I did some testing using DDMF's Plugin Doctor in standalone mode paired with a Radial X-Amp reamp box with all of it feeding back into a MOTU 16A audio interface and a Line Audio 8MP, which is the most linear preamp I own. It's quite linear as confirmed with null testing on a variety of tasks over the years. No analog chain will null perfectly, but this setup is adequately linear to make some high level statements with confidence. Output from the X-Amp was set such that the Anagram's input meter was hitting around 75% of the way up the scale. Essentially it was hot enough to deliver an adequately useful signal, but not so hot that it would cause clipping. My guess, based on some findings you'll read below in the compressors analysis, is that this puts the input gain somewhere around -15dbfs, give or take. Everything on the digital side of things was set to a sample rate of 48khz to match Anagram's internal sample rate. With that disclaimer out of the way, let's do a little digging.
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My first test was to better understand the inherent frequency response of the Anagram with nothing loaded into any of the processing block slots, and to maybe get a glimpse into how the unit deals with content below 20hz as well as the all-important content above Nyquist, which in the case of the Anagram with it's 48khz sample rate, is 24khz. This was all done with an empty channel strip and no global EQ.
As you can see from this example, Anagram does nothing with content below 20hz. It comes screaming through with no built-in high pass filtering. That's fine for this application. I always use a high pass filter in my live rig, so this is a non-issue but it's something I wanted people to be aware of when building their chains. Ignore the odd shape of the amplitude curve (gray line) below 20hz. In my experience, Plugin Doctor's metering gets a little weird down in these sub-20hz ranges.
There's an odd dip in frequency response starting around 8khz, being most pronounced around 18khz at something like -2db, then spiking up roughly 6db from that point at around 22khz before falling like a rock to Nyquist at 24khz. I'm guessing the super steep low pass filter is Anagram's anti aliasing filter. The dip and spike ahead of this filter is curious, but I'm sure there's a good reason for it. This all occurs in a frequency range that, for most applications, is inconsequential for bass guitar and not something I would be concerned about. I did not test for cramping of EQ as it approaches Nyquist. Nothing I do as a bassist relies on frequencies that high, so I didn't find it a relevant test. If someone wants to understand the cramping behavior at Nyquist for high frequency EQ bands I can run that test upon request.
For the range of frequencies that most bassists should actually care about (20hz-10khz with content from 4khz-10khz becoming increasingly less meaningful as frequency increases) I would say that Darkglass did an excellent job of creating a unit that is sufficiently flat to be an excellent tone building platform. Good job!
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Next, I wanted to dig into the dynamics issues I've been talking about since last night, most notably with the Hyper Luminal compressor not actually compressing. Let's start there and then we'll look at the Buss compressor and a software compressor for comparison afterwards. Here's the Hyper Luminal's dynamics plot using its 20:1 ratio, fastest attack time, and slowest release. I did this in the default setting, again with input maxed and again with the input at minimum. You'll note that the amplitude curve (gray) behaves oddly at input levels below -50dbfs, which is largely inconsequential other than possibly indicating what the lowest level the input threshold can detect. What is meaningful is that once -50dbfs is reached, extending all the way up to 0dbfs in +1db increments, the compressor is completely linear and shows absolutely no gain reduction.
Let me be clear when I state this: As of the current firmware version (today's date is June 6th, 2025) the Hyper Luminal Compressor processing block
DOES NOT PROVIDE ANY GAIN REDUCTION OTHERWISE KNOWN AS COMPRESSION.
I still believe this is a bug that Darkglass did not intend, and it will likely be fixed in the future. For now the Hyper Luminal processing block is simply not a compressor. If you are using it as such and think you're hearing compression, you are incorrect. Hopefully Darkglass fixes this soon.
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Next let's look at the Buss compressor, which I'm happy to say does actually provide gain reduction. This analysis was done with the threshold control at its lowest allowable value of -40db, a ratio of 10:1, the fastest attack time of 0.1ms, the slowest release time of 1.2s, and the blend control set to 100%. Content below -50dbfs is once again showing up a little strangely in the Buss compressor, but I'm inclined to not care as it doesn't cause any audible artifacts. With this setting, content around -15dbfs triggers the compressor and the ratio even looks right for a 10:1 compressor. Unlike Hyper Luminal, the Buss compressor actually compresses. I did not think my testing of this device would reveal a compressor that doesn't compress, but here we are.
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As a comparison, loaded Fabfilter's C2 compressor into Plugin Doctor and duplicated the Buss compressor's settings as closely as possible using the C2's "clean" algorithm, which was my best guess at a similar type of circuit (feed forward, soft knee, low distortion). In hindsight I should have made the knee slightly harder to better match what Buss is doing, but the important thing to check was the amplitude where compression begins and the gain reduction slope. The slope looks very similar, but the onset of compression occurs a little higher than with Buss. These differences are more as a sanity check than anything and are largely inconsequential. The key thing to take away is that as of today, Buss compressor is the only viable option for compression within Anagram, and as a bonus, it's a decent sounding option. I still think my Keely Compressor Pro sounds a lot better than what Buss does, but Buss sounds good enough for my purposes on live dates.
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Next I wanted to take a look at how the Anagram addresses aliasing. My hope was that this would reveal the source of the inharmonic content that I can readily hear when using non-linear blocks (i.e. various forms of distortion), and curiously also with the pitch shifting block. As mentioned previously, I had this same problem with HX Stomp but never took the time to test it. Let's see how Anagram does with aliasing prevention.
The first test was using the B3K block on its default settings. The results are actually quite good. You can see how the harmonic series more or less dies right at Nyquist (top of the Y axis). This is what you want to see and is encouraging. The problem is that now I'm even less certain as to the source of the inharmonic content. I have ideas for other tests I can conduct to find a possible source, but I need to do a bit more work before those can come together.
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Lastly, I wanted to check the Pitch Shifter block set to -1 semitone. This should be a completely linear process but given how glitchy it sounds, I really didn't know what to expect. Honestly, the results don't paint a much clearer picture. The harmonic series doesn't appear to be aliasing, but there's so much other "stuff" going on that it's hard to say anything other than the fact that isn't an adequate test to deduce what I was hearing. What is interesting is the pairs of near-parallel harmonics. I suspect these are a semitone apart and may indicate that even with the wet level set to 10.0 and the dry level set to 0.0, some dry signal may still be leaking through the circuit. It certainly sounds like that in the room and in my IEMs. I'm going to play around with the filter cutoff a bit to see if that helps, but I'm inclined to think that pitch shifting block is not going to be a viable way to play tunes in Eb tuning without having to grab a different bass or physically detuning the one I'm playing, which is a bummer. I think Helix's pitch shifter sounds better than Anagram's, but even that one I didn't like.
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