ILD panned source closer to the position of an ILD+ITD panned source, and at the same time not to change its perceived tonality too much. Obviously the results of panning using different techniques will never sound identically, however, I could come close enough. As a result, the sound scene has moved a bit away from me, behind the speakers plane:

I have pictured the scene as going beyond the speakers because this happens with some well-made recordings like the track 47 “General Image and Resolution Test” from “Chesky Records Jazz Sampler & Audiophile Test Compact Disc, Vol. 2” where the sounds of doors being shut are rendered well beyond the speakers in a distance.

It’s interesting that correction of the purely level-panned images really helped to decouple the speakers from the sound they are producing! I used tracks from the full OST to “The Holdovers” movie which feature a number of records produced in 60s and 70s. Note that as far as I know, the full version with all tracks is only available on vinyl—the usual issue with licensing on streaming services prevents them from offering all the tracks. And the producer of the OST decided not to bother themselves with offering a CD.

Banded Correlation and Decorrelation

Since my speaker system is not a dipole across the entire spectrum, and walls are located nearby, there was still some “unnaturalness” of the image, even though the quasi-anechoic frequency response looks correct. How can we do further tuning without noticeably affecting the frequency response? The trick is that we can change it depending on the correlation.

For example, while listening to the bass line of the “Spanish Harlem”, I have noticed that the first note, which is mostly delivered by the subwoofer does not sound as strong as following notes, which are higher and are delivered by the main speakers. I did not want to raise the level of the sub, because I know it is at the right level, and listening to OSTs by Hans Zimmer proves that—I don’t want the sub to be any louder :). Instead, my solution was to decrease level of the correlated component (the phantom center) in the frequency range served by the woofers—they are omnidirectional, thus their sound is reinforced by the walls. For that I used the “Phantom Center” plugin by Bertom Audio.

Another correlation tweaking needs to be done in the high frequency region, above 4.7 kHz. I took the track I often use—the left / right imaging test from “Chesky Records Jazz Sampler Vol. 1” and overlaid the “midway” position announcement with “off-stage” position. Initial lack of correlation due to somewhat excess of reverberation at high frequencies causes the off-stage announcement to sound either in the front, in the position similar to the “midway” position, or even “inside the head.” By increasing the correlation I was able to move it to the intended location. However, having too much correlation causes the phantom center to become too strong and too narrow, which makes the “midway” position to collapse to close to the center. Thus, by hearing both announcements at the same time I can increase the correlation to the just right amount.

Finally, I used a set of 1/3 octave band-filtered dry mono recordings of percussion instruments converted to stereo: first with identical left and right channels, then with the right channel inverted. This is the same set of sounds that I used in this post about headphones. I compared how loud the correlated version sounds to relative to anti-correlated. It is expected that it should be of the same loudness or a bit louder, however I have found that in the region between 400 and 900 Hz anti-correlated sounds are perceived to be louder than correlated. Unlike my previous experience with traditionally arranged speakers, this time I was able to reduce loudness of anti-correlated sounds in this band.

This perceptual correction helps to reduce attention paid to the details in the sound produced by speakers that get amplified by the room too much. The sound becomes less fatiguing—that’s yet another aspect of “naturalness.” As Linkwitz has put it, it’s better to make our brain to add missing details than trying to force it to remove extra details—that costs much more mental effort which manifests itself in exhaustion resulting from long listening sessions.

Processing Chain

The description of the tuning process has turned out to be a bit lengthy. Let’s summarize it with a scheme of the filters that I put on the input path. They are inserted before the digital crossover and correction filters that I described in the Part II of these post series.

So, first there is the Tone Control applied to the “Side” part of the M/S decomposition, which is intended to move ILD-panned sounds a bit deeper down the virtual scene to match ITD+ILD-panned sounds. Then there go 3 instances of the Phantom Center plugin tuned at different frequency bands that perform the job of correcting the effects of the room-speakers interaction. I wish there was kind of an “equalizer” plugin that could apply phantom center balancing to multiple bands—Bertom Audio, take a note :)

Some Tracks to Listen To

Having achieved a good imaging through my speakers, I had re-listened to many albums that I have pinned down in my collection. Here are some tracks that I can recommend listening to: