Monday, April 9, 2018

Peter Millet's Soundcard Interface, Part 2

Lots of things happened while assembling my Millet's Soundcard Interface. First, I discovered that the voltage mode switch can be restricted to the required 4 positions by inserting small metal bits that came with it into the holes around the shaft. This is where they need to go:

If you misplace the bit, use a magnet to pull it out. After inserting the bits, cover the front side of the switch with the metallic sticker that came with it.

While finding the right places for the bits I was turning the unit on and off frequently and managed to break the power switch. I had to solder the switch out—tough task, as it has 8 pins in total, and then solder back a replacement. Fortunately, the board continued to work as before.

By this time, Mumetal sheets have arrived and I started covering the inside of the box with them. As a reminder, Mumetal is an alloy that offers shielding from magnetic fields. There was one problem—there are round mounting studs on the inside of the box so I had to make holes in Mumetal. Doing that with a drill has turned out to be impossible—the metal is very thin, and the drill was just tearing it.

I've found that the diameter of the mounting studs is the same as in a standard paper puncher. However, the holes had to be done very far from the edges. So I had to cut the Mumetal sheet into smaller pieces:

This, of course had made the coverage to be non-continuous. In order to restore it, I glued strips of Mumetal over the stitches, and then made solder connections between the strips and the sheets under them (I used an old style lead solder for that.)

This way, I've got a solid shield with close to zero resistance which made an excellent chassis ground. One thing that was left to do is to connect the board's signal ground to it. As a connection point, I've chosen one of the outbound ground legs of the Murata DC/DC converter—so the connection is as close to the power input as possible. It is also convenient that the converter is on the edge of the board, so I could use a very short wire between the it and the side wall of the box:

I've checked that there is indeed a low resistance path between the balanced input ground pins and the Mumetal walls. Note that all this shielding didn't help to bring down the "zero level" of the device as I was hoping.

Sunday, March 25, 2018

Peter Millet's Soundcard Interface

From excellent series of articles by Stuart Yaniger "Practical Test & Measurement" I've learned about Peter Millet's Soundcard Interface device. In short, this is an analog adapter for transforming voltages in the range from 20 millivolts to 200 volts into the range accepted by soundcard's line input—about 1 volt. The device also offers protection for the soundcard inputs from accidental high voltages.

The application of this device is for testing audio amplifiers. Below is a general connection scheme:

The amplifier under test is connected to a load. The load can be either real speaker or a dummy load—a resistor, for example. The soundcard interface is connected in parallel to the load. Since the interface's input impedance is about 100 kOhms, it draws only little current by itself (for a comparison, typical oscilloscope input has 1 MOhm impedance.) The output from the soundcard interface goes into a soundcard, which is connected to a PC (or Mac) running measurement software.

The input to the amplifier is provided directly from the soundcard's line output. Though if it's a mobile phone with an analog output or a portable audio player, the test signal can be played from the device itself.

The Good

What is so good about the Mr. Millet's device? First, you must build it yourself, which is fun. Second, it's cheap compared to "pro" measurement frontends. Let's see:
I've spent $265 (shipping excluded) on the parts for the Soundcard Interface. If you are in electronics DIY hobby seriously, you may have some parts already or buy them in larger quantities, which will take the price even lower. Sure, the Millet's device plus a soundcard may not deliver the same precision of measurements as those frontends, but for home projects it's enough.

The third good thing about this interface is that it doesn't require any specialized software. All the "pro" interfaces mentioned above come with dedicated software which you may or may not find easy to use. But for soundcard measurements, there are lots of audio analyzer software tools, some of them are even available for free. So it's possible to compare and choose what better suits your needs.

Speaking about alternatives, I must also mention recently emerged Jan Didden's L|A Autoranger which from its description seems to be similar to the Millet's interface, and has a comparable price. One great thing about the autoranger is that it automatically adjusts itself to the input signal level, like "pro" measurement interfaces do. Sounds interesting, I will perhaps try it some time later.

The Bad

So far, I have only soldered up the device and calibrated its zero and full scale levels.

One really important issue that must be addressed is the level of noise produced by the device itself. While having the PCB lying open on my table, I couldn't get the "zero" level lower than 0.9 mV (P. Millet is saying that "zero" on his unit is 0.3 mV.)

Folks from the diyAudio forum report that putting the board into a shielded enclosure helps to reduce the noise. Since the enclosure suggested by Mr. Millet is made of plastic, I've ordered a couple of MuMetal Ultraperm Permalloy sheets to cover the box from inside. In fact, this is the same approach that E-MU was using for their line of audio interfaces, e.g. 0202 and 0404. Their bodies are made of plastic, but on the inside of the boxes there is isolating coating. Hopefully, this will help.

What's Next

After I finish with shielding the enclosure and assembling the device I will start measuring the parameters of the device itself, and will try it on some headphone amplifiers.

Tuesday, February 27, 2018

JDS Labs Subjective3, Part 2

As I was unhappy with the imprecision of "Mid" tone control on Subjective3, I decided to get rid of it—I don't see a point in having it anyway. So I opened my unit, and desoldered VR2 pot away. After that I've re-measured the frequency response and channel balance, and it got more even:

The inter-channel difference is ~0.3 dB, which isn't superb, and I'm thinking how could this be fixed.

I've also measured the pot I desoldered, and indeed it's not very well balanced. At the middle (notched) position it shows:

  • for the left channel: 23.5 kOhm one side, 24.8 kOhm the other side;
  • for the right channel: 24.98 kOhm one side, 23.94 kOhm the other side.
So if you are soldering Subjective3 yourself, think twice if it's really needed to put the "Mid" tone control in. At least, measure it first to check how well its channels are balanced.

Another thing that was puzzling me since I saw the partial schematics of Subjective3 here is whether it preserves absolute phase. Clearly, the schematics shows that the filter network is attached to the inverting input of the opamp. And I've actually checked whether the polarity is preserved by listening to in-phase stereo pink noise, where left channel was going through Subjective3, and right channel was connected directly, and there is no inversion.

Then I've found an explanation of this fact by examining the board and finding another stereo opamp there, near the output connectors. So apparently, it is used to restore the polarity.

And yet another thing I've figured out by disassembling my unit is that ready-made units soldered by JDS Labs employ DC source coupling. As it can be seen, C2_L and C2_R capacitors are left out, and their mounting holes are short-circuited. This may be a bit risky if the signal source is not of a good quality.

Sunday, February 25, 2018

JDS Labs Subjective3 Tone Control

I know, JDS Labs themselves call it "3 Band EQ", but I think it's more appropriate to consider it as a good old traditional tone control. There is nothing bad in having tone controls, in fact I've already been looking for one—I've even fitted BBE's Sonic Maximizer into its role, although it's missing attenuation capabilities, so I was very glad to learn that there is a real tone control available.

The first thing I've tried using Subjective3 was to tame excessive high frequencies in Beyerdynamic T90 headphones, and add a bit of bass to them, and it worked out great! Here are the knob positions I've ended up with for that:

And here is the resulting frequency response—it's what we can call a "tilt":

Then I did some exploratory measurements of the device to check how transparent it is. My general worry was that there are phase shifts, or extra group delay, or distortions. But it turns out that the device is pretty nicely engineered.

So let's see. First I've checked the bypass mode, then engaged mode with all the controls set to neutral position. The only graph where I saw any changes was THD:

Here, the bolder plots are the 2nd harmonics level, and the thinner plots are the 3rd harmonics.

Then I've started twisting the knobs to their minimum and maximum positions. In fact, JDS Labs have already shown the frequency responses, but no information about phase behavior and harmonics. So I've checked what happens if all knobs are turned to maximum—in fact, nothing bad, assuming that the input of the headphone amplifier doesn't overload, because Subjective3 adds about 20 dB of gain! But on the other hand, it does not add any significant harmonics or phase shifts. So it's all quite good.

One thing where I've encountered subpar performance was channel balance. The first problem was, after I've played with the knobs it has turned out to be impossible to bring them back to neutral positions. Subjective3 employs Alps knobs, and they have a center notch that can be felt physically. But this center position isn't a true neutral! Below are measurements from two positions of the knobs that were at the "neutral", super close to each other, yet they are both not truly flat, and employ a difference of >0.2 dB!

So, that was left channel. After I've measured the right channel, I've become even more sad, because the right channel had its own deviation (remember, that's at the same knob setting!):

And as we can see, the inter-channel difference is >0.1 dB. That's a lot! I think, I will unsolder the "Mids" knob because it doesn't seem to be really needed, and it affects frequency balance in the region where the ear is very sensitive. I will also check if it's possible to balance the channels better.

Other than that, I've found no problems with this device. One thing I would also like to note is that the frequency response shapes in Subjective3 are after Baxandall, with no plateau typical for shelving filters. Bob Katz explains why Baxandall's sloped shape is great for tone equalization compared to the plateau of shelving filters:

"We notice [by ear] the sloped portion of the curve more than the flat portion at the frequency extremes, since the flat portion becomes a sort of reference for the ears, even though both extremes are boosted entirely above the midrange"
Bob Katz "Mastering Audio", third edition.

Concluding, I would say this little unit is a must for all day listening at desktop, because it easily allows to adjust the sound to your personal preferences. The alternatives I'm aware of are:
  • Software digital EQ—not always applicable, also missing nice feeling of real knobs; although, digial EQ can be made really transparent, can use linear phase filters, and is super precise;
  • BBE Sonic Maximizer—lacks attenuation side, acts as an expander for highs, adds harmonics;
  • RME ADI 2 Pro—that's a DAC/DSP/Headphone amp package with lots of capabilities, including tone controls, EQ, and loudness contours, really nice but not cheap;
  • Dangerous Music Bax EQ—this is a mastering equalizer employing Baxandall tone curves and adjustable frequency ranges, superb quality, but the cost is really high.
So I think, in terms of price / quality ratio Subjective3 has its sweet spot. Hopefully, I will be able to improve it even further.

Thursday, February 8, 2018

112dB Redline Monitor Plugin

While looking around for other crossfeed implementations, I've found Redline Monitor by the audio company called "112dB." It's a plugin which is intended to be used with DAWs for simulating loudspeaker sound on headphones. On Mac, with Audio Hijack Pro it's quite easy to hook it up to the system audio output. The plugin has several emulation options, some of them missing on Phonitor headphone amplifiers, but it's price ($69) calls for a thorough evaluation before buying it. Thanks to generosity of 112dB, the plugin is available for a 60 days trial period, which I'm using the get some insight into how it works.


Let's check out the controls of the plugin. Those are very similar to the controls found on Phonitor:

I would start with the rightmost control—the simulated distance to loudspeakers. I think, it's the most interesting control because the type of processing and the resulting sound changes dramatically depending on its current value. When the value is "0 m", the plugin effect is the most non-intrusive, and resembles Phonitor's processing. All other settings of this switch introduce rather serious phase shifts and comb filtering to simulate room reflections.

The "Soundstage" control defines the total angle between the simulated speakers. "Center" is center signal attenuation. The "Dim" switch pre-attenuates the input signal to make sure that the processes signal doesn't clip. The rest of the switches are mostly needed for professional monitoring purposes, they are covered in the manual.


While doing my comparisons between Phonitor line of headphone amplifiers and Redline Monitor, I went through manuals for Phonitor 2 (this manual I have never opened before) and Phonitor mini. This was the first time when I discovered the tables and graphs in Phonitor 2 manual (page 17), and also noticed the statement "With the Angle switch you define the frequency-corrected channel crosstalk. In this case, we are dealing with „Interaural Time Difference“ (ITD)" in Phonitor mini's manual.

This was a very surprising discovery for me, because previously I was sure that Phonitor has almost linear phase response, and doesn't introduce any group delay. The delay was never showing up in my measurements, for reasons yet unknown. But from the manuals, it was obvious that Phonitor also employs ITD.

But that was all for good. And in fact, making such discoveries that I would otherwise have missed is one of the reasons I write this blog.

Equalization Differences

First I've checked how the equalization graphs of Redline Monitor look like, compared to similar settings on Phonitor mini. I've made the following settings on Redline Monitor:

Phantom center: -1.2 dB
Soundstage: 60 degrees
Distance: 0 meters

Which are semantically equivalent to the following settings on Phonitor:

Crossfeed level: Low
Angle: 30 degrees (this is between the speaker and the center, thus soundstage is 60 degrees)
Center: -1.2 dB

Surprisingly, the graphs look very different:

The amplitudes, the knee frequency, the inter-channel level—just everything is different, only the general principle stays the same.

However, if we look at group delay graphs, they look very much the same (again, at equivalent settings):

One small difference is that Redline Monitor has 300 μs ITD as low frequencies, while Phonitor 2 has 200 μs ITD.

Sonic Differences

In order to perform an ABX test, I've processed several music excerpts using Redline Monitor, and also recorded them via Phonitor mini crossfeed matrix (after I have discovered that my Phonitor simulation lacks group delay, I decided I shouldn't use it for tests). The same processing settings were used that are specified in the section above. The goal was to check can Redline Monitor and Phonitor mini be distinguished, and which one would I prefer.

The results are not very conclusive. Perhaps, the choice of tracks wasn't revealing enough, or I do need to train my listening skills better. With the modest processing amount I was applying, I couldn't even reliably distinguish the source from processed tracks, and distinguish Redline Monitor from Phonitor. The good news is that there isn't much change to the tonal balance with either crossfeed implementation.

"Distant" Modes

Let's go back to Redline Monitor's settings and check what happens to its transfer function when we start increasing the simulated distance to the speakers. Here the center image is at 0 dB attenuation, the soundstage is 60 degrees. I started with 0 m distance, proceeding in 0.5 m increments up to 2 m setting. Below is the graph of resulting frequency response, where darker colors represent larger distances. The blue plot is for the left channel, the red plot is for the right channel:

I guess, the ripples simulate the interaction of reflected sound with direct sound that happens when listening to loudspeakers in a room. The farther the listener is, the more enveloped they are in the reverberant field. As we can see, the amplitude of ripples is increasing with the distance, making the sound more and more colored.

It would be interesting to judge correctness of this simulation from the psychoacoustic point. In real conditions, the ears and the brain can "listen through" the room, discarding these colorations, but the brain has much more information, e.g. changes in received sound with subtle head moves, which are absent in this simulation. So the question is open whether these ripples just color the sound, or are they "converted" into speaker distance information by brain, or both processes happen to some degree simultaneously.


I think, Redline Monitor can be used as a substitute for Phonitor mini when the latter is unavailable. Although, their processing is a bit different, one needs a very trained ear in order to distinguish between those two implementations.

For Redline Monitor, I would recommended to use 0 m distance setting in order to avoid comb filtering occurring with the other settings of the "Distance" control.

Thursday, January 18, 2018

BBE 802 Sonic Maximizer Measurements and Teardown

While watching this YouTube video that analyzes the transfer function of the older model of the BBE Sonic Maximizer—the 802 model, I've noticed one thing that I miss on the current generation of Maximizers—the ability not only to boost high frequencies (HF), but also to attenuate them. Out of curiosity, I've bought an 802 unit on eBay and performed the same measurements I've done previously for 282i.

What's Inside

But first, I looked under the cover of the unit to see if it's based on the same NJM2153 chip as the 882 model, and I've found out that it in fact isn't! This is what we can find inside:

The first thing we can see is a pair (one per channel) of giant chips marked "BBE." That's the original "sound enhancement" chip. It's interesting, that compared to NJM2153 package which has 20 pins, of which 18 are actually used, this "BBE" chip only has 18 pins, minus 1 not connected, thus only 17 are in use:

It would be interesting to figure out what is the extra input that NJM2153 receives compared to the old BBE chip, but for that I will need to trace the connections on the board. Although, that shouldn't be hard since the board has in fact only one layer, I'd leave that for some time later.

The other chips we can see here are opamp assemblies. There are 3 of them per channel:

NE5532N—the ubiquitous audio opamp, used for balanced output;
SGS LM324N and SGS TL074CN—used for driving LEDs.

These are pretty much the same components that are used in the 882 model, except that 882 uses electronic balancing of inputs, and for that purpose it employs two more pairs of NE5532. Whereas 802 only uses old school transformer balancing (you can see a pair of small transformers per channel.)


I'm presenting the measurements in the same order as for 282i here.

Group Delay

Unlike 282i, this 802 unit doesn't affect group delay at all when it's in bypass mode. From what I've seen in the frequency response measurements, when in bypass mode, the 802 excludes all its circuits from the signal path, which perhaps isn't true for 282i.

So, this is the group delay plot when processing is enabled:

The numbers are pretty close to what manual is saying, discounting by this unit's age. We can conclude that this functionality didn't change much with the Maximizer evolution.


Here the things are becoming more spooky. Look at how harmonic distortions increase when the unit is in processing mode (orange) versus the loopback measurements (black):

The faint line is the level of 3rd harmonics—it reaches 0.1% for middle frequencies and crawls up to 1% for bass. Although, it doesn't contradict the official specs—they say "less than 0.15% @ 1kHz", this is much worse than the modern 282i shows.

According to John Siau's calculations, 0.1% of distortions translates into -60 dB noise below playback SPL, which can be audible.

Frequency Response

So, what about the ability to attenuate HFs, is it really there? Yes, indeed:

As we can see, setting the "Processing" knob to the minimum position ("1") attenuates the HF by 6 dB. Setting it to the middle setting ("5") provides a flat-ish response, and turning "Processing" all the way up produces a bump at about 4 kHz.

However, we can also see that HF roll down quickly after 10 kHz on any setting, which is much less exciting. The modern versions of the Maximizer demonstrate a flat FR up to 20 kHz (when the knobs are at their minimum positions.)

Finally, what about that non-linear frequency response that we have seen on 282i and that effectively acts as an expander for HF. Yes, it's there:

(Note that the graphs were produced using white noise as a source signal, thus at low frequencies the plots are wiggly.)

With both knobs at the maximum setting, we can see that the 802 unit doesn't boost the HF if the signal level is low. Even more interesting picture is when the "Processing" knob at the minimum level:

As we can see, at low signal levels, the HF are less attenuated, so the unit works as a compressor! If we align the lowest (red) plot with the highest (magenta) one at 1 kHz, the delta at 6 kHz is about 4 dB.


The older BBE Sonic Maximizer model 802 provides some interesting abilities to manipulate high frequencies not available in the modern models, but unfortunately suffers from high distortions level, and compromised frequency range. Perhaps, at the time when it was introduced (around 1980-s), these specs were acceptable, but currently they clearly don't meet the bar. So unless you intend to process sources that by their nature has high distortions and reduced frequency range (e.g. analog tape), there is absolutely no point in using this ancient unit.

Sunday, January 7, 2018

BBE 282i Sonic Maximizer Measurements

The Need for Tone Controls

I'm reading the new edition of awesome F. Toole's book "Sound Reproduction." Here, in Chapter 4.4 he grieves for the demise of tone controls on modern hi-fi preamps. Indeed, I'm recalling domestic vintage radio + turntable combos my father and his friends had—there always were "bass" and "treble" knobs. More expensive systems featured multi-band graphical equalizers. Definitely, at that era everybody understood that both program and the reproduction chain are not ideal, and some tonal correction may be required.

However, the aspiration for a "clean" reproduction path shaved all the extras off (they contaminate the sound!), and left us only with volume controls on most of hi-fi units. This would work if all recordings were perfectly balanced, and our hearing were linear (or we were always listening at the same reference volume level). But since this is simply not true, it's often desirable to shave off some extra high frequencies that were added by the mixing engineer in order to "reveal" the vocals, but ended up sounding really harsh, or to add some bass when listening at low volume levels.

BBE Sonic Maximizer Mystery

I've started searching for a desktop unit that would implement just tone controls, not a full equalizer—they are bulky and require too much tweaking. But due to the aforementioned "purity" trend in audio equipment, it's next to impossible to find such an unit. Of course, I could just implement the tone controls as a DSP plugin, but at least with Pulse Audio and LADSPA, it's not trivial to add real-time controls to it. Also, virtual knobs never feel as good as physical ones.

Somehow, I've stumbled upon the family of units jointly called "BBE Sonic Maximizer" featuring just two control knobs—a good sign! However, the labels of the knobs were quite cryptic: "Lo Contour" and "Process", and there was nothing about "tone control" in the description the unit, but rather lots of promises in marketing-speak of achieving audio nirvana once this unit is inserted into recording or reproduction chain. That looked really suspicious.

Even more suspicious were reviews on different forums (mostly related to sound recording), where people were either raving about how this unit improves the sound of recorded drums and makes their sound "punchier", or advising not to waste money on this unit because it's snake oil. A lot of YouTube videos demonstrate processing results, and from watching them it seemed like the unit really adjusts the tone, but then there were always people saying that it's not a tone control, however they couldn't provide any specific details.

Digging into Details

I was looking for any objective measurements of the Maximizer, but finding none. At last, I've found three manuals: one for an older 802 unit, another for a newer 882 unit, and another for the modern version 882i.

The manual for 882i is the most useless one—it only says about "envelope distortion" occurring in speakers that this unit is designed to solve, provides unit connection schemes, and brief technical specs stating some distortion figures, and the fact that tone correction happens at 50 Hz and 5 kHz, with maximum attenuation of +12 dB.

The term "envelope distortion" is equivalent to "group delay distortion" which means adding non-uniform delay to different groups of frequencies. According to "Electroacoustics" book by M. Kleiner, horn and transmission line speakers are susceptible to noticeable group delays. Seems like the BBE unit can actually be useful for PA and old studio monitors if you don't have a DSP processor. But I think, modern speakers and especially modern powered studio monitors have required compensation circuits built-in.

The manual for 802 actually explains what the unit does in terms of group delay. The audio signal is split into 3 frequency groups by dividing the spectrum at 150 Hz and at 1200 Hz. The LF group is delayed by 2.5 ms, the Mid-Frequency (MF) group is delayed by 0.5 ms. The HF group is left intact.

As for tone correction, the LF group is simply boosted according to the "Lo Contour" knob. The amount of HF boosting actually depends both on the "Process" knob, and the RMS level of the MF group. This is the most intriguing stuff.

Here the manual for 882 comes to help. Very atypically for commercial electronic products, this manual contains the actual electronic circuit scheme of the device. It shows that the heart of the 882 is NJM2153 chip which also has a technical manual. Finally, we can see some graphs!

This plot supposedly shows the amount of frequency correction applied when both "Lo Contour" and "Processing" knobs are at the maximum position. The LF boost remain constant +12 dB regardless of the input signal level. Whereas the HF boost depends on the input signal level. Here is another graph, a "cross-section" at 10 kHz:

Interestingly, that the description of the NJM2153 chip doesn't align well with what the manual for the 802 unit is saying. The chip description says that the amount of HF boost depends on the overall input signal level, but the 802 manual states that it depends on the RMS of the MF group only. Perhaps, this implementation detail was changed from 802 to 882.

It's also interesting, what happens when the "Processing" knob is at the minimum value—does HF group get attenuated if the input signal level is low, or maintains the input level? On the EC schematics, the VCA is controlled from directly from the signal level meter, so it should not depend on the "Processing" knob. But it's better to check.


I've bought an unbalanced desktop version of the BBE Sonic Maximizer—model 282iR. From the scarce technical specs revealed in the manuals, it seems to use the same processing pipeline as 882 or 882i does, but is made in a different form factor, and with combined knobs for left and right channels. Also, as I've said, 282iR uses unbalanced RCA or 3.5 mm inputs and outputs, so has 3 dB lower output level than 882i which uses balanced XLR connections.

Since the BBE unit is an analog line-level signals processor, it's quite trivial to measure it using an ordinary sound card. I was using MOTU Microbook IIc.

Group Delay

Let's start with group delay. It remains the same for any input signal level, and the only parameter that affects it is whether the unit is in bypass mode:

The bypass mode is the red plot, the green plot is when processing is engaged. As we can see, in processing mode the unit indeed adds ~2.5 ms group delay to LF, and ~0.5 ms to MF (as an average value). So, the unit adds some GD distortion even when it is in bypass mode.


The manual for 282i states < 0.1% at -10 dBu input across the entire 20–20000 Hz range. That's actually quite a lot (not good). In fact, it seems to be an order of magnitude better:
This plot shows the 2nd harmonic. The black plot is loopback measurement for Microbook. Red is bypass, green is processing mode with both knobs at the minimum setting. As we can see, the level with processing enables is < 0.01%.

Channel Balance

Since the 282i unit is designed to process both channels at once, I'm expecting the unit to maintain the original balance of the input signal.

As I've checked, in bypass mode the balance is held very much precisely. Looking at the 882 unit schematic, the bypass mode just directly connects output to input, so that's what I would expect. In processing mode, the difference is about 0.1 dB at 1kHz—not too bad, but could be better.

Frequency Response

Finally, the most interesting part. Since the FR of the unit changes with the signal level, I was using Microbook's hardware white noise generator and was performing a real-time FFT analysis in Room EQ Wizard. The method was to change the level of the noise, and observe how it affects the output frequency response. The resulting curves are not that pretty as obtained from sine sweeps, but still reflect the trends.

As it can be seen from the graph, the frequency response plots at maximum "Lo contour" and "Processing" knob setting indeed resemble of those from the NJM2153 chip manual shown above. The level of bass boost remains unchanged, while the level of HF boost falls down once the sound level becomes low, thanks to the attenuator controlled by the input level monitor.

With the knobs at the minimum position, the HF range can even be attenuated for low power signals.


Recall that I've encountered the BBE Sonic Maximizer while looking for a tone controls device. So, can Maximizer be used as a tone control? Somewhat. It definitely can boost LF or HF, which is good. As for the opposite direction—cutting, it depends. For bass it's not needed as often. For treble, I'm curious how the variable attenuation actually helps. Need to check with actual commercial recordings.

Another thing—the group delay. It's definitely not needed for headphones because over-ear models anyways use a single driver. Does the group delay introduced by the unit affect the sound negatively? I will need to check with wide spectrum transients like drums and percussion.

And some additional distortion that the unit adds when processing is engaged. Certainly, 0.01% of 2nd harmonics isn't fatal, but specs of Grace SDAC and Phonitor Mini feature at least 10x less distortion. Although, we can say that those add some warmth to the sound. Again, need to do some listening.