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Article: Audiolense Digital Loudspeaker and Room Correction Software Walkthrough

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19 minutes ago, nefilim said:

 

Thanks Dallas, curious, which channels are you connecting to the Benchmark ? :)

I use the JBL M2 speaker. The Benchmark DAC3 channels go to the horns. The M2 horns are crossed over at 800hz and play beyond 20khz. 

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33 minutes ago, nefilim said:

Not finding a lot of documentation on Audiolense' site, I'm wondering if anyone know what this means: 

 

"Clock drift correction in measurement is now practically perfect."

 

Does this imply I can measure on a different device than the playback chain? eg one USB device for ADC and another USB device for a DAC?

 

Yes, hard to find, but from the help file attached at the bottom of the article: "If you know that you’re using separate audio devices for recording and playback, or if you’re unsure, check the “separate play and recording device” option under advanced settings. That will prevent any clock drift related problems. When the clock drift correction is engaged, one of the speakers / tweeters will be measured twice."

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Hi @kravi4ka sorry for the delay, the notification did not come through. Re: RME Adi Pro while I don't have any personal experience with the product, it's specs look excellent and has the requisite 2 ch analog in and 4 channel analog out if you are considering biamping. Maybe you can outline your scenario? There is a thread here about it on CA: 

 

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Dear @mitchco , I have always been a huge fan of the Hilo but then some people whom I really trust said that this one is better, not much and if I had a Hilo it would not be worth swapping these two but I am buying a new one so I will probably get it.

 

My idea is to add subs to an existing pair of monitors. Will try to describe my system asap. BTW I sent you a PM.

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Hi @mitchco, awesome article, thank you so much!

 

Until I get a second Sub, what would be the additional steps (e.g. summing, anything else?) and compromises of using one sub? 

You mentioned most of the low bass content is centered so I guess summing the channels arbitrarily would work for most content? 

What would be the correct summing formula? 

SUB=0.5R+0.5L

or 

SUB=(0.707946 x L) + (0.707946 x R)

 

Something else? 

If the SPL is sufficient with a single Sub is another one worth double the cost and the extra troubles? 

 

 

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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

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Hi @mitchco just want to say thanks 

I'm starting on this exciting DRC journey because of your articles.

I think I understand what needs to be done on Audiolense side of things.

I'm a bit confused on what needs to be done in JRiver side, besides adding the convolver.

Do I need to do routing in JRiver too?

Sorry for the basic question, I'm new to Audiolense and JRiver.

I did some research and I believe I need a 2 channel in a 5.1 container? 

and routing of 6 channels in PEQ?

Just getting more confused as I read more... I think I have step 2 and 3 but missing step 1 :) it's time to ask for help.

My plan is to use JRiver/AudiolenseXO and Lynx2B for a 3 way active setup.

Thanks again for your articles and help.

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Hi @Jesal, 3 way active, nice! Yes, I was confused on this point when I first started. There is a specification for the config file so JRiver and other convolvers can read a common file format.

 

The good news is that Audiolense automatically builds the config file, that references the filters, so that all you need to do is load the config into the Convolver. In JRiver, you need 6 channels, so you set the output format in DSP studio to 5.1. I.e. Tools, Options, DSP & Output format, Convolution, Channels,5.1 That’s it. No upmixing donwmixing.

 

A couple of tips. Look at my Audiolense triamp setup in the sub article. See the Speakers->Crossover configuration, note the Channel and Channel ID’s in your setup. Then map it accordingly in the Measurement Window. Also in the measurement window is check speaker connections. Click on that and a steady midrange chirp is sent to each channel/speaker combo one at a time to ensure proper channel mapping before you measure. It’s not very loud.

 

Cheers, Mitch

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Thanks a lot Mitch, spent a few hours on this following your instructions.  After a 2-3 hair pulling sessions i managed to get all channels routed and measured. Feeling a sense of accomplishment that I've gotten this far.

Just a couple of issues to sort out.

1. I only managed to measure once, I ran the Measurement the second time and I keep getting a device not available error.

2. My umik1 has a very loud feedback when plugged in. No issues with my blue yeti Mic. 

 

Step 1 accomplished, thanks to you :)

 

 

 

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Mitch, have you done any experimenting with near-field setups and Audiolense?  I am thinking the process would be no different than with your setup but I was just wondering.  Thanks!

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Hi James, yes, I have experimented with near-field setups, even a set of computer desk top speakers. Sounds great! The process is indeed no different than what is described in the article.  Kind regards, Mitch

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@mitchco !

 

I have stumbled across your masterful DSP write-ups at a time when I thought maybe audio wasn’t even for me anymore, having invested so much money in gear and being so woefully dissatisfied with the results, even with other popular room correction DSP (which does not do custom channels, XOs, etc).

Background: A few months ago I decided to upgrade to the big leagues (at least entry-level big leagues haha). Here’s my initial setup:

RME Babyface XLR> Bryston 4B pro (old model from 1986)>modded budget bookshelf speakers.
Plus two subs:  one Martin Logan Dynamo and one Martin Logan Abyss (connected via phones output [TRS adapter>RCA). Acoustic treatment: six 2’x4’x4” ATS acoustics panels – 2 front, 1 above, 1 each side, 1 backwall

The upgrade: Sonarworks Reference 4>RME Babyface TOSLINK S/PDIF>Crane Song Solaris > Bryston 4B3 > ATC SCM12 Pros plus the aforementioned Martin Logan subs (connected via highlevel input from Bryston 4B3) and a shiny new JL Audio F112v2 (via Solaris Main-out-2 XLR). Treatment: previous six 4” ATS panels, four GIK Tri-traps, four GIK 4” bass traps, two GIK” 6” bass traps, twelve GIK Gridfusors, twelve GIK 4’x1’x2” spot panels, two old model GIK QRD diffusors, and two Auralex LENRDs.

View detailed setup of room here: https://www.gearslutz.com/board/studio-building-acoustics/1201900-room-studio-move-mixing-spot-south-wall.html

So back to the present. I knew my room would be a challenge to work with bass-wise (see dimensions in link above). But it’s turning out to be much more stubborn than expected – substantial bass woof (beer bottle effect) and nulls, even after correction. I thought maybe I need more traps on the front and side ceiling-wall corners. But I finally recognize after reading a bunch of your posts (and eagerly buying your book) that there is absolutely no way around my dissatisfaction without addressing the obvious XO issues, timing issues, phase, etc. So maybe there is hope and I won't have to sell all my gear and take up pottery?

My plan now:

Use Acourate or Audiolense and Jriver with the following routing (which I’m hoping will work):

For ATCSCM12 pro monitors (passive, no bi-amping yet): DSP>Jriver> RME Babyface TOSLINK S/PDIF>Crane Song Solaris > Bryston 4B3 > ATC SCM12

Question: Can the filters target the S/PDIF digital stereo channels on their way to a secondary 2-channel DAC? Or only the RME’s Analog Stereo channels?

JLA F112v2: RME Babyface analog 1/2 XLRs > JLAF112v2

Question: Can I pre-sum the stereo bass signal within the DSP so I can output on only one channel using only one XLR cable (saving on extra tangle)?

Martin Logan subs: if pre-sum is possible, then Jriver>RME babyface> phones analog channel 3 to ML Abyss and phones analog channel 4 to ML Dynamo (via TRS>RCA adapter)

Does this plan seem at all viable?

Thanks for all your work @mitchco!

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@H-space Welcome to CA! Thank you for your kind words!

 

I had a look at your Gearslutz link - you have been busy! I think you will find Acourate or Audiolense to be a big help here. Given the size and dimensions of your room, I would give Audiolense a try as it does cancel low frequency standing waves and room modes. Also, I think the time alignment would be easier with your setup.

 

To answer your questions, I believe on the surface, yes and yes are the answers. But there are some caveats. @dallasjustice has experience using a 2nd DAC... You can find some info a few comments back in this thread. Yes, you should be able to presum the bass, but you may be shortchanging yourself, as one sub may need different correction than the other sub, as mine did in this article, if you look at the measurements. One sub had a wicked maximum phase reflection and the other did not... Also, I am a believer in time aligning subs to mains...

 

Another option is Mult-Sub Optimizer (MSO). The link to my integrating subwoofer article in the paragraph above has user comment information where a CA member is using it successfully with Acourate.

 

Note, if you plan on tracking along with mixing, then the linear phase FIR filters used introduce a 3/4 of a second delay. However, there is a minimum phase mode that can be used and only sacrifice a little bit of the time alignment magic. Audiolense is able to produce a minimum phase filter that is almost indistinguishable from it's linear phase counterpart.  Bert has done a really good job here. If interested, you will get more feedback from other members if you join the support forums. Audiolense is https://groups.google.com/forum/#!forum/audiolense

 

I think you will find that DSP will make a major improvement in the bass response in your room. Short of stuffing the room with bass traps, and I do mean stuffing, this is the only other way to get an even bass response to your ears.

 

Kind regards, Mitch

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On 8/20/2018 at 7:28 PM, mitchco said:

@H-space Welcome to CA! Thank you for your kind words!

 

I had a look at your Gearslutz link - you have been busy! I think you will find Acourate or Audiolense to be a big help here. Given the size and dimensions of your room, I would give Audiolense a try as it does cancel low frequency standing waves and room modes. Also, I think the time alignment would be easier with your setup.

 

To answer your questions, I believe on the surface, yes and yes are the answers. But there are some caveats. @dallasjustice has experience using a 2nd DAC... You can find some info a few comments back in this thread. Yes, you should be able to presum the bass, but you may be shortchanging yourself, as one sub may need different correction than the other sub, as mine did in this article, if you look at the measurements. One sub had a wicked maximum phase reflection and the other did not... Also, I am a believer in time aligning subs to mains...

 

Another option is Mult-Sub Optimizer (MSO). The link to my integrating subwoofer article in the paragraph above has user comment information where a CA member is using it successfully with Acourate.

 

Note, if you plan on tracking along with mixing, then the linear phase FIR filters used introduce a 3/4 of a second delay. However, there is a minimum phase mode that can be used and only sacrifice a little bit of the time alignment magic. Audiolense is able to produce a minimum phase filter that is almost indistinguishable from it's linear phase counterpart.  Bert has done a really good job here. If interested, you will get more feedback from other members if you join the support forums. Audiolense is https://groups.google.com/forum/#!forum/audiolense

 

I think you will find that DSP will make a major improvement in the bass response in your room. Short of stuffing the room with bass traps, and I do mean stuffing, this is the only other way to get an even bass response to your ears.

 

Kind regards, Mitch


Thanks for the reply!

Regarding the subs, to clarify, I very much intend to fully correct each of my three subs independently.  For my two Martin Logan subs, rather than connect right/left RCAs to the Abyss and then another right/left RCAs to the Dynamo (allowing them to mono-sum internally), I thought I'd just send a pre-summed R/L signal to each, so each sub only requires one RCA connection, thus enabling me to use my RME's Phones 3/4 output as two separate correctable channels. Does this still seem shortchanged? I'm not good at explaining this technical stuff coherently, haha.

I took a closer look at both Acourate and Audiolense, and I think I'll be choosing the latter since it might be the faster albeit less flexible option. Thus no “source/sink" or MSO unfortunately. Perhaps these options will be added later?

My subs are positioned as follows:

  • One JL Audio F112v2 against the front wall behind monitors, 33" (1/4 room width) away from the left wall.
     
  • Two Martin Logan subs vertically stacked on top of each other (10" Dynamo on top of 12" Abyss), against the front wall behind monitors, 33" (1/4 room width) away from the right wall.


I'd like to replace the two ML subs with another F112v2 in the future (like Bob Katz!) - the DARO feature (internal frequency correction) is pretty cool. 

Glad to hear any further feedback you might have. Thanks and I'm looking forward to that new book edition!
 

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Man, it wasn't easy but after a week long crash course, I finally have the routing sorted out and a good preliminary filter(s) going. Wow. This is the future. This is what I've been wanting for a long time, which - until I found @bobkatz's and @mitchco's posts - I did not know existed; advanced OS integrated DSP systems.

I am really glad I held back on that JL Audio CR-1 subwoofer crossover (no time domain processing) or outboard DSP like Trinnov (undesirable AD/DA stage which  completely negates the benefit of owning a DAC like the Crane Song Solaris). It's just a matter of more in-depth learning and fine tuning now. 


This really is the future I believe. I wish the mainstream could appreciate the value of accuracy. Thanks.

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On 7/24/2018 at 9:14 PM, mitchco said:

Hi James, yes, I have experimented with near-field setups, even a set of computer desk top speakers. Sounds great! The process is indeed no different than what is described in the article.  Kind regards, Mitch

 

Hi Mitch:

 

First, thank you for all the work you've done to shed light on DSP and room/speaker correction. I feel you have made a significant  contribution in bringing world class sound to the masses.

 

Along that line of thought, there are literally tens of thousands of music enthusiasts who sit at desks, in front of monitors, up against walls listening to music on headphones and 'computer' speakers.  For someone in this sort of 'extreme nearfield' situation, is there a suite of hardware that you recommend using that can take somebody much of the way to the midfield quality you achieve in your home?  Specifically, what speakers would you recommend if they had to be placed behind/beside a typical computer monitor up against a wall? Computer interface? Amplifier(s)? DAC?

 

I am particularly wondering what you would recommend for those of us wanting to pursue time aligned and phase coherent setups? I come from a long background of listening to John Dunlavy speakers, that I really enjoy. When I have used fairly high quality and well measured Adam Audio studio monitors at my desk, they have left me wanting. I can never honestly put my finger (or ear) on the specific problem, but I suspect that my ears clearly prefer time aligned sound.

 

I'd like to start working with Audiolense in a desktop setup, but I am not sure what speakers to start with. Right now, all I have is a Centrance DACMini and headphones. Which speakers would you start with that would allow you to use the time alignment features in Audiolense?

 

Thank you.

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Hi @TooSteep Thanks for your kind words.

 

That’s a great question! Before getting into the specifics of a particular loudspeaker, there are three approaches:

 

Audiolense can apply an overall time correction (using TTD) to an existing loudspeaker that is not time aligned. Assuming a 2-way design, the caveat is that the z-offset between the drivers needs to fall within the time correction window. It is likely for most desktop speakers that this would work. One would have to measure the step response and evaluate the sonic result to see if it is time aligned/phase coherent enough… You could download a trial version of Audiolense XO and try it on your existing speakers. I think the trial version corrects for 90 seconds.

 

A second approach is to biamp a two-way so that one could use Audiolense linear phase digital XO and the software would automatically time align the drivers and apply an overall time correction. This is the approach I chose for the article and my preference.

 

Finally, acquire a set of time aligned speakers for the computer desk. I have not looked into this much Vandersteen makes a time and phase coherent smaller speaker the VLR Wood. I have not seen any measures or listened to the product. Green Mountain Audio is another… Plus there are dozens of near field studio monitors, some of which are time aligned/phase coherent...

 

Again, great question/suggestion. I will start looking into this more and see what I can come up with…

 

Kind regards, Mitch

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 Thanks Mitch.

 

"Plus there are dozens of near field studio monitors, some of which are time aligned/phase coherent" 

I had no idea about this! It isn't obvious from searching or reading literature. Reading the rave reviews of the Dynaudio LYD48, there were some implications about this, but I couldn't quite tell for sure. I will try to find some.

 

I actually own a set of Green Mountain Audio Europa's, which I really enjoy. But the sound is not quite right up close, 2-3 feet away. I feel I need to be at least 7-8 feet away from them before I can benefit from their time aligned magic.

 

When you biamp a speaker, how does it not still run through the internal crossover? Or does it just effectively HPF the tweeter and LPF the mid?

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1 hour ago, mitchco said:

 

Finally, acquire a set of time aligned speakers for the computer desk... there are dozens of near field studio monitors, some of which are time aligned/phase coherent...

 

 

Oh shoot, hoping my ATC SCM12 PROs are sufficiently aligned? I'd hope so, they're ATC ffs lol. I got an XO license last week. Working through a tweak session now. Aiming mic directly in between woofer and tweeter. Hopefully I'm getting a decent measurement.

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On 9/16/2018 at 10:02 AM, TooSteep said:

 Thanks Mitch.

 

"Plus there are dozens of near field studio monitors, some of which are time aligned/phase coherent" 

I had no idea about this! It isn't obvious from searching or reading literature. Reading the rave reviews of the Dynaudio LYD48, there were some implications about this, but I couldn't quite tell for sure. I will try to find some.

 

I actually own a set of Green Mountain Audio Europa's, which I really enjoy. But the sound is not quite right up close, 2-3 feet away. I feel I need to be at least 7-8 feet away from them before I can benefit from their time aligned magic.

 

When you biamp a speaker, how does it not still run through the internal crossover? Or does it just effectively HPF the tweeter and LPF the mid?

 

Several near field monitors are active and typically already have digital XO's and if lucky, time aligned/linear phase operation. Genelec, HEDD, Adam,  Neumann, PMC, Fostex, Amphion, are just a few brands that come to mind.

 

JBL larger speakers like the M2 and Cinema speakers like I have require external XO and amps. So there is no passive XO network. For other speakers that say they are biamp-able, it usually is indeed still passing through the legs of a passive XO... One has to look really carefully as a "real" biamp scenario is where the amps connect directly to the drivers.

 

I am a diyAudio guy and have built/modified speakers and have either unhooked the passive XO and ripped it right out, depending on whether reselling or not.  That's another possibility...

 

As soon as I get some time, I will start looking into this deeper...

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Hi Mitch,

 

I have been looking at Audiolense and Acourate for a while now and have read the manuals, your e-book and tried evaluation copies but have a question, well, quite a few probably but one will do for now. I can follow the procedures you give for measuring and applying amplitude correction to an existing system. I can follow the process of generating crossovers and applying individual driver amplitude and delay corrections. Where I get lost is, for a fully active system, how do I get from crossover design and driver corrections to measuring the room response, applying those corrections and creating a set of filters to use in the JRiver convolver? Could you briefly outline a workflow to achieve this?

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