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Understanding the state of the DSP market

Keith_W

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I thought I would make a public service post about DSP, because I have noticed that many people don't seem to understand it. I hope this post helps you understand what a DSP product offers you, what the advantages and disadvantages are, and ultimately help you choose what you want or need. This is a complex topic with a lot to say, and I have to keep it brief so that it fits in a forum post and can be easily understood by a general audience. So I am sure I will be criticised for not making certain points by the usually tough ASR crowd. To those people I say: screw you, write your own DSP introduction and see how easy you think it is to condense everything you know into a few posts! No marketing BS in this post, by the end of it you should be able to understand exactly what you are buying when you choose a DSP product.

The most fundamental difference is how DSP products implement DSP. They may use IIR filtering, FIR filtering, or a combination of both (mixed phase).

IIR filtering

1745160800005.png


IIR, or Infinite Impulse Response filters, are recursive filters, meaning they have feedback. Because of the feedback, the impulse can theoretically go on forever. The structure of an IIR filter is shown above - note that there is a feedforward stage and a feedback stage. Each feedforward/feedback stage has gain coefficients and delays. Without going into the math, each feedforward/feedback stage is a quadratic equation. Because the output of the filter depends on the ratio of the feedforward and feedback stage, an individual set of coefficients is known as a biquad.

Each biquad is a 2nd order filter, meaning it has two poles and two zeros. Therefore, a 2nd order filter (for example, a 12dB/oct Linkwitz Riley LPF) uses one biquad. An 8th order (48dB/oct) uses 4 cascaded biquads. A single PEQ is one biquad. Therefore, how many biquads you have available determines how complex your filter can be. With more biquads, higher-order slopes are possible, and so are more complex corrections.

This is how the filter works: first, the input signal (music) must match the sample rate of the filter. Either the music is resampled, or the filter designed for the music sample rate. At t=0, the first sample is multiplied with b0 and sent to the output. At t=1, the second sample arrives and is multiplied with b0. Meanwhile, the first sample progresses to b1 and a1 with and sent to the output or feedback stage respectively. And so on it goes. t refers to a time period which is 1/sampling rate, so if the sampling rate is 48kHz, t = 0.021ms.

Because IIR is economical with CPU power, it is commonly found on devices which have limited computational capacity such as AVR's and notably, MiniDSP, Danville Nexus, and the like. However, these devices may also have a limited number of biquads, depending on the model. The MiniDSP SHD which I am playing with at the moment has 18 biquads per channel, with a total of 4 channels. Of these, 8 biquads are reserved for the crossover, meaning you have 10 biquads available for PEQ's per channel.

As mentioned, complex corrections are achieved by cascading biquads, however this creates its own issues. Quantisation errors can occur, especially at low frequencies where precision is crucial. Because of the recursive nature, these errors lead to noise accumulation and potential stability issues. Each biquad is minimum-phase, meaning that phase distortion accumulates the more biquads are used. This makes design of a filter with multiple biquads difficult to design if we want to maintain the desired response and avoid unwanted artefacts.

Biquads also have poor portability. Because different manufacturers have different implementations of coefficients, biquads have to be specifically tailored to one model of DSP. The same set of biquads will produce a different output from a different manufacturer. Also related, biquads are tied to the sample rates they were designed for. For example, if a biquad designed for 48kHz were applied at 96kHz, frequency shifts would occur. Specifically, they would double - so a 100Hz LPF becomes a 200Hz LPF.


FIR filtering
1745161753804.png

FIR, or Finite Impulse Response, has a fundamentally different filter structure. It contains a series of taps. Each tap is a delay-gain pair. If you look closely at the structure of 1 tap, it is exactly the same as one coefficient in an IIR filter. The difference is, IIR filters are recursive and FIR filters are not. This means that if we want to replicate what an IIR filter does with a single biquad, a great number of taps are needed. What this also means is that it is impossible to exactly replicate an IIR biquad, because an infinite number of taps is required.

You may have heard of taps and bins. Taps are time domain entities, and bins are frequency domain entities. For our purpose, it is enough to think of them as equivalent since they can be converted into each other by either inverse FFT or FFT. If you have 65536 taps, you also have 65536 bins. The bin width determines the resolution of the filter. It can be calculated by Fs/n (sampling rate/taps). So if you have 65536 taps and a 48kHz sampling rate, each bin is 0.73Hz wide. This is roughly analogous to a graphic equalizer with 65536 bands of adjustment, with each band being 0.73Hz wide.

This is how it works: at t=0, the first sample is arrived, multipled with the gain coefficient, and sent to the output. At t=1, the second sample arrives, is multiplied, and sent to the output. Meanwhile the first sample has been delayed by 1/Fs before it is multiplied with the second gain coefficient, and on it goes. It keeps going until we reach the filter length which is the total number of taps that we have.

More taps and higher sampling rate means that computational requirements go up. If a filter is 65536 taps long and we have a 48kHz sampling rate, this means we need to hold an absolute bare minimum of (65536/48000) = 1.37 seconds of audio in memory. Multiply that by the number of channels you wish to process, add a bit to prevent starvation, and you can see why we need an ARM processor or a PC to implement FIR filtering with long filter lengths with multiple channels.

This all sounds terrible, so why bother with FIR filtering at all? This is because any filter can be created provided you have enough taps. The same 65536 taps can be used to create very high order filters or very complex corrections with no additional cost (beyond the computing requirements which have already paid). With IIR filtering, you will have to use more biquads to achieve the same outcome, and as we have seen, using more biquads has its own disadvantages. FIR filters can do everything an IIR filter can do, and more. Well, everything except run on low CPU power devices. FIR filters are robust, not prone to instability, and not prone to numerical precision errors. But more importantly, FIR filters can be used to create linear phase filters. More on that shortly.

Mixed Phase
This is achieved by cascading an IIR filter with a FIR filter, sometimes the other way around. Many MiniDSP products (though not all!) are mixed phase. This is done to obtain some of the advantages of FIR filtering (ability to design more complex filters) with IIR filtering (low computational requirements).
 
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Minimum Phase vs Linear Phase
Both IIR and FIR are capable of creating minimum-phase filters, however (for reasons I won't go into), only FIR filters are capable of creating linear phase filters. I prefer the terms minimum delay and constant delay respectively, because that is exactly how those filters work. Minphase means the signal passes through the filter in the shortest amount of time possible. Because of this, minphase filters are causal, meaning they act only on past and present inputs. Linphase means that an artificial delay is deliberately imposed on the signal so that we can obtain other advantages. Linphase filters are acausal meaning they can act on past, present, and future inputs. This means that corrections can be made before they happen.

There are many important differences between minphase and linphase:

1745166291352.png


The maximum energy is concentrated at the beginning of the impulse for minimum phase (red curve). For linear phase, the maximum energy is concentrated in the middle of the filter length (green curve) to obtain the constant delay characteristic. Since this is a 65536 tap filter, the impulse is positioned at sample number 32768. This is one of the quickest ways to tell if you are looking at a minphase or linphase filter - look at where the impulse is positioned.

We can also see that the ripples (or ringing) of a minphase filter occur only after the impulse response (post-ringing), whereas a linphase filter has both pre-ringing and post-ringing. Not obvious in this illustration is the fact that minphase has approximately double the amplitude and length of post-ringing of a linphase filter. Also, when a symmetrical HPF/LPF linphase is summed, the pre and post ringing perfectly cancel.

However, the cancellation is only perfect on a computer simulation. In the real world, drivers may be misaligned, have different directivity characteristics (and thus may sum on-axis but not off-axis), it may be necessary to have asymmetric HPF/LPF's, and so on. So in reality, pre-ringing may be audible if a filter is poorly designed. Whether it is actually audible depends on the length and volume of pre-ringing, and whether it exceeds psychoacoustic thresholds of temporal masking.

Because of the constant-delay characteristic, linphase filters have a built-in latency. Half the filter length needs to pass before the signal emerges. This can be calculated by (filterlength/2x samplerate), so for our example of 65536 taps at 48kHz, the latency is 0.683 seconds. This makes linphase filters unsuitable for any situation where latency is unacceptable - for example, it will create lip sync problems with video. If you are a fighter pilot, planes can travel a long distance in half a second.

1745167131302.png


In a minphase filter, the amplitude response is inextricably tied to the phase response. If you know one, you can calculate the other using the Hilbert transform. The steeper the slope, the more the phase rotation. Inversion of the curve also inverts the phase response. Phase can not be manipulated independently of amplitude, with the only exception being an all-pass filter.

In contrast, a linphase filter allows you manipulate the amplitude and phase response independently of each other. The illustration shows a minphase and linphase version of a 4th order Linkwitz-Riley LPF. We can see that the phase angle remains unchanged at 0deg for the linphase filter, while the minphase filter rotates phase at a typical 90deg per filter order.

What this means is that filters are much easier to design. It is well known that certain types of traditional crossovers do not sum properly. Vance Dickason dedicates an entire chapter discussing crossover phase and summation characteristics in his book. If he were using linear phase filters, there would be no discussion. All symmetrical linear phase HPF/LPF's sum to a flat line, end of story.

What this also means is that phase distortion is the inevitable outcome of minphase filters. Lately there is growing consensus that phase distortion is audible (even though Toole and Linkwitz didn't think so). See the discussion on ASR here and here and pay particular attention to JJ's posts in the later posts in both threads.

1745168469082.png
1745168834575.png


What is more remarkable is that linear phase filters not only do not introduce phase distortion, it can actually correct phase distortion! This graph shows a minimum phase all-pass filter (in green) with a f0 of 200Hz and Q of 0.3. In red, we have Acourate's filter for correcting the phase distortion of my woofer. Note some characteristics of the filter - it is time reversed, and there are little wiggles and bumps to correct every last degree of phase distortion. The outcome is a perfectly flat phase response from the driver - see the graph on the right. Red is before phase linearisation, brown is after.
 
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The state of the DSP market

If you wish to implement DSP, you will need (1) software to design filters, and (2) a way to use these filters for music playback. We will discuss the latter first.

Playback
In turn, there are two ways to implement DSP filters for playback - via a hardware processor (like MiniDSP or AVR's) or via software. All hardware processors on the market (at the time of writing) use IIR filtering, and some of them have a limited number of FIR taps for mixed-phase processing. You typically get 1024 taps per channel, which is nowhere near enough to implement a high resolution FIR filter. The sole exception is the DEQX Premate 8, which has 32768 taps per channel.

Software processing can be FIR or IIR. An example of an IIR processor is @charlielaub GSASysCon for Linux. Most people take advantage of the higher computational power of a PC and use FIR processors. Many examples abound - @HenrikEnquist CamillaDSP, @mitchco Hang Loose Convolver, and so on. As mentioned, Camilla is lightweight enough to run on a Raspberry Pi.

The disadvantage of software processing is that a computing device (PC or Pi) always needs to be in the signal chain and an outboard DAC or interface with the appropriate number of channels needs to be purchased. This is generally not a problem if you are playing back music which is stored locally on a HDD or network, or using streaming services. But it can be difficult (but not impossible) to use external devices, such as HDMI, turntables, and so on. This involves routing audio in software, which is surprisingly complicated. In a typical audio system, when you want to change inputs, you simply switch the dial on the preamp. With a software processing, you may have to dive into settings or configs and re-route audio from one input to another output every time you want to use another device. This is enough of an inconvenience that my wife finds my system impossible to use.

This solution is also not robust. We are only one bad software update away from the signal chain breaking. Right now, there is some weird bug in JRiver that stops me from reliably sending Tidal to it. I have heard that Linux based solutions are more reliable, but I am all thumbs when it comes to Linux. Not possible for one man to know everything.

The advantage of hardware processors is simplicity and robustness. You CAN change inputs at the flick of a switch. You even get a remote control! Some of them are Roon ready or Tidal/Spotify ready. That MiniDSP is going to stay exactly the way it is unless you fiddle with the software settings whilst intoxicated or it gets burned down in a house fire. You don't need to worry about software updates breaking the system. And if you need video, AVR's offer features that are very difficult (if not impossible) to implement on a PC, for example Dolby Atmos decoding. I have been told that it is now possible to decode Atmos on a PC, but I understand that it is not easy.

However, hardware manufacturers often don't tell you what their devices are capable of. Or more precisely, they try to hide the limitations. Take a look at MiniDSP Flex Eight and tell me how many biquads and FIR filter taps it has. Can't find it anywhere? That's because it is hidden deep in the manual. Answer: 10 biquads each for 8 channels, and 4096 taps implemented at the input stage for 2 channels, meaning 2048 taps per channel. And good luck trying to find out how many biquads are in your Marantz AVR. When I met the CEO of Storm Audio, I asked him how many biquads are in his processor. He gave me a blank look, and said he has to ask his chief engineer. I should bloody hope it has a lot of biquads and FIR taps, considering the exorbitant price of the unit.

In short, the concern with hardware processors is not the quality of the DAC's as many people seem to think (especially those of the subjectivist persuasion). We know from Amir's testing that the quality of MiniDSP DAC's is excellent. What I am concerned about is the quality of the DSP. Low SINAD is meaningless if actually using the unit to cascade biquads creates audible quantisation noise. It's not hard for ordinary users to DIY Raspberry Pi solutions capable of FIR filtering, so I wonder why MiniDSP doesn't do it. They would really eat DEQX's lunch - who, IMO, have priced themselves out of the market.

Filter Design Software
The number 1 reason for bad sounding DSP is NOT hardware, NOT whether you choose IIR or FIR, and NOT whether the SINAD is >100 or not. The number 1 reason is user error! If you take bad measurements, try to make inappropriate corrections, you WILL create a filter that sounds horrible. And believe me, I have made my share of horrible sounding filters. In the early days, it was because I did not know what I was doing. These days, it's because I get hare-brained ideas, mistake my stupidity for genius, and go try something inappropriate. "Why hasn't anybody else thought of this??" becomes "they probably have, and realised it's a dumb idea".

But anyway, I digress. The point I am trying to make is that software is very important. Software varies in price, ease of use, and flexibility. If it is difficult to use, you will make mistakes. If it is inflexible, it will prevent you from learning more or "escaping" the software's algorithm.

Unfortunately, most software that aims towards "ease of use" also sacrifice flexibility. For example, Dirac is a wizard-based program that walks you through all the steps required to generate filters. Time alignment is automated, no need to manually inspect the impulse response. But if it gets it wrong, there is no way to use Dirac itself to determine the correct delay, you have to use REW. And what if you don't want to time align, you want to phase align? Answer: REW. You might then be asking yourself why you need to use REW if you paid close to a thousand big ones for Dirac? Umm, that's a question for Dirac :)

REW and Acourate are at the opposite end of the spectrum. Both are very manual tools, and the tools need to be deployed in the correct order and in the correct situation. When I say "manual", I mean that you have to inspect the measurements yourself and decide what you want to do. Acourate has a few more "luxury" features compared to REW, which is why I prefer it. REW can not be used on its own, you need RePhase. Both are extremely flexible, but also close to impossible to use if you do not know what you are doing.

Acourate has a free guide (written by myself) and an excellent book by @mitchco. REW has a lot of videos on Youtube, but unfortunately I disagree with most of them. e.g. I saw someone direct the viewer to invert the response before windowing and minimum phase extraction. Worse, there were user responses saying "Excellent!" and "Thank you!". If you don't know, you can easily be misled. Acourate's procedures can be adapted to REW, you just need to figure out how. I try to explain why these procedures are necessary and in that order in my free guide. In my humble opinion, it is the best guide to REW even though it is written for Acourate, if only because I don't direct the reader to do dumb things like try to invert the non-minimum phase response.

There are more DSP packages like Audiolense, Focus Fidelity, Eclipse Audio's FIR designer, and so on. I have not had a chance to take a close look at these packages offer, so I can't comment on them. All have varying degrees of automation to make it easier for the user, although I disagree with some of those procedures. I would have to buy a software license before I know for sure whether those procedures can be bypassed, but i'm not in the mood to do that so for now, no comment.

In closing, I should note that I have my own biases. I am strongly biased in favour of linear phase FIR's, and manual filter design. But regardless, I have tried to be fair in explaining the differences and drawbacks. There is much more to cover and plenty has been omitted.
 
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Great writeup! KUDOS for taking into account all the aspects

From my end (having 15 years of experience with software based DSP) I would add:
- totally agree on linear phase supremacy. In the last few years I have only been using linear phase crossover and post-EQ, simply love the sound vs. traditional min. phase EQ (probably due to way less post-ringing)
- do try and experiment with various VST plugins, there are real gems there! (e.g. Pro-Q 4, thEQorange, GrandEQ, etc.)
- as a general rule: don't believe us or anybody, do try everything for yourself to see what works best for you
 
Just what I needed. I've learnt a lot. Thanks.

A big part of what I've just leant is: I need to stay away from manual filter design unless I want to expend a LOT of time and effort on the learning curve. :p

Audessey ONE for me. :)
 
Completely over my head :) Bookmarked and I will re-read! Thanks for posting this.
 
Very interesting. Thanks.

I would be most grateful for your opinion on MathAudio RoomEQ if possible. I have been using it for several years, (as a baseline/starting point) alongside a PEQ VST (EasyQ) via Foobar2000. It is free and has improved my sound hugely. I cannot see whether it employs linear filters or otherwise, but I have been very impressed by it's effectiveness and ease of use personally.

Link here.

Most grateful for any info or recommendations .
Thanks again.
 
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Very well written and clear @Keith_W thanks!
 
Where does GLM fit?

I haven't played with it so I don't really know. But when I evaluate a DSP product, what I want to know is:

- What type of DSP filtering does it do? IIR or FIR?
- How many biquads/taps?
- What does the software do? Can you use third party software if you don't like the software they recommend? Are you locked in to their software? Do you need to pay additional license fees (Dirac) or worse, a subscription (anything that uses Audioweaver)?

If they hide this information on their website, I look somewhere else. I need to know exactly what I am buying, and if they don't tell me - I cross it off my list. A bit harsh, I know. But I would expect a website to provide this information.

So you can go to Genelec's website and see what they tell you. Hint, not a lot. It may or may not be a great product, but I am not interested in reading marketing material. I want to see the specs. There are Genelec reps on ASR - if they read this, they should realise that I am not criticising their product since I don't know anything about it. I am criticising their website which does not provide the information that I need!
 
Keith_W

Corporate espionage is so dangerous these days that it's good that companies protect their assets well. And you don't have to know all of them yourself.

Genelec's glm software works like the most powerful electric motor found in the most expensive cars on the market, because it's embedded in the speakers. It's just testing the limits with them.
 
Very interesting. Thanks.

I would be most grateful for your opinion on MathAudio RoomEQ if possible. I have been using it for several years, (as a baseline/starting point) alongside a PEQ VST (EasyQ) via Foobar2000. It is free and has improved my sound hugely. I cannot see whether it employs linear filters or otherwise, but I have been very impressed by it's effectiveness and ease of use personally.

Link here.

Most grateful for any info or recommendations .
Thanks again.

I had a quick look at it. It appears to belong to a class of DSP product that I call "one button DSP". You take the measurements it tells you to take, load them into the software. Then you push a magic button and the correction comes out. User is happy.

Remember what I said about the no. 1 cause of bad DSP is user error? In the case of products like these, you can also add: "how smart is the algorithm?". We know that REW has trouble automatically determining delays if the conditions are not right. I prefer to eyeball the impulse myself to determine the delays, determine for myself what should and should not be corrected, and so on. Algorithms get it wrong, especially if care is not taken when the measurements are done. And the great unknown is the algorithm - without testing it myself under different situations to see where it breaks, I don't know how good it is.

This is great if you don't want to learn the nuts and bolts of DSP. I am sure that products like these (and also Dirac) will make most people happy most of the time. But what if you're not happy with the correction? Your only recourse is to re-measure and repeat the process until you are happy. You will never know what went wrong, why you're not happy, or what you need to do to prevent it from happening again. You learn nothing from the process, except some vague idea that DSP worsens the sound.

So I can't give you specific comments about that product. The webpage doesn't tell me very much, and if I really want to know I have to test it.
 
Thanks. It definitely improves my sound hugely and I guess as it's free, I have no cause for complaint. Would love to see a comparison with DIRAC et al. I use it primarily as a starting point and use EasyQ to tailor to taste. Not sure I'm prepared to invest the time and effort inherent in the more involved manual options quite yet as I realise it's a fairly complex rabbit hole to go down. Will be following this thread with interest, so may dive in in the future however.

Curious if anyone knows anything more about the MathAudio algorithm and how it compares to others?

Thanks again.
 
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Genelec's website and see what they tell you. Hint, not a lot. It may or may not be a great product,
I found a few links about some function in GLM.


I agree with most of it, since there are many black‑box products out there. However, regardless of the form of DRC that meets the SOTA, there are surprisingly many consumers who want a one‑click solution, so those using such one‑click solutions might have a different opinion.
(Of course, putting that aside, the convenience of integrating a company’s own products—like Genelec’s GLM or Neumann’s MA1—is one of the biggest merits for light users.)

Also, user‑friendly one‑click solutions are even used in immersive audio seminars, such as AES Europe 2024, as shown in the links below.
Of course, I prefer a transparent DRC that meets SOTA standards—so I agree.
However, as I mentioned above, there may be quite a few people who don’t share that view.

 
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I'm using MathAudio RoomEQ with Neumann KH310s. I use BK subs as I didn't want to have to pay the premium prices for either the Neumann subs and/or MA1 DRC.
I also wanted a solution that would easily integrate with other equipment, as I frequently switch between amplifiers and speakers within my main system. (ie The KH310s sit on top of my Tannoys and it's a simple switch to change the output between the two.)
MathAudio RoomEQ enables me to switch saved presets/curves for each speaker very easily. Doubtful this would even be possible with MA1.

Very interested to hear if any Genelec/Neumann users have compared results using both GLM/MA1 with the likes of non brand-specific solutions such as MathAudio, Dirac etc...
 
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Very interesting. Thanks.

I would be most grateful for your opinion on MathAudio RoomEQ if possible. I have been using it for several years, (as a baseline/starting point) alongside a PEQ VST (EasyQ) via Foobar2000. It is free and has improved my sound hugely. I cannot see whether it employs linear filters or otherwise, but I have been very impressed by it's effectiveness and ease of use personally.

Link here.

Most grateful for any info or recommendations .
Thanks again.
Quick and dirty that's how it looks like with a modest correction down low:

Edit: the below are invalid, down the thread there is a correct comparison.

Raw is purple,MathEQ is yellow.

unwraped phase.PNG
Unwraped phase after generated min phase


impulse.PNG
Impulse


ETC.PNG
ETC


Step.PNG
Step response


(WARNING: simple,amateur, single point measurement)
 
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I haven't played with it so I don't really know. But when I evaluate a DSP product, what I want to know is:

- What type of DSP filtering does it do? IIR or FIR?
- How many biquads/taps?
- What does the software do? Can you use third party software if you don't like the software they recommend? Are you locked in to their software? Do you need to pay additional license fees (Dirac) or worse, a subscription (anything that uses Audioweaver)?

If they hide this information on their website, I look somewhere else. I need to know exactly what I am buying, and if they don't tell me - I cross it off my list. A bit harsh, I know. But I would expect a website to provide this information.

So you can go to Genelec's website and see what they tell you. Hint, not a lot. It may or may not be a great product, but I am not interested in reading marketing material. I want to see the specs. There are Genelec reps on ASR - if they read this, they should realise that I am not criticising their product since I don't know anything about it. I am criticising their website which does not provide the information that I need!
I would like to hear from any Genelec GLM users, or reps, on some technical information mentioned here. I use their speakers and GLM software myself and not sure exactly how it works technically. Like the results though.
 
I had a quick look at it. It appears to belong to a class of DSP product that I call "one button DSP". You take the measurements it tells you to take, load them into the software. Then you push a magic button and the correction comes out. User is happy.

Remember what I said about the no. 1 cause of bad DSP is user error? In the case of products like these, you can also add: "how smart is the algorithm?". We know that REW has trouble automatically determining delays if the conditions are not right. I prefer to eyeball the impulse myself to determine the delays, determine for myself what should and should not be corrected, and so on. Algorithms get it wrong, especially if care is not taken when the measurements are done. And the great unknown is the algorithm - without testing it myself under different situations to see where it breaks, I don't know how good it is.

This is great if you don't want to learn the nuts and bolts of DSP. I am sure that products like these (and also Dirac) will make most people happy most of the time. But what if you're not happy with the correction? Your only recourse is to re-measure and repeat the process until you are happy. You will never know what went wrong, why you're not happy, or what you need to do to prevent it from happening again. You learn nothing from the process, except some vague idea that DSP worsens the sound.

So I can't give you specific comments about that product. The webpage doesn't tell me very much, and if I really want to know I have to test it.
One of the reasons i use Mathaudio is because of the extreem simplicity ( for tech dummies like me:facepalm:) an it excellent ( undermine acoustic conditions wich requiers Full correction) result. I do know Mathaudio favours Full correction some other solution favours the Schröder correction till let say 400hz. Looking here on ASR it is about 60% favours Schröder 40% Full correction.


Despite limitations of DSP regarding correction range don't you know of developments that are focussing on full correction. ?
 
With MathAudio RoomEQ you can choose what frequency range is corrected quite simply by dragging the mouse upwards, away from the range you want left un-corrected.
I allow full-range correction of my Tannoy V12s, but only below appx 400Hz for my Neumanns.
 
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