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Article:  Integrating Subwoofers with Stereo Mains using Audiolense


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The DEQX HW-SW system will also time and frequency align the speaker as part of their DRC, and it is pretty much an automated process. The company will also logon to your system and set it up for you if you want.

Main listening (small home office):

Main setup: Surge protector +>Isol-8 Mini sub Axis Power Strip/Isolation>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three (on their own electric circuit) >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three .

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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@Ron Scubadiver and @kravi4ka thanks!

 

@firedog I have used DEQX in the past... I had a look through their latest 172 page user manual -  still quite a few manual steps, including figuring out time alignment. Maybe I am missing something? Nonetheless, my goal is to eval as many DSP DRC products as possible. Maybe DEQX is at the Munich show and Chris @The Computer Audiophile can see about getting one for review...

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Mitch,  I have enjoyed reading your DSP pieces, thanks for your effort!

 

I am not familiar with the Lynx Hilo, and the web hasn't been much help yet.  Can you briefly review your setup?  I take it that you have a stereo source file, process it with your filter and digital crossover in JRiver and send 6 signals to the Hilo.  How do you connect the six signals to your Crown/Pass/Rythmik equipment?  I only see balanced, single ended, and headphone outs on the Hilo.  Thanks!

Jim

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Mitch,  I have enjoyed reading your DSP pieces, thanks for your effort!

 

I am not familiar with the Lynx Hilo, and the web hasn't been much help yet.  Can you briefly review your setup?  I take it that you have a stereo source file, process it with your filter and digital crossover in JRiver and send 6 signals to the Hilo.  How do you connect the six signals to your Crown/Pass/Rythmik equipment?  I only see balanced, single ended, and headphone outs on the Hilo.  Thanks!

 

I guess it is simpler than I thought it would be, using the 3- different types of outputs!

Jim

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Thank you Mitch,

 

It's refreshing to be availed of someone's obvious talent, and who is willing to help others, without the obligatory "what's in it for me". You are incredibly generous to share your passion and knowledge with us.

 

In a world where most (if not all) our music is recorded and stored in digital it just makes sense to me to utilise the attributes of that format by using DSP to maximise the potential of the recording to suit both the playback equipment and the room it is located in i.e. sending a correctly analysed and "manicured" digital signal to active speakers and converting back to analogue at the last possible moment - i.e at the driver's individual amplifier.

 

My own experience with DSP is via my current system being a Devialet 200 driving ATC SCM 19 mains with 2 x SVS SB 2000 subs. My room is 20' wide with one speaker and sub against glass doors to a balcony and the other adjacent to an open plan kitchen and dining room... its a BIG awkward space to fill with music. The Devialet DSP system is several years old now and is very simplistic compared to your Audiolenese software, however, it does allow the selection of a variable XO frequency for the subs as well as the ability to add time delay. I agree with you that the subs when correctly set up make a huge difference and despite its obvious limitations the room actually sounds very good.

 

IMO what you are doing is the future for audio - instead of spending obscene amounts of money getting the last 1% out of a device you are using DSP technology to ensure the whole system (including the room) is working together in "concert" and that has to provide the biggest bang for the buck.  

 

My hope is that manufacturers such as Devialet take note of your work and continue to enhance their products by building in DSP software into their products so that that those of us who do not profess your knowledge and are time poor can obtain the obvious benefits of DSP.

 

Thank you again for your wonderful effort to educate us all!

LOUNGE: Mac Mini - Audirvana - Devialet 200 - ATOHM GT1 Speakers

OFFICE : Mac Mini - Audirvana - Benchmark DAC1HDR - ADAM A7 Active Monitors

TRAVEL : MacBook Air - Dragonfly V1.2 DAC - Sennheiser HD 650

BEACH : iPhone 6 - HRT iStreamer DAC - Akimate Micro + powered speakers

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@Archimago Thanks man. Does kick ass... smoothly... with time domain correction. Drums are concussive and solid bass you can feel. Not bad for a ten year old cinema loudspeaker design with updated Pro drivers. I can smooth the response enough to sound high fidelity. I would expect folks to achieve similar or better results using a well designed audiophile loudspeaker. The JBL's are no slouch and will produce a large dynamic sound with low distortion, as they are barely working  in a home environment.

 

Yes, optimization is what it is about. Just like @Ajax says:

 

7 hours ago, Ajax said:

you are using DSP technology to ensure the whole system (including the room) is working together in "concert" and that has to provide the biggest bang for the buck.  

 

Thanks for your kind words and sharing your experience with Devialet. I like your comment above, that's it exactly. Takes a bit of know how and elbow grease, but it is amazing how much sound quality improvement can be gained to ones sounds reproduction system using modern DSP. The results are audibly better sound quality to my ears and measures significantly better in both the frequency and time domain... For not a lot of money. 

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Great article. I am looking forward to a new edition of your excellent book. I bought your previous one even though I dont own Acourate now.  This is the best source of knowlegde about DSP software I know about.  After this article I can not again decide which software I should use: Audiolense or Acourate. At the moment I am using Dirac Live but I need a crossover capability and multisub (4-6) integration.  Unfortunately none of these software can deal with multisubs in a proper way and I am forced to use Multisub optimizer with minidsp just for that.

Macmini/ Jriver MC26 - Audiofire 12 - MSB-MVC-1 volume control - Cinepro 2k6 amp - Geddes Abbey speakers plus 4 x 10" Aurasound subs

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Hi @dziemian thanks for your kind words and book purchase.  Audiolense and Acourate are excellent DSP packages. I will be testing Dirac Live soon. Not sure what you mean by "none of these software can deal with multisubs in a proper way"... 

 

I have looked at Multisub Optimizer (MSO) closely and Audiolense does the same as MSO, but better as it uses FIR filters instead of PEQ's - i.e. Audiolense has considerably more resolution with 65,536 filter taps. Audiolense also performs multi-seat correction. While MSO does perform individual delays, as does Audiolense, it has no facility for time domain correction - i.e.excess phase correction. Or the capability to define a target response or linear phase XO's, etc. I believe MSO is designed to supplement/replace simple global room eq's like Audessey for example. Audiolense has complete frequency and time domain control over each individual sub, including multi-seat optimization.

 

Have a look at @dallasjustice 4 sub setup using Audiolense: https://audiosciencereview.com/forum/index.php?threads/jbl-m2-audiolense-digital-crossovers-w-subs.2369/  His results with Audiolense are excellent. Perhaps Michael can comment or reach out to him via PM to get his thoughts. 

 

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10 hours ago, dziemian said:

 Unfortunately none of these software can deal with multisubs in a proper way and I am forced to use Multisub optimizer with minidsp just for that.

What do you mean by that?  I agree that Dirac does a poor job of integrating subwoofers with stereo playback. However, Acourate and Audiolense can accommodate just about any configuration you want. 

 

MSO is very useful for getting smooth bass frequency response across a wide listening area. Some people believe that is all one needs for optimal results. IMO, time domain matters too. I agree with Mitch that linear phase crossovers and time aligned drivers are needed for best results. Either Acourate or Audiolense can accomplish almost anything you need. There are pros and cons between Acourate and Audiolense. I’ve extensively used both for multiple sub integration. 

 

There are are many different subwoofer setup techniques. I think there are 2 categories:

1.  Mono/summed arrays. 

2.  Stereo sub arrays. 

 

However, there are variations within each category. For example, some mono sub arrays are simply time aligned to seated position. Either Acourate or Audiolense can handle these arrays. All stereo sub arrays should be time aligned to listening position. When I say time aligned, I mean flat group delay throughout the crossover region. As you can see from Mitch’s plots, it’s just about impossible to get flat group delay down to 20hz. It really doesn’t matter that much as long as both subs are consistent and the group delay is consistent throughout the crossover. Stereo subs can get a little more complex though. I personally use a 4 stereo sub array (cascaded subs). I believe Mitch linked to another thread which shows how and why I do that. 

 

Finally, there are the non time aligned subwoofer arrays. These are mono/summed sub arrays. Some folks advocate the use of non-time aligned mono subs. I personally don’t see the advantage of using those types of setups. These include Welti. That’s a different topic.

 

However, there is a very effective mono sub array which is not time aligned. It is called “source/sink.”  It’s mostly done using only two mono subs. The frontwall/midwall sub is the “plane wave.”  It is time aligned with R/L speaker. The rearwall/midwall sub is set to opposite electrical polarity from front sub. It is also delayed so that the plane wave and the rearsub wave meet each other behind seated position. The phase rotation of the rear sub needs to be adjusted using RTA function in REW while both subs are playing a LF pink noise. The phase rotation is carefully dialed in until all the room length modes are eliminated. Source/sink has two huge advantages for those with rectangular rooms who have nasty length modes. 1. When properly setup, it can mostly eliminate all length modes, without any DSP using only two subs. 2.  It will eliminate any rearwall boundary interference at listening position. Most people in rectangular rooms sit behind the room length midpoint. In these cases, the rearwall will likely destructively interfere at a specific frequency based on its distance from listening position in relation to the front wave source distance to listening position. This is called the “Allison effect”. Others call it SBIR. Still others call it a “null.”  They are all the same thing. It is NOT a room mode. Because it is non-minimum phase, DSP can’t fix it. Only speaker placement can overcome this issue. Of course a rearwall sub setup in a “source/sink” array will eliminate this boundary interference. 

 

Back to your question about Acourate vs. Audiolense. The only sub array I know about that Acourate can do which Audiolense cannot do is this “source/sink” array. The reason is that Acourate Convolver can be setup to simultaneously measure two mono subs with delay added to rear sub. OTOH, Audiolense can not measure in this way. Audiolense can only measure one channel at a time. 

 

I’ve tried just about every subwoofer array I’ve described, except MSO.   In my room I’d rank the 4 stereo cascaded sub array first. Second place goes to “source/sink.”  Other rooms are different. There is no ideal or perfect setup.  You have to tryout different arrays in your room, measure them and see what measures (frequency and decay) best. 

 

Subwoofery done right can be a very iterative process. This is true for most any array. Because there may be a lot of move-and-measure, it’s important to have an easy/fast method to loopback measure each array. This is where Audiolense beats Acourate. From the time one setups up a speaker array with crossovers to the time the .cfg files and FIR impulses are in a folder for Jriver/Roon, it may take 5-10 minutes when you get the hang of it. Acourate won’t go that fast. You’ll need to create your own .cfg files, crossovers and the speaker setups in Acourate Convolver will take some serious practice to get really fast. I know Uli can do it very fast. But my brain works much slower. 

 

I think both Acourate and Audiolense are outstanding. I’d say buy both. That’s what I did. I still use both of them; best audio money ever spent. 

 

Michael. 

THINK OUTSIDE THE BOX

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I meant that Dirac Live, Acourate and Audiolense  deal with subs separately. The only software I know of which takes into account all of them is Multisub Optimizer and Dirac Unison. Both softwares use all available drivers to help each other in obtaining good frequency response. MSO works for subs mainly and improves  frequency only, improving phases' feature has already been requested by some users but it seems not so easy. Unison on the other hand work with full spectrum and deals with both frequency and phases. But  as you know it is not available yet. I recommend a good book about the subject i.e. "Multichannel Audio Signal Processing. Room Correction and Sound Perception" by Adrian Bahne. It is a dissertation from University of Uppsala written by one of the people who worked on Dirac Unison.

Macmini/ Jriver MC26 - Audiofire 12 - MSB-MVC-1 volume control - Cinepro 2k6 amp - Geddes Abbey speakers plus 4 x 10" Aurasound subs

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BTW I have a L-shaped room which is not that easy to setup using source/ sink or Dr Floyd's methods. I tried Dr Geddes aproach before but MSO is much better in this regard. 

Macmini/ Jriver MC26 - Audiofire 12 - MSB-MVC-1 volume control - Cinepro 2k6 amp - Geddes Abbey speakers plus 4 x 10" Aurasound subs

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1 hour ago, dziemian said:

I meant that Dirac Live, Acourate and Audiolense  deal with subs separately. The only software I know of which takes into account all of them is Multisub Optimizer and Dirac Unison. Both softwares use all available drivers to help each other in obtaining good frequency response. MSO works for subs mainly and improves  frequency only, improving phases' feature has already been requested by some users but it seems not so easy. Unison on the other hand work with full spectrum and deals with both frequency and phases. But  as you know it is not available yet. I recommend a good book about the subject i.e. "Multichannel Audio Signal Processing. Room Correction and Sound Perception" by Adrian Bahne. It is a dissertation from University of Uppsala written by one of the people who worked on Dirac Unison.

 Understood. Acourate can measure and apply correction to more than one sub. I described this above with regard to the “source/sink” array. 

 

I don’t believe that is a weakness in most cases. Audiolense and Acourate will accurately predict the result. It’s easy to test with loopback measurements. The only time this wouldn’t be the case is in the source/sink setup I described. Otherwise there’s no need to measure subs at the same time.

 

Dirac Unicorn has been discussed for many years now and still nothing in reality. I understand the approach.  I’ve only heard it in Volvos. I think it sounds wonderful. 

THINK OUTSIDE THE BOX

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MSO doesnt measure at the same time. It measured all the subs individually but when calculating filters uses all the measurements. It is done live so you can see how the outcome is changing in time. Sometimes it takes hours. In my case I aplied mso filters for 4 channels in minidsp (4 subs) and then I run Dirac in htpc on mains and  the sub channel in 2.1 system. However I am not able to verify the results in REW as nanoavr hda has only hdmi inputs and I dont know how to make a loopback. I guess you can use the same approach in Audiolense and Acourate.  MSO for multisubs plus minidsp 2x4hd (for example ) on a sub channel in preferred software.

Macmini/ Jriver MC26 - Audiofire 12 - MSB-MVC-1 volume control - Cinepro 2k6 amp - Geddes Abbey speakers plus 4 x 10" Aurasound subs

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2 hours ago, mitchco said:

Hi @ronkuper, I am glad you enjoyed the article.

 

1.0 left + 1.0 right. Simply let the sub handle the full LF part from both speakers. As if it were two subs spaced together.

 

If the SPL and the sound quality to your ears is sufficient, good to go!

 

Kind regards, Mitch

 

Hi @mitchco ,

 

I accidentally posted my subwoofer integration related question in the previous article's thread so moving here :)

 

I'm very happy that you replied but now I'm even more confused with all the conflicting data*, maybe you can help shed some light on this topic. 

 

*On a JRiver thread with Matt they discussed summing - 

https://yabb.jriver.com/interact/index.php?topic=58646.0

 

Quote from one of the guys - 

Quote

I think you would need to do

(0.707946 x L) + (0.707946 x R)

to get a flat frequency response curve when the bass from 2.0 is redirected to 0.1.

Before the redirection you had two output channels that added to each other. After the redirection you have only one output channel so it should receive a louder signal than one of the two outputs channels received before (the above math results ~1.41).  [edited this part a bit]

Of course you can't go over 0 dBFS in the digital domain so you would need to attenuate all channels to compensate.

 

The discussion there continued mainly on summing formulas for 5.1 content so I was left with uncertainty about summing for stereo content.

 

So what is the correct logic?

I would really like to understand the considerations better.

 

 

A bit of context of my setup - 

1. I currently use Roon and Dirac Live 

2. Plan to soon follow your article and try Audiolense 

3. Unlike JRiver Roon doesn't have built-in BM so ATM I need to manually configure BM before Dirac. 

4. I use equalizerAPO for the task (and other tasks such as equal loudness) as it is more flexible than Roon's DSP  

 

 

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Hi @ronkuper this stuff makes my head spin :-)

 

That's an old JRiver thread... I sent a note to Bernt and his view is slightly different than JRiver's on the summing. Here is the note I got back: Imagine the following simple scenario: Playing old Beatles over a 2.1 rig. All bass in the source signal is in the left channel. You have to use the factor 1.0 in the bass offloading to make that sound right. If you use two subs, the factor has to be 0.5 since two subs will play 6 dB louder than one.

 

Imagine another example: Mono played through stereo speakers: The bass from the two speakers combined will have 6dB higher SPL than each alone. In the bass, the signal will be in phase unless a very bad setup. To get the same spl out of a mono sub you need to use the factor 1.0 as above.

 

Another case that may be relevant: Playing a center channel through left and right speaker. This is where they use 0.71 in AC3 filter. To get a correct phantom center in the sweet spot you need to redirect the center channel with a factor of 0.5 to both speakers. It will be just like the mono scenario above. The sum will be 6dB louder than the contribution from just one speaker. Note that I am still talking about a corrected pair of speakers that are practically in phase for the whole bandwidth – in the sweet spot. Outside the sweet spot above some frequency the signals from  two speakers will have a random phase difference. The combined output for that region will as a theoretical rule of thumb be approx. 3dB higher overall than each of the speakers. But there will be plenty of frequencies where the figure is 6 db, and also plenty of frequencies where the two speakers cancel each other out. If the listening seats are spread out from  left to right, the best compromise might be to use a factor that is higher than 0.5, but it will be substantially lower than 0.75 (sqrt of 0.5). But I wouldn’t bet much money against using 5.0 here too. Those on the “left wing” will get extra spl from the left speaker and vice versa on the right wing…

 

The errors in AC3 filter will amplify the center channel and the bass above what’s correct and neutral This will probably sound sweet to a lot of listeners, and that may explain why the error prevails (if it does).

 

My own experience mirrors the last paragraph above. If I use 5.1 with JRSS surround processing, the center channel and bass is a bit above what I would normally expect. If I use 2 channels (inside a 5.1 channel container) the output does not have the center or bass channels amplified. At least that is how I remember it, and watching movies, I do like the former, but can switch to the latter for a more neutral sound.

 

You can choose either way and your ears can be the judge of which one you like better. 

 

If you want to drill down further, there is a section in the Audiolense help file on bass management. Also, it may be good to post to the Audiolense support forum to get other user experiences as well.

 

Kind regards, Mitch

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Hi @3ll3d00d (Matt?), I had a look at MSO and docs. Tried to find some of the referenced material that @dziemian pointed out, as well as the optimization algorithm: JADE: Adaptive Differential Evolution with Optional External Archive referenced in the MSO help file. I have not had a chance to look in detail, but it is different than Audiolense. I would love to see some time domain measurements like step response and group delay as measured at the LP for comparison... 

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Hi @mitchco , you have identified me correctly :) In case it's not clear from the docs, MSO is basically an automated search algorithm which varies the filter parameters in each iteration and then tests the results against a score (where the score is IIRC the RMS error against the target across the listening positions subject to the per seat weights), the top score is selected as the solution. This repeats for as long as you like to run the algorithm for and the time required can be very long indeed (e.g. if you have a lot of subs and/or give each channel a lot of filters). 

 

IME the final outcome (with respect to step response etc) is similar to any other method because I run acourate on top of MSO though I also take care with the constraints put on the MSO filter ranges (inc delay). I'm not sure whether I have good comparison graphs for my current setup, the sub end of it (e.g. from a group delay point of view) basically looks like a textbook (minimum phase) response. 

 

My workflow is basically

 

- measure subs at various positions as usual for MSO

- configure MSO with those measurements, listening positions and with whatever filters you want to let it play with

- run that for as long as you see fit and as many times as you see fit (til you get a result that looks promising)

- translate the resulting filters into an appropriate format for acourate (I used rephase for this, life is too short to manually create those filters in acourate and convolve them one by one!), convolve with your sub XO(s) 

- now run acourate as normal

 

i.e. MSO is playing the "driver linearisation" role in this setup where the driver is actually composed of many independent subs. FWIW I decided to anchor the MSO timing relative to my front subs (i.e. it could vary the delay on the other subs but not what I consider to be my "main" sub which is at the front) and then I aligned the result to my mains.

 

It's quite time consuming but the results are excellent IME. 

 

Does audiolense allow you to add arbitrary filters to the XO (or via prefilters) like acourate does? If so I would think the same basic process would work. 


 

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