• TalkBass has been independent since 1998. Add your voice.
    Create a free account to reply to discussions, view embedded media, and browse with fewer display ads.
    Join freeLog in
    Want zero display ads or expanded classifieds tools? Compare plans.

Darkglass Anagram Multi FX Announced

You're not alone in this. Even as an audio engineer I sometimes struggle. My usual move is to intentionally make a compressor work too had and get mega squashy to have a good audible reference to its character, and then back it down to something reasonable and compare against it in bypass. When I couldn't get either of Anagram's compressors to provide meaningful levels of squash I knew something was up. Plugin Doctor will tell me more later today when I get that setup for further testing.
I live by the meter
 
  • Like
Reactions: Killerfridge
Sorry for the scattered posts...

I also tried using the effects loop return as an aux input. This is a common need for me, particularly on road and fly dates. I somewhat frequently find myself sitting in a hotel room the night before a gig needing to work out new material from charts that were given to me very near the actual gig date. A viable headphone solution with an aux in helps a lot in these situations. I didn't want the output of my iPhone to be processed by all of the stuff in my preset chain. My first thought was to stick an effects loop block at the very end of my chain, but this resulted in either the sound from my phone being audible with the block engaged, or the sound of my bass being audible with the block disengaged, but there was no way to have both be audible simultaneously. It's entirely possible that I was doing something wrong, but I did find another solution. I used a split block at the very end of my chain, then in the lower row I placed the effects loop block with the send level at 0, and then a merge block after the split block in the top row brought the return signal back to the output. This worked great, but it was a bit fiddly and did require some tweaking of various output levels to get everything working properly. I'm guessing there's a better way to do this that I'll get figured out at some point, but for now it was definitely possible to send the output of my iPhone into the Anagram without that output being processed by the preset chain, and then hearing it along with the processed sound of my bass.
Unless I am understanding what you are trying to do wrong, all I had to do was put a Return block at the end of the chain and it worked for me.
 
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.

-----------------------------------

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!

Empty Channel Frequency Response.png



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.

Hyper Luminal Dynamics.png



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.

Buss Dynamics.png



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.

Fabfilter C2 Dynamics.png



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.

B3K Sweep.png



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.

Pitch Sweep.png
 
Last edited:
Unless I am understanding what you are trying to do wrong, all I had to do was put a Return block at the end of the chain and it worked for me.
I tried that, but as described in my previous post I could either get the sound from the return (i.e. my iPhone) by engaging the block, or the sound of my bass by disengaging the block, but I could not get both to happen simultaneously. More testing required on my part. It's likely I simply did something incorrectly.
 
@Rockyy12 can you address my findings with the Hyper Luminal compressor processing block in posts 1077 and 1103? Is Darkglass aware of this behavior where Hyper Luminal provides no gain reduction, and if so is there a planned firmware update to fix the issue? Am I somehow using Hyper Luminal incorrectly?

I don't mean to put you on the spot, and as I've stated previously, I suspect this is simply a bug in the code that needs to be resolved. Your participation in this thread is much appreciated.
 
@Rockyy12 can you address my findings with the Hyper Luminal compressor processing block in posts 1077 and 1103? Is Darkglass aware of this behavior where Hyper Luminal provides no gain reduction, and if so is there a planned firmware update to fix the issue? Am I somehow using Hyper Luminal incorrectly?

I don't mean to put you on the spot, and as I've stated previously, I suspect this is simply a bug in the code that needs to be resolved. Your participation in this thread is much appreciated.

Might seem like a silly question but have you increased the input control on the Hyper Luminal? The way that compressor works is that you get more compression by turning the input control up due to how the threshold is built within it. It's explained at 23:35 in the video below by the wonderful @Paajanenae

 
Might seem like a silly question but have you increased the input control on the Hyper Luminal? The way that compressor works is that you get more compression by turning the input control up due to how the threshold is built within it. It's explained at 23:35 in the video below by the wonderful @Paajanenae


As stated in previous posts, I tried it with the input maxed and with the input at minimum. No change occurs. The lack of a threshold control is common among FET designs, the most common of which being the 1176.
 
.. first thought was to stick an effects loop block at the very end of my chain, but this resulted in either the sound from my phone being audible with the block engaged, or the sound of my bass being audible with the block disengaged, but there was no way to have both be audible simultaneously
I went down this rabbit hole before and luckily Olli helped me out through a fb group. Other than the Device settings set to Stereo, you have to add a FX RETURN block, NOT the split and merge wizardry
 
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.

-----------------------------------

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!

View attachment 7245469


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.

View attachment 7245468


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.

View attachment 7245466


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.

View attachment 7245467


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.

View attachment 7245465


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.

View attachment 7245464
Thank you @silky smoove for the in depth analysis you are providing. I would LOVE to hear from the Instagram and YouTube shills on all of this! You know who you are!
 
I just finished building preliminary presets for the four applications I’m in most frequently.

1. My “corporate” band doing weddings, private events, corporate events, etc. This band is almost exclusively IEMs. Since I still use an amp to rehearse tunes, I built a chain with an output block after all of the processing that I want happening in both locations. The output block sends output 3 to my amp. After that I have an amp and cab that only go to my IEMs via output 1.

2. An original band that I’ve described as “progressive soul.” This was a simple chain up front, but the footswitch effects were complicated. A synth sound using five blocks under one macro, an overdrive sound using three blocks under one macro, and a modulation sound using two blocks under one macro. The modulation is a combination of chorus and low level shimmer reverb that’s really cool. The synth sound used three blocks I previously said I disliked; Sublime, Alpha-Omega, and rhe Moog filter. It doesn’t sound as good as the version I built using three 3Leaf pedals (Octabvre, Doom2, Proton) but it’s good enough for live purposes since it’s only application in this band is during a solo feature.

3. A preset for gigs on electric upright. This was very basic up front always-on stuff. After that I used a split block set at 800hz with equal output to each row so that I could apply a room reverb to just the top end of the signal. This makes the electric upright feel a bit more like an acoustic. A merge block brings it back together.

4. An indie pop and blues band that didn’t need much. A little always-on processing up front, an overdrive footswitch, and a low pass filter at the end of it all to keep it less bright than my usual tone for other bands.

My next task is to build some purely creative stuff. I like having a few of these types of sounds available to spark inspiration when writing. Time to get weird…
 
I just finished building preliminary presets for the four applications I’m in most frequently.

1. My “corporate” band doing weddings, private events, corporate events, etc. This band is almost exclusively IEMs. Since I still use an amp to rehearse tunes, I built a chain with an output block after all of the processing that I want happening in both locations. The output block sends output 3 to my amp. After that I have an amp and cab that only go to my IEMs via output 1.

2. An original band that I’ve described as “progressive soul.” This was a simple chain up front, but the footswitch effects were complicated. A synth sound using five blocks under one macro, an overdrive sound using three blocks under one macro, and a modulation sound using two blocks under one macro. The modulation is a combination of chorus and low level shimmer reverb that’s really cool. The synth sound used three blocks I previously said I disliked; Sublime, Alpha-Omega, and rhe Moog filter. It doesn’t sound as good as the version I built using three 3Leaf pedals (Octabvre, Doom2, Proton) but it’s good enough for live purposes since it’s only application in this band is during a solo feature.

3. A preset for gigs on electric upright. This was very basic up front always-on stuff. After that I used a split block set at 800hz with equal output to each row so that I could apply a room reverb to just the top end of the signal. This makes the electric upright feel a bit more like an acoustic. A merge block brings it back together.

4. An indie pop and blues band that didn’t need much. A little always-on processing up front, an overdrive footswitch, and a low pass filter at the end of it all to keep it less bright than my usual tone for other bands.

My next task is to build some purely creative stuff. I like having a few of these types of sounds available to spark inspiration when writing. Time to get weird…
This isn’t helping my GAS for the Anagram. This year I’ve been searching for an amp and clean DI sound solution for live shows where I use IEMs, specifically where FOH can mix my clean DI and an amp and I’m only hearing an amp in my ears. Seems like Anagram would be perfect for this as there are two DI outs and plenty of routing options. I’d love to hear the compressor get fixed though as I’d expect this to be my all in one solution for shows.
 
  • Like
Reactions: IvanBassist
Another big question is this:

Will plugin in devs ACTUALLY make Anagram specific versions?

Because thst would solve a lot of problems.

Given a market place, there is bound to be some level of participation and it would make sense to see an officially licensed SVT (or other) plugin directly from the manufacturer.

While I’m not specifically an audio dev by trade, I’ve poked a bit at the JUCE framework, which follows a pattern that separates what it does from its UI (e.g., MVC or comparable). This would significantly lower the bar by more or less let the existing guts keep working as is, while just needing to be wired up to a different user interface and recompiled (a step DG said they’ll help with). Obviously glossing over a good bit, but it isn’t a complete rewrite from ground up.
 
What are our options if we want to use an expression pedal and run something through the FX loop? Is it one or the other (which I think is the case) or can we do both?
Disclaimer: I haven't tried any of this myself. The mono effects loop and the expression input are separate jacks. That is not true if you need to do stereo for the loop. But I imagine few people will need a stereo loop, as bass players.

On the other hand, if you need to use a wired input for an audio aux source, then you can't have an expression input at the same time. That jack is used for one or the other.

Hope that helps!
 
  • Like
Reactions: bruno40gr
What are our options if we want to use an expression pedal and run something through the FX loop? Is it one or the other (which I think is the case) or can we do both?

You can run an expression pedal in the return, and then use an insert cable in the send to run a single loop.

Alternatively when MIDI support arrives you should be able to use that.
 
  • Like
Reactions: bruno40gr
Had my first use of the Anagram for worship this Sunday. The preset I put together has everything I need and I thought it sounded good. All worked perfectly and I got a few compliments on tone (which for me is a very rare thing 😂) and FOH loved how clean and silent it was. Our guitarist with a QC loved what he saw and was saying what he's really waiting for is a device you can play audio through via Bluetooth👌This thing is building a reputation.

Thanks to all involved in making the Anagram - it's worth every penny and it's a joy to use.