Article: Integrating Subwoofers with Stereo Mains using Audiolense

[mitchco]

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[mitchco]

@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...

[mitchco]

@james45974 Thanks! Details of setup are in previous article, but yes, balanced line out to the Crown, single ended Monitor out to the Pass, and headphone stereo out to the Rythmik's. No digital or analog attenuation. No hum and just faint hiss when ear pressed up to the waveguides. 

[mitchco]

@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. 

[mitchco]

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. 

 

[mitchco]

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

[mitchco]

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... 

[mitchco]

Hi @3ll3d00d Cool. I will have to look into MSO more, as it seems like a an interesting approach. Thanks for the info Matt. Wrt Audiolense and prefilter, no such feature at this time.

 

@R1200CL Thanks. Wrt Roon, I believe the answer is yes: https://community.roonlabs.com/t/audiolense-convolution-filters-in-roon-resolved-build-298/35603 @dallasjustice may be able to say more.

[mitchco]

Hi @ronkuper I would let Audiolense generate the FIR filters and use them in JRiver with no manual intervention. I have not run into any digital clipping. Cheers, Mitch

[mitchco]

Hello, @RobertinMn, re: ... I maybe wrong here... Yes, you are incorrect. This is an active triamp setup. You can read the gear list here:

 

The added subs make it a triamp setup from the biamp article above. Both these articles on Audiolense, and if you search on CA, two articles on Acourate, show you what digital XO is all about and what can be achieved. I have used electronic XO's for decades as well, including being a FOH sound engineer touring. Digital XO using software and a computer is the next evolution of active cross over technology. One can also time align and linearize the speaker drivers, correct for excess phase and precision eq the system. There is no passive xo in any of these systems.

 

Do you have a measurement mic? A computer? What about an AD/DA converter? If so, download REW acoustic measurement software. Place the measurement mic at the listening position, calibrate REW for 83 dB SPL C weighting, slow integration. Then take some measurements. My book has all of the details on how to do this... Once you have done this, then we can share our measurements in a common format and compare...

 

Kind regards Mitch

[mitchco]

Robert, Audiolense does everything the Rane hardware device does, and more, in a software program running on a computer. This site is called Computer Audiophile. There is no box like a Rotel or some AVR/Receiver hardware with a firmware solution like Audessey in this article. There is no "one surround sound kit." In my pic is a computer, like your Mac, but a Wintel box running Windows 10. It just looks like a receiver...

 

I use JRiver Media Center software program to play movies and music. Audiolense measures, analyses, and designs the XO's, time alignment, excess phase correction, frequencies eq etc., all in a commercial software program. The generated output is a 64 bit linear phase FIR filter which is hosted in JRiver's convolution engine in which  the music is convolved with in real time. Therefore, the music arriving at ones ears is time aligned, phase coherent. and frequency response shaped to a known industry frequency response target (e.g. Olive and Toole). This is all in the article...

 

JRiver and the computer are connected via ASIO/USB to a 6 channel DAC and the analog outputs are direct input to 6 separate amplifiers, of different types. The volume is controlled digitally via JRiver's software program. There is no hardware preamp.

 

I think you are stuck in a loop on Class D amps driving subs. If you spend a few moments and continue reading the article, you will see there is an acoustic measurement showing a flat from 12 Hz on up frequency response, matching the preferred target response in both the frequency and time domain. This model Rythmik sub is designed for music, with a very flat measured frequency response. These are not HT subs and there is no boom.

 

How about showing  us an acoustic measurement of your system? How about a measurement of the impulse, displaying a step response like I have above? This will verify that you are listening to a time aligned system. How about a measured frequency response at the listening position? Let's see what you really have so we can compare apples to apples. Use REW, as it is free, but mostly importantly, it is a highly regarded acoustic measurement software program that will run on your Mac and makes it easy to exchange measurements and produce overlays for comparison. Show me the money!

[mitchco]

Hi Peter,

 

Nice gear! And thanks for the information. That's not preringing you are seeing, but the result of using a linear phase target, which does looks correct. Try this, load your linear phase target, don't generate a correction, turn off all chart view details, but leave the target button enabled. Switch to the impulse view and then select from the Analysis menu, Simulation, Step Response Simulation plus target and you will see what I mean...

 

I would use the same target, then click new target and switch to a minimum phase target, save that and use that for your correction. You can check what that looks like with the procedure above before you generate a correction. Then you should see a nice right triangle step response in the simulation. Give that a listen too. I ended up liking a mixed phase target the best. 80% minphase on bass and 100% on top. Look for mixphase menu item in the target designer.

 

Up to you, but I would (initially) turn off any phase adjusts on the subs and let Audiolense do it's thing. The simulated step response looks good over time and you will see with a minphase target, the right triangle step will be there. Also, I would possibly drop the overall subwoofer levels down a bit to better match the level in the mains. You kinda want to shape the response with levels before you add a target. That way, there isn't so much filter insertion loss as the filter needs to attenuate the overall sub level. to be in line with the mains...

 

Under the measurement menu, you should see Automatic Polarity Correction enabled, so no worries on polarity and won't cause any issues, regardless of polarity of subs.

 

Under the Correction menu, is the Correction Procedure Designer. Click on that and select TTD measurement and click on new procedure and give it a name.  I would uncheck Prevent treble and bass bass boost. I would up the Max correct boost to 12 dB. I would turn on TTD correction per driver in addition to TTD correction. From there I would enter some values in the TTD subwindow only. This where you want to play around a bit. Try some small values like 3/2 then 6/3 and note the frequency and step response differences and then give the filters a listen. The longer correction in the bass, will tighten up the bass considerably and may get rid of that dip you have, which is not likely audible anyway.

 

Have a look at the manual for XO help with respect to width of the overlap. I went with the default values, but it is another area for a bit of experimentation.

 

Lastly, you may want to reach out to Bernt on the Audiolense support forum with respect how to best handle your 3 subs from an Audiolense configuration point of view. What is the recommended approach and tradeoffs... 

 

Your results are looking good! Just a bit more fine tuning with the info above should get you most of the way there.

 

Hope that helps.

 

Mitch

[mitchco]

19 hours ago, Ralf11 said:

Any thoughts on how well this would work if Magnepans were used as the main speakers?

 

Yes, works just fine. There are folks on the Audiolense support forum that have used this for both Maggies and electrostatic panels. Also, one of the reviewers of my book at Amazon uses it on an Open Baffle design...

[mitchco]

Hi @blue2 Thanks for the kind words. I see no-one has got back to you on this. Purchasing a calibrated measurement mic and REW is excellent acoustic measurement software to get you going, is a great idea. I don't know how you are going to get around the 2 sounds devices as seen by the computer though... I don't have any experience with Audirvana or HQPlayer,... However, getting the analog split off your preamp to the subs should work. With REW you will be able to assess your setup and then make a plan from there... Of course, I recommend Audiolense as the fastest way to integrate the subs with some room correction...

[mitchco]

Hi @Trdat, sorry for the delay as I am not getting email notifications on new content...

 

To answer your questions. These are more along the lines of speaker design than room correction...

 

1).  re crossover point. Without seeing the sweeps of both the subs and Deltalites, it is almost impossible to recommend a crossover point. But, having said that if you have an f3 of 51 Hz then it would be around there. Just make sure the subs are actually "subs" and can make it down to or below 20 Hz. In addition, I would use a room mode calculator to identify your room modes and taking into consideration the sweeps of the drivers, try and cross between room modes for the best integration. My 4722's f3 is 40 Hz, but I have a room mode there, so I crossed at 46 Hz, which is in-between room modes and got a much better measuring and sounding integration.

 

2). Yes, typically a 2" CD would go lower than a 1" CD. However, Earl Geddes used the D250 in his systems and crossed at around 800 Hz I believe. Again, you would need to take sweeps of the Deltalite and DE250 and see where a good XO point might be and use your ears to see which you like better, which seems you already have done that. For sure the 15" is likely beaming, but if it is not bothering you, then what me worry :-)

 

[mitchco]

Hi @Iansr yes, you want to get the sub(s) level balanced with the rest of the mains to be relatively the same level to match your target curve before applying DSP. Same applies for the rest of an active system using digital XO with an amp per driver.

 

In my triamp system, I try and balance all driver levels to more or less be representative of the target curve I am going to apply. Why? I don't want unnecessary attenuation in the correction filter because I did not have one (or more) of the drivers level matched. For example, if I had the subs turned up too loud, then in order to hit the target curve, the DSP has to attenuate the subs by X db overall to match the target curve with the rest of the drivers. So I end up with a filter that has way more insertion loss then it should. 

 

Audiolense only cuts frequencies and does not boost. Look at the correction curve in Audiolense and you can see it never exceeds 0 dBFS. It is always a good idea to look at the correction filter anyway to see if one is passing excessive subsonic or utlrasonic energy aside from too much correction. Overcorrection is a thing.

 

Good luck!

[mitchco]

You want to balance everything as best as possible towards whatever target curve you are interested in before AL. Otherwise, you will be getting too much filter attenuation. So wherever you can make gain adjustments, using line level attenuation, amp input level controls, or Lpad's on the high sensitivity speaker. If it is a compression driver, don't forget a protection cap in series.

[mitchco]

Cheers@asdf1000Looking good! I bet sounds good ;-)  No LFE for me. Best to XO in-between room modes if you can...

[mitchco]

23 hours ago, asdf1000 said:

I've ripped out the 50 year passive crossover in a pair of ESS Fortura 10 speakers, which feature gorgeous Heil AMT tweeters. Also replaced with new ESS woofers that have same frequency response as the original.

 

A pair of SVS SB-2000's.

 

Audiolense created all the files of course. A lot fo measurements in REW to find optimal crossover. Funnily enough best was the original crossover.

 

Roon -> HQPlayer doing all the DSP including convolution -> Focusrite 18i20 Gen3 -> pair of Denon amps & pair of subs

 

Sounds sublime.

 

Never heard these speakers sounding so great.

 

Incredible bass!

 

I opted for a bass boost 20 to 200 Hz !

 

I verified these measurements in REW too.

 

Thanks for these Audiolense guides @mitchco - really saved me a lot of pain and heartache

 

 

 

 

Perfecto!

[mitchco]

Hi @BlimanThank you for purchasing my book. Attached are two procedures, one written by Uli and the example by Bob Katz. I have not had time to try these, but are on my way too long todo list. If you run into issues, Uli is helpful on his forum. I thought I would link Bernt's Audiolense forum as well since there has been a recent change of venue.

Good luck!

Time Alignment of Drivers by Sinewave Convolution.pdf time_alignment_by_sine_wave_convolution_-_example.pdf