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ASR Acourate users

Keith_W

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There are a number of threads about Acourate on ASR, but there does not seem to be an "Acourate user group". There is an actual Acourate User Group, but that place is pretty quiet and full of existing Acourate users. I thought I would create a thread so that we can discuss and share what we are doing and get some new ideas on how to use this powerful software, and maybe spread the word to encourage other users to try.

It is my experience that even seasoned audiophiles are blown away when they encounter a fully corrected DSP system for the first time. We all know that speakers corrupt the signal, and rooms corrupt the speaker even more. Software like Acourate allows you to remove these influences and achieve transparency, dynamics, and impact like you have never heard. Veils were lifted, wife heard it in the kitchen, etc. - all that really happens. Forget those miniscule differences in SINAD between DAC's, the objective and subjective benefit of Acourate is magnitudes above that, and I think that every ASR member who is serious about audio should have some kind of DSP in their system.

Acourate is described by the author as a "digital audio toolbox". Although it does do room correction, it is much more than that. There are other room correction products on the market, e.g. Audiolense and Dirac Live. Although I do not own licenses for AL or Dirac, I have friends who use it and I have seen what they do. AL and Dirac are very much focused on crossover generation and room correction. They do a good job, and they do it quickly and conveniently. For comparison, I saw a friend generate a set of correction filters in Audiolense, and it took him 15 minutes from start to finish, from setting up the microphone to filters ready to be loaded into the convolver. I was very much impressed by the speed and ease of use of the software.

However, Acourate is not like that. While it does have room correction macros that provide some degree of automation, performing almost any other function is entirely manual, and requires your participation and interpretation of what the software is telling you. It really is a toolbox, in that you have a bunch of various tools that you have to learn to use and deploy at the right time, a bit like deciding whether you need a screw or a nail to do the job. While both will nominally hold two pieces of wood together, they have advantages and disadvantages and it is up to you to decide which is better. One example I recently encountered in Acourate was when I found that my tweeter had a 90 degree phase lag. My solution was to apply the Hilbert transform three times to rotate it back to position. Uli's solution was to invert the polarity of the tweeter and correct the measured phase lag via a reverse all pass filter.

There are other audio toolboxes as well, such as rePhase, OpenDRC, and others - I have no experience with those, and the intention of this thread is not to discuss other types of room correction software. This is for what Acourate can do, and how to use Acourate only. I politely request that discussion of other types of software should be limited to features that Acourate does not have, or better ways of doing things.

To get the full potential from Acourate, it is best to have full control of each individual driver in your speaker system. In my case, I already own the speaker. My solution was to bypass the internal passive crossovers and solder the cables from the drivers directly to the binding posts. Each driver has its own amplifier and DAC channel, and each DAC channel has an individual correction filter designed with the help of Acourate. I know of another person with a DIY cardioid speaker emulating Kii 3's with 30 drivers, 30 DAC channels, 30 amp channels, and filters for each individual driver made by Acourate. I am also aware that Acourate can work with a passive filter in place, but I do not know how well it would work since I do not have direct first-hand experience.

Here is the thing I love most about Acourate. Unlike ANY other audio upgrade, once you pay the cost of entry (a modest sum of €340 for EU countries, and €286 for everyone else), all upgrades are FREE. Since Uli introduced the first version of Acourate in 2003 (?) there has been one version change, the only time he has ever asked for more money from his users. And even then the upgrade was optional, your old copy of Acourate would continue to work. The other thing that is free are refinements to the way you use the software - because I have learnt a lot since the start of this journey, the measured and audible performance of my system has improved - all without me having to spend a cent. Improving the accuracy of your workflow, refining the way you use this software, and rethinking the way you do things WILL improve the sound objectively and subjectively, which is yet another reason Acourate users should be discussing this.

Unfortunately, for such a complex and powerful piece of software, online resources and even an explanation of what all the functions do is frustratingly lacking. I have to learn by trying out different functions to see if they work, and even then I have no idea of what side effects I may be causing. One aim of this thread is to help complete that information.

These are the corrections I have implemented in my current system:

- Crossovers. Acourate allows the creation of a number of crossover types, including Linkwitz-Riley, Neville-Thiele, Butterworth, and some exotic types like Horbach-Keele and some of Uli's new crossovers which I have never heard of. I have experimented with a number of these crossovers, and I would be keen to discuss with other users the pros and cons of different types, as well as your subjective listening impressions of these crossovers.

- Driver linearization. Uli has a new method of driver linearization which involves convolution of a reverse all pass filter which to my knowledge has not been described anywhere else. I will post the method shortly.

- Time alignment. Although the time alignment procedure in Acourate is well described, there are several insights which I learnt recently that I will share in an upcoming post.

- Virtual Bass Array. I have done some experiments with this, which has been discussed in a separate thread on ASR.

- Bass phased array. I am currently experimenting with this. Discussion to follow once I have results.

- Room Correction with different target curves and settings. There are different strategies for room correction (e.g. up to Schroder, or full range, with harder or softer settings, etc). I have my own thoughts on target curves and I would be keen to hear other opinions. I am still experimenting with "hard" and "soft" settings (e.g. FDW 30/30 vs. FDW 5/5) and my opinion of the subjective sound quality is still out. I am undecided. Let us discuss.

- Interchannel Phase Alignment (ICPA) - a new feature introduced in Acourate v2.0 a few years ago. This aligns the phase of the left and right channels. Although I can see measured improvements, I am not sure I can hear a difference. Keen to hear thoughts on this.

Acourate resources:
- Accurate Sound Reproduction using DSP (Kindle, paperback) by Mitch Barnett (@mitchco on ASR). This is essential reading for every Acourate user.
- Acourate Digital Room and Loudspeaker Correction Software Walkthrough by @mitchco on Audiophile Style, 2013.
- Advanced Acourate Digital XO Time Alignment Driver Linearization Walkthrough by @mitchco on Audiophile Style, 2013.
- Understanding the "state of the art" of Digital Room Correction video on Youtube by @mitchco 2021, with ASR discussion thread (sadly closed)
- Some basic articles and white papers hosted on Acourate.com by Dr. Uli Brueggemann, author of Acourate (@UliBru on ASR)
- Archimago's Musings in 2015 and 2019 update, by @Archimago
- Digital Room Correction HK by ACK Cheng
- Acourate User Group at the Audio Vero website.
- Acourate Wiki (sadly very incomplete)
 
I suspect I might be talking to myself here. But anyway, I had a discussion with Uli about a new driver linearization technique that also flattens phase by convolving a Reverse All Pass filter into the correction. I believe that this method has not been described anywhere online. As a teaser, this is what it does:

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Red = uncorrected, Green = corrected and convolved with crossover. These are actual before and after verification measurements, and not a sim. You can see how the phase angle remains absolutely flat, never deviating from 0 degrees, even through the crossover.

Uli sent me the instructions on how to do it. I have re-organized what he said, and inserted some screenshots, Mitch style, to make the method easier to follow. Any mistakes or errors are mine.

STEP 1: Create Crossovers

1. Create a working directory so that it looks like this:

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2. Create your crossovers (Generate-Crossover) and save the raw, unmanipulated crossovers into "00 Naked XO". Uli has persuaded me to move away from NT2 crossovers, and I am now using his new "UB jPol 11" crossover, 1st order. Separate discussion to follow.

Step 2: Measure and Create Filters

3. Set one of the drivers as your project working directory.

4. Logsweep Recorder to record your Pulse48L (work on one driver at a time). The focus is to only measure within the range of the intended crossover, usually 1-2 octaves before the corner frequency depending on the crossover configuration. Load Pulse48L into Curve 1, and note its maximum and minimum gain (in this case, max of 20.655 @ 606.445Hz, and min of 14dB @ 3500Hz; see top right panel). DO NOT MOVE YOUR MICROPHONE until you have completed your verification measurement and you are 100% happy with the result.

The principle of linearization is to avoid too much gain loss through the crossover by correcting too large a range of volume. Uli suggests a maximum of 6dB should be corrected. However, this method involves magnitude limitation followed by normalization, which compensates for the gain loss to an extent, so perhaps up to 10dB can be corrected. This is valid for bandpass filters, for low pass or high pass filters (with rising volume at each extreme frequency), we may need to limit the range of correction, depending on the circumstance - e.g. there may not be any point sacrificing tweeter gain to correct a rising response above 20kHz, since it is less audible.

- TD-Functions - Gain. We are going to add or subtract the gain to avoid over-correcting the gain. In this case, it was 20.655dB, with a minimum of 14dB. So we accept a 10dB gain correction range. I subtracted 10.655dB to bring the maximum gain down to 10dB. This step might need several iterations (see step 6). As Uli said to me in his email, "IMPORTANT: the proper amount of correction is selected by "feeling".

5. Make a linearization filter. All these steps apply to Curve 2 ONLY.
- Copy Curve 1 into Curve 2. (Ctrl-C, 2).
- FD-Functions - Magnitude Inversion (linear phase) into Curve 2. This mirrors Curve 1 along the 0dB axis.
- FD-Functions - Magnitude Limiter 0 into Curve 2. The result is always minimum phase.
- TD-Functions - Frequency Dependent Window (F3): 15/15, result into Curve 2. You may prefer a softer correction with FDW 10/10.
- Save Curve 2, I called mine "LinearizationL.dbl". At this stage your screen should look like this:

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6. Linearize your measurement and use it as a guide slope to help create the All Pass Filter. Convolve Curve 1 (raw measurement) into Curve 2 (linearization filter). Result into curve 3. Hit the "PK" button (you will find it at the bottom of the phase window) to reveal the Peak. This is what you should get:

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Now study your guide slope, if there is significant phase lag (i.e. most of the phase angle is below zero), you will not be able to correctly create the All Pass filter. It is important to note that the peak of Curve 3 is centered at Sample 6000. Left-Right click on the Time display, and make sure Max is at 6000. If it isn't, then adjust the center in the box circled. Sometimes unconventional thinking is required - e.g. my tweeters had a 180deg phase lag. The solution was to invert the tweeter at the amplifier. To avoid having to re-measure, the inversion was simulated in Acourate and the filter was applied.

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7. Create your Reverse All Pass Filter. Set Active Curve 4 (it should currently be blank) and hit the IIR button (or Generate - IIR Filter).
- Play with the f(0) and Q until the curve reasonably matches the guide slope - see blue line in the above diagram.
- Once satisfied, TD-Functions - Reversion (F12) to invert the All Pass filter. This will correct the guide slope.
- Save this as "RevAP-L".

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8. Create the correction filter. Convolve the linearized measurement (curve 3) with the reverse all pass filter (curve 4); result into Curve 5. It should show some improvement. In the above image you can see that the frequency response (in blue) has been flattened compared to the initial measurement (in red). The phase angle has also shown improvement.

9. Load the raw crossover (in this case XO3L48.dbl) into Curve 6. Now we are going to complete all the operations into the raw crossover and turn it into a driver correction filter.
- Convolve Curve 5 (correction filter) with the crossover, result into Curve 1.
- Set Curve 1 Conv4_6 active: FD-Functions - Magnitude Normalization
- TD-Functions - CutNWindow; Cut length 65536, position 65536, result into Curve 1.
- Save Curve 1 as XO3L48.dbl

STEP 3: Verification

10. Now we are going to verify that the driver correction filter is working as intended.
- Create a "Verify" subfolder in your current working directory. Change workspace to this folder.
- Set active Curve 1. File-Create Mono .WAV. I called it XO3L48.wav
- Logsweep recorder, and make sure you load XO3L48.wav as a filter!
- Perform your measurement and compare the before and after.

STEP 4: Time Alignment

11. After you have completed linearization of all drivers, proceed to time alignment using your preferred method. NOTE that the reverse all pass filter will affect your time alignment. A normal All Pass filter causes propagation delay. A reverse all pass filter causes a negative delay - i.e. it moves the impulse forward in time. So do not be surprised if all your previously measured delays are thrown off. As an example, without the reverse AP filter, my subwoofers are about 200 samples delayed. After the reverse AP filter was applied, it is now 37 samples delayed.

12. After all these steps are completed, you have a complete set of filters ready for application of room correction and target curves.

Listening Impressions

I generated a new set of filters using the same target curve that I normally use to ensure that the only variable that has changed is the driver linearization technique. The old crossovers that I was comparing to have also had driver linearization, but these were performed using the method described in Mitch's book. Those of you who are familiar with the book know that it does not include the reverse all pass filter in the driver linearization. These old filters that I made already sound very good, and I wondered how the sound could possibly improve from this because I did not think it was possible.

With the old filters, the crossover points were difficult to detect, and in fact I thought I could not hear them. Careful listening with the new filters makes me realize that I can actually hear the crossover points in my old filters, although the difference is fairly subtle - but according to the maxim "once seen, can not be unseen", now that I know what to listen for, I can hear the crossover points in my old filters reliably consistently now. The smoothness of the filters generated by this method is pretty unbelievable. NOW I have a new standard for "can not possibly be improved". The other noticeable difference was clarity - with the old filters there was a bit of smudging here and there (in hindsight, probably at the crossover points) but now everything sounds clean top to bottom.

The other impressive improvement was with the impact of transients, although I am more inclined to credit this to improved time alignment rather than linearizing the phase angle. I use a recording of Japanese drums for this, and when there is a big whack on a Taiko drum, you can hear the skin of the drum, the huge bass transient, and rapid decay back to silence. To me, I had to rub my eyes (and ears) in disbelief as to how realistic it is.
 
Hi Keith, I think it's great if you make acourate a little more known.
I've been using acourate since 2011 and am totally thrilled with the results. For me it was the best investment in my audio career. Since I live in Germany, I also had the opportunity to participate in one of Uli's workshops. You can also let acourate generate the filters at the beginning with the standard settings. That gives already good improvements.
For xover creation I have once made a contribution in a German forum Xover step by step
 
I believe I’ve never seen an Acourate spectrogram, decay and waterfall plot. I’m curious if it simply does not have these graphical displays as a feature…
 
There is an Add on Program STransform

The program for the Stockwell transformation allows the representation of a pulse response (length 65536 samples) in the amplitude-frequency-time diagram.
 
There is an Add on Program STransform

The program for the Stockwell transformation allows the representation of a pulse response (length 65536 samples) in the amplitude-frequency-time diagram.

Any examples?
 

Yes, it will achieve the same outcome. But Uli has changed his method slightly. He sent me an email with the procedure. I think Uli assumes that you are as smart as he is and you can work out what to do. But I personally need a bit more hand holding and I would wager that a lot of people need that too. With Uli, every word he writes is important because he writes in an information dense style. So I wrote it up Mitch style to make it more readable.
 
Can you highlight any differences? My current setup used that other thread as a guide I think

The main difference is that Uli now recommends driver linearization before designing the reverse all pass filter, as driver linearization can change the phase response.
 
I had studied the process of creating filters with Acourate a while ago (when I was considering buying it) and it seemed very teutonic to me...

Conceptually this involves creating a freq filter (crossover) for each driver, a band pass to linearize driver+crossover phase and magnitude, and then time aligning the drivers.
Then there is the precise phase alignment between the two (or more) speakers to enhance the stereo image.
Then room is corrected with additional magnitude (and maybe phase) compensation and reflection cancellation.
At the end all of these filters combine into a single filter per logical channel.
Again, I mean the conceptual steps, not the practical ones.

Essentially, it's the same things other software does... with automated steps and pre-set parameters to suit common usage. Obviously Dirac does not manage various drivers but only the subwoofers (in the Bass Control version) and is not tweakable like Audiolese or Acourate.
But mathematically it does the same steps in theory. I see no other way to achieve the goal...

Perhaps even the phase alignment between the speakers is peculiar to Acourate, but I think there are variables that weigh much more (room) in the correctness of the stereo image.
And if we want to make ends meet, there would be other variables that influence and yet are not corrected. I am reminded of amplitude linearity, thermal effects, etc... but who knows how influential they are on the experience.

It is definitely a toolbox in short.

Surely the whole Teutonic procedure is very precise in the end. But after all, an automatic procedure doesn't provide such a different result... for months of your life spared in studies and trials.
 
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The main difference is that Uli now recommends driver linearization before designing the reverse all pass filter, as driver linearization can change the phase response.

Ok I see, I do that anyway so no difference for me then.

I didn't get why you need to iterate over gain btw, don't you just normalise it? What are you trying to achieve here?
 
I didn't get why you need to iterate over gain btw, don't you just normalise it? What are you trying to achieve here?

You are trying to avoid excessive loss of gain through your crossover. You want to choose a gain setting that gives you maximum correction but minimum gain loss. And you may have to decide whether you would rather have gain loss or correction. Or decide to forgo some correction in exchange for gain for things which may not be audible, e.g. at the frequency extremes.
 
I wonder what response you are trying to linearise where that becomes an issue, i.e. some response with unusually large peaks/troughs
 
I recently made an impulsive purchase of Acourate software after reading many positive reviews and seeing the significant improvements others experienced with their equipment. I also bought Mitch Barnett's book due to the feedback. I call it impulsive because when I began reading Mitch's book, I hit a snag. The microphone I had bought for optimizing my critical listening room with REW and rePhase was the UMIK-1. Though I tried using REW and rePhase, I was unsuccessful. That's when I decided to give Acourate a shot, which is why I have this microphone.

Mitch Barnett's book, published in 2016, contains some links to resources that no longer exist and some steps are different from current UI and you need to assume whats the new name for the tool discussed. At the end of the first exercise, titled "Quick Start Guide", I applied the filters to Roon, and my system lost detail and bass. The overall sound quality deteriorated. I made a few more attempts, tweaking parameters without fully grasping their effects. I might have set my expectations too high, expecting a night and day difference. I'm unsure if my disappointment stems from my lack of understanding or from not knowing what to listen for in the resultant sound. My system already sounds good without any modifications, but I was hoping for a revelation in improvement.

Upon further exploration, I found out that Acourate and Mitch's book are primarily designed for those willing to modify their speakers for tri-amplification, and remove their passive crossover which is a must to unlock fully potential that Acourate can give you. I don't wish to do this with my speakers (Yamaha NS-1000, Infinity Kappa 8, Celestion SL-700). This has been a source of frustration. Correct me if I'm wrong, but it seems that Acourate and Mitch's guide aren't suitable for my traditional setup unless I'm willing to modify my speakers for tri-amplification. Admittedly, I'm speaking from limited knowledge since I bought the software just 4 days ago.

Uli has been responsive to all my email inquiries and has been very helpful and prompt. I'm hesitant to bombard him with more questions without doing my research first. But at this point, I feel like I wasted my money on this solution and I feel I can’t try any other paths since I’ve been fully working on make it work during last 4 days. If anyone believes I can use Acourate to linearize drivers, correct some aspects of my crossover and time alignment, or integrate a single subwoofer, please let me know. I’m focusing only on my Yamaha NS-1000 before moving to other listening rooms.
 
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You are not able to linearize drivers, perform time alignment, or correct your crossover if you do not have individual control of each driver - i.e. an active speaker with a crossover created by Acourate.

You are able to perform overall speaker correction and integrate a subwoofer into the system however.

Re: worse sound quality after using Acourate ... I totally believe it. There is a learning curve involved, and if you mess up somewhere you will get garbage results. Acourate is capable of giving you incredible results, but it is equally capable of destroying your sound if you do not use it properly.

It all starts by taking proper measurements. Although it is not mentioned in Mitch's book, I feel that Acourate's approach is to start with the ideal speaker, and then integrate it into the room. This involves two major steps - the first step is to measure and correct your speaker under anechoic conditions, or as close to anechoic as you can get (measure your speakers outside, 1m from the tweeter axis). Perform an overall correction for your speaker so that it is as ideal as you can make it. Then take it into your room and repeat the measurement from your listening position.

BTW: I did not get good results with Acourate for months. I had multiple learning curves - learning to use the software, learning loudspeaker engineering, learning about room acoustics. This was a major bummer and huge source of anxiety because I had bitten the bullet and converted my speakers to active, and spent a lot of money buying amps and extra DAC channels and reconfiguring my system. If you have knowledge in those other aspects, you will have considerably fewer learning curves to overcome.
 
I had bitten the bullet and converted my speakers to active
It would help if you mentioned this in post 1 or 2. I was thinking you must be full active to get the benefits you describe, otherwise you are just "fighting" the passive crossover(s).

Also if you could mention that the first graph in your screenshots is magnitude (dB) and the second phase (degrees).

Acourate is described by the author as a "digital audio toolbox".
It seems quite capable, but it is very "proper" and expensive.

By proper I mean the way it works in samples and with functions, whereas something like REW turns samples in to time for convenience, and functions are just things you click/do (tick invert, set time offset rather than do "rotations", do arithmetic, etc).

And by being expensive it has probably hampered its own reach (adoption within the DIY community). Between REW and RePhase, it seems many of the same things can be achieved for free now?

Uli has persuaded me to move away from NT2 crossovers, and I am now using his new "UB jPol 11" crossover, 1st order. Separate discussion to follow.
I am keen to see that info.

If I want pretty looking charts to please the demanding ASR crowd, I crack out REW and make a separate measurement.
Couldn't you just export the measurement / impulse, and then import that in to REW?

modify their speakers for tri-amplification, and remove their passive crossover which is a must to unlock fully potential that Acourate can give you
You don't have to tri-amp, you could just bi-amp. A lot of a speakers group delay is because of the high and low pass filters applied to the bass driver(s), so if you can control the bass separately to the mid/high driver(s), you can still get benefits. So that's 4 channels of output. And if you have subs, you need at least 1 more channel.

BTW, have you seen dualazmak's thread? He has actively amped his NS-1000's and says the results are great!
 
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