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     Integrating Subwoofers with Stereo Mains using Audiolense

     Integrating Subwoofers with Stereo Mains using Audiolense


    image1.pngIn this article, I walk through the steps of using Audiolense to create a digital crossover and time align dual subs with stereo mains. In addition, showing how to smooth the frequency response and reduce group delay at the listening position.


    This results in a smooth frequency response (12 Hz to 22 kHz ±3dB on my system) with all direct sound arriving at my ears at the same time. The phase response and group delay is mostly flat at the listening position.


    In addition, I walk through a time domain experiment designing two correction filters with the same frequency response, but one with time domain correction and one without.  I discuss the audible differences through listening sessions.




    Why subs for music?


    In Audiolense Digital Loudspeaker and Room Correction Software Walkthrough, I was able to smooth the frequency response of my JBL cinema loudspeakers, time align the drivers, and achieve relatively flat phase and group delay at the listening position. I am happy with the results, but something is missing…


    My JBL speakers have solid output to 40 Hz and can extend in room response close to 30 Hz. Great high efficiency kick and punch, but missing a bit of weight on the bottom octave. I listen to rock, blues, and alternative music, most of which does not have deep bass (bass guitar low E is 41 Hz), so if I added subs, would I notice?


    Enter Rythmik Audio. A company that has been around for many years with a good engineering and “no-hype” reputation. Being a tech geek, I was intrigued by their direct servo technology and the most extensive FAQ I have seen from any sub manufacturer. Most importantly for me, one of the few sub manufactures that publish their own measurements and validated by 3rd party testers.


    I purchased two F12 entry level music subs direct from Rythmik. They arrived in a timely fashion and extremely well packed – thick cardboard box, within another thick cardboard box, with the sub floating in high density foam. Could drop off the end of the truck with no damage. Very nice.




    Sub Setup


    Much has been written about setting up subs in numerous configurations. One can even use room simulation software to determine best placement. However, since we are using DSP, try whatever sub setup and configuration you wish. If your ears (and measurements) disagree with the sound, then some fine tuning of placement may be required. However, using this advanced DSP, it is likely the measured results and listening experience will be more than acceptable.


    It is generally accepted for frequencies below 80 Hz, it becomes difficult to determine a sound's location. If you click on the link, you can try it in on your own system. As an ex-recording/mixing engineer, I can say for the recordings and mixes I worked on, low frequencies below 100 Hz, whether from bass guitar, drums, piano or synthesizer, were always in the center of the mix and never panned. This seems to be the case for most music, except from the 60’s and other recordings where it is intended as an effect.


    So, why “stereo” subs? I wanted more sub output, without having to buy a bigger sub. Whether it is one or multiple subs, it is important to be able to individually control the frequency and timing response of each sub with respect to the mains. This is the key takeaway from the article. In my case, I have control over the frequency and timing response of all drivers in the system.









    Is this the best location for my subs (i.e. between the JBL’s and electronics)? Probably not. It is more about convenience than anything else. Once measured and listened to, they sounded good no matter where I am in the room. However, I am mostly interested in how they sound across my three seat listening area. 


    Sub configuration:








    I am letting Audiolense take control of adjusting delay/phase and XO duties for the best possible integration with the rest of the drivers. I level matched the subs to the mains with the volume control.




    Configuring Audiolense


    See my intro article to Audiolense for basic configuration and operation, as most of this article will focus on the differences, so as not to duplicate content.


    I have taken my existing stereo two way biamp (XO at 630 Hz) speaker setup, and turned it into a three way triamp setup, crossing the subs at 40 Hz:








    Why did I choose 40 Hz as the sub XO corner frequency? The JBL cabs with 2 x 15” drivers each, have solid bass output to 40 Hz, as will be seen in the measurements. I like the high efficiency punch and slam, but looking to supplement the bottom octave to give it the full weight.


    The source material I listen to does not have much output below 40 Hz (we will see about that), which allows me to get away with smaller subs, as the JBL cabs take the majority of the bass signal. As it turns out, even with 525 watts @ 4 ohms into dual 15”woofers per side, the music beat can trigger the limiters first before the subs run out of gas (at 300 watts per sub). This is at concert level of ~105 dB SPL continuous output at the listening positon with peaks above that. That’s just for short term fun, as hearing impairment begins at around 5 minutes at this continuous SPL, even though it sounds perfectly clean.


    Most of my critical listening is performed at reference level, i.e. the magic of 83 dB SPL. For lower levels, I calibrate using JRiver’s dynamic loudness control, which provides a more natural sounding volume control based on the frequency response characteristics of human hearing.


    The main Audiolense screen shows the newly designed digital XO:







    In my previous Audiolense article, there is more about XO choices, steep slopes, etc. Bernt has a really good section on XO choices in the help file. Also, Rod Elliott’s article on, “Phase, Time and Distortion in Loudspeakers” has a good read on crossover filters. 



    Taking Measurements


    The detailed steps of setting up and taking measurements are covered in my previous walkthrough. Here, I am simply taking the measurement:







    A few things to note. See how the channel outputs are matched to which speaker. It is likely that you will need to enable Output Channel Override from the Advanced Settings menu. While I have a triamp config, a passive stereo mains with added sub(s) (i.e. biamp config), will require one or more additional DAC output channels, one per sub.


    Note the frequency sweep ranges for each of the channels. Keep this in mind when looking at the frequency response charts, relative to the crossover slopes. When a correction filter is made, the speaker’s raw response is convolved with the corresponding digital cross over slope. In the case of linear phase crossovers, which these are, the advantage is that all direct sound frequencies arrive at the same time to the listener’s ears, in phase.


    Check the time delay in the last column. This measurement has already been taken and these are the resultant delays between drivers, relative to the tweeter. If you look back at my previous article, you will see the same delay values for the midrange channels. While the subs appear to be in the horizontal plane, I did tape measure them, one can see the measured delays for each channel are different, which is why we want independent time domain control for each sub.


    Below is the Audiolense filtered frequency response that better represents what we hear versus the raw, unfiltered measurement. I used a custom filter procedure with True Time Domain (TTD) correction turned on, as well as TTD per driver and selective preringing turned on. See the Audiolense help manual or my previous article for details for designing your own custom filter procedure:









    I left the crossover slopes in so one could see how the each driver’s response would be convolved with their corresponding digital XO filter. Meaning, with the corner frequencies chosen, and steep XO slopes, each driver is working well within their normal operating range. Therefore, the acoustics slopes of the driver become the linear phase digital crossover slopes, and sum perfectly both in the frequency and time domain.


    Let’s focus in on the subs (Rythmik L12) response. Here I zoomed in on the horizontal frequency scale and left the 40 Hz low pass XO displayed so one can see how the subs measured response will be convolved with the XO. I.e. the acoustical slope will become the digital XO slope after 40 Hz:








    In my room, solid response down to 12 Hz (-3 dB). After 40 Hz, starting to see real room effects and then by 110 Hz, smoothing out and rolling off. Resembles the measured spec from Rythmik.


    Now let’s look at the bandpass (JBL 2 x 15” ported cabs):









    Good response to 40 Hz, which is the tuning frequency of the JBL vented cabs. Note in both the subs and bass measured responses, the left side of my system has nulls around 70Hz and 90 Hz and for the right speaker, non-minimum phase response at 110 Hz, 120 Hz and 140 Hz. In part, because my stereo is set up off center of the room so the left speaker is more in the corner and the right speaker positioned at the middle of the long wall. We will see Audiolense do its room correction job in these areas so both the timing and frequency response arriving at ones ears matches, even though the stereo is offset to one side of the room.


    40 Hz looks to be a good XO point, again within the normal operating range of the woofers. Folks may choose a higher XO point if using bookshelf speakers. Same goes for 630 Hz XO point, well within the normal operating range of the compression driver. Let’s turn our attention towards the measured time domain (i.e. step response). 


    In the Audiolense main form, select Impulse Response from the Chart View radio button group. Then from the Audiolense Analysis menu, select Measurement, then select Step Response: 









    I have labelled the timing diagram to help identify which driver goes with which peak. We are looking at the direct sound plus the next 20 milliseconds of sound arrival at the microphone at the listening position. The first arrival at the listening position are the tweeters with positive polarity (i.e. the compression driver and waveguide). Second arrival is the bass and lower mids (i.e. the double 15” cabs) at .69 (left) and .71 (right) milliseconds later, again with positive polarity. Finally the subs arriving 2.75 (left) and 3.38 (right) milliseconds later after the tweeter, with negative polarity. The delay values are from the measurement window shown earlier. Note that Audiolense assigns 100 ms on the horizontal scale as the start of arrival of the sound. In our relative terms 100 ms = 0 ms.


    That’s approximately 3 milliseconds of delay for the subs, even though they are in approximately the same physical horizontal plane as the double 15” cabs. If we were to visualize that, sound travels ~1 foot per millisecond and would be as if the subs were physically placed 3 feet behind the mains from their current location. 






    Certainly relative to the tweeter peak, the subs are delayed and have long wavelengths. Now let’s expand the horizontal scale to see 40 ms (i.e. ~ over 40 feet of sound travel in the room).
















    Whoa! The left subwoofer has a huge reflection which shows up as an amplitude peak at 28 milliseconds (i.e. 128 ms on the chart). Its magnitude is bigger than the direct sound, which means it is maximum phase peak at 28 ms. Let’s look at 100 ms of sound travel to see if there are any other room issues:









    The tweeters spike is just a sliver compared to the subs peaks. No other major issues after 100 ms of sound travel. That’s quite the peak at 28 ms. Something to ponder. Let’s see what Audiolense can do about that.


    The other aspect of long sub wavelengths is where is the peak located? The peak can occupy several samples with the same amplitude values… and even peak higher later in time, as we see in the example above. Because we are using linear phase digital XO’s, the peak is half of the filter length in number of taps. In the case of our 65,536 tap filters, the peak would occur at sample position 37,268 – i.e. the peak of the waveform.  For a minimum phase XO, the peak would occur at sample position zero, which would be the start of the rise of the waveform. Audiolense will automatically calculate and align the peaks of each of the drivers and ensuring all drivers are positive polarity. 


    As a side note, there are several techniques and software tools available to measure the time alignment of drivers in a loudspeaker system. As mentioned above, it gets tricky in sub territory due the long wavelengths involved, especially with an XO of 40 Hz. I have tried most of the tools and techniques available. I must say that Audiolense has exhibited the best accuracy with a high degree of precision that is both predictable and repeatable for time alignment. I demonstrate that here with two sets of measurements taken months apart and can replicate the exact time alignment with the woofers and tweeters (see previous article). The fact that the process is automated is a real time saver. Manually time aligning drivers requires many steps and is prone to carbon unit failure.


    Finally, time alignment is not just for one mic location either. There are not enough pages here, but in my book I show time alignment of a three way triamped system maintains perfect time alignment after moving the measurement microphone to 14 different locations, covering a 6 foot by 2 foot grid area at the listening position. Basically the area of a 3 seat couch, whether sitting upright or back into the couch. Same goes for phase and group delay, virtually flat over the same listening area. 


    Let’s see what Audiolense can do with this “typical” mess.




    Designing a Custom Filter


    Correction Procedure Designer (CPD):


    As mentioned above, this is the same procedure as described in the Audiolense intro article and not going to repeat here. Rather, let me share some tips. It is worth the time to read Bernt’s help file on what each of the CPD controls do, as it makes a big impact on the sound quality. 


    Essentially one is defining how much correction both in the frequency (i.e. dB of correction applied) and time (over how long in milliseconds) domains, using a user defined frequency dependent window and psychoacoustic filtering that best represents what our ears/brain hear. This offers considerably more flexibility that any other type of eq, plus time domain correction is being applied. Not only time aligning drivers, but correcting for room reflections. There is an example of that in the Audiolense intro article where the group delay in the bass frequencies were greatly reduced.  


    As shown in the above step response measurement, the left sub has a huge amplitude peak at 28 milliseconds that is greater in amplitude (i.e. maximum phase) than the direct sound. We will see Audiolense correct for that.


    It may take a few filter procedure iterations, similar to narrowing down the target response process, as described next, where one is happy with the sound quality. I encourage experimentation to try several graduated settings, generate/save filters and while listening to music, switch filters in real time and listen/compare. The workflow is fast, only taking a minute or two to cycle through it.



    Target Design:


    Let’s start with a quick tutorial on preferred target frequency responses. I have covered some of this in the Audiolense intro article and my series on measuring loudspeakers. In this article, I am going with Sean Olive’s and Floyd Toole’s research on The Subjective and Objective Evaluation of Room Correction Products and The Measurement and Calibration of Sound Reproducing Systems respectively. 


    From Sean’s slide deck, is a preferred ranking of average magnitude responses, measured at the primary listening position:








    The top preference (red trace) is a flat, but tilted measured response. If 0 dB is 20 Hz, then it would be a straight line to -10 dB at 20 kHz.

    Note that this tilted measured response is perceived by our ear/brain as subjectively flat or a neutral response according to Sean’s research:








    A measured flat in-room frequency response is not the preferred target. See how an objectively measured response of 20 Hz and straight line to -10 dB at 20 kHz is subjectively perceived as a neutral or flat response to our ears/brain (red trace overlaid in the above chart). Reading the articles linked above and JJ’s research on Acoustic and Psychoacoustic Issues in Room Correction (See PowerPoint presentation) explains further.


    Armed with that knowledge, I designed a similar target response in Audiolense’s Target Designer:








    0 dB = 20 Hz and a straight line to 24 kHz, so it is -10 dB down at 20 kHz. A good place to start and one can fine tune to taste by moving the 24 kHz red marker up or down in one or two dB increments at a time and listen.


    I use a “bracket” method by first making one target sounding too dull and the other too bright. Might take a few tries to establish this. Then between the two targets, I move the 24 kHz maker up or down, in 1 dB increments, generate and save the filter. Using JRiver’s Convolution engine, as an example, I open the file dialog and select a FIR filter while the music is playing. There is less than a second gap of silence as the filter is switched. It is fairly easy to hear the spectral differences between filters using 1 dB increments. 


    One can cycle through the process fairly quickly and in no time, be narrowing it down to a couple of candidates. Once you are down to two candidates, this takes a bit more time, as you cycle through more music, switching back and forth a number of times listening to more and more tunes. But after a couple listening sessions over a few days, one will emerge as your top preference. Whether your preference is for a neutral tone, or whatever your preference is, you can narrow it down quickly using this method.


    As a side note, and not directly related to our subwoofer discussion, a loudspeakers directivity index and measured off axis frequency response is an important consideration when using DSP. The JBL 4722’s I use have a tight, but constant or controlled directivity polar pattern. It responds well to on-axis eq, as off-axis response is virtually identical, due to the constant or controlled directivity design of the waveguide used for this speaker. The Harman/JBL “spinorama” chart for my loudspeaker shows good constant directivity from about 400 Hz on up. With these speakers toed in an equilateral triangle of 10 feet, is just enough distance to illuminate the entire couch area with full range frequency response.


    Note these constant directivity waveguides require high frequency compensation by design. No audible issues were heard with the HF compensation engaged and the measured HF distortion is well within the capability of the JBL Pro 4” large format compression driver.

    One can also see that the target design follows the natural roll off of the subs in the room. This is best practice. 


    It is worth spending the time on target design. For best accuracy, one has to zoom right in on the red marker (i.e. dot) to line it up exactly at what frequency and dB setting you want. I mean zoom way in. It will take multiple zooms by clicking down and dragging the mouse from top left diagonally to bottom right and releasing. And vice versa to zoom back out.


    Final guidance relative to custom filter procedure design and target design. Pick one and optimize first. Only adjust one variable at a time in order to train ones ears to know what to listen for. Personally, I optimize the spectral timbre (i.e. frequency response) first. In other words, the target design. For me it is about getting that neutral sound. Pick a target response, like the Olive and Toole one referenced above, use it and if you don’t like it, then fine tune to your preference. With the recommended bracket procedure, it won’t take long to zero in on what you prefer.


    If you look in my book or search online, there are several recording/mixing production guidelines, well known monitoring procedures and industry specs that one does try to attain as a professional in the industry. I spent 10 years in the recording/mixing chair and the “sound” is ingrained in my mind, as I heard each recording and mixes hundreds of times and on many systems outside the control room before committing to final mix for mastering. When your number one goal is to have the music “sound good” on a wide range of playback devices and environment’s, the pros try and mix and master on neutral speakers in a neutral environment (i.e. a control room acoustically designed to a specification). So whatever is artistically rendered, translates the intent as best as possible across a wide spectrum of sound reproduction systems.


    While there is quite a bit of variability in the sound quality of recordings, mixes and masters, I find that the vast majority of recordings sound good across my system, including the mixes I made in the recording studio. I should know how they are supposed to sound, as I was there and mixed it! All meaning to say that one target response does work well for the vast majority of recordings I have. For example, all of these tunes sound great on my system. Rolling Stones Top 500.


    It is not so much about the variability of sound quality that I object to. It is the excessive dynamic range compression that is crushing the ever living beat out of the music, is what I object to most.




    Filter Generation and Simulated Output


    Let’s look at the simulated frequency response, with the target:







    The -3 dB points are 12 Hz and 22 kHz, and within ±3 dB variance across the frequency range, plus a tighter tolerance than that through most of the range. It did take a number of iterations of CPD tuning and listening to achieve this result. Just like in the intro article, Audiolense’s simulation is virtually identical to the measured response, using a 3rd party acoustic measurement software like REW, for example.


    Not let’s look at the timing (step) response. Here is the simulated step response plus target over 100 milliseconds:








    Virtually text book timing response closely following the target. No preringing, perfectly time aligned, (that’s the vertical step showing no discontinuities) and the elimination of the maximum phase peak. Pretty much as good as it gets, for my speakers in my living room.


    Let’s zoom the horizontal scale to show the first 40 ms of sound arrival:








    A note on reading the chart, consider the 780 millisecond start of the step as reference = 0 milliseconds. Meaning the signal has reached the microphone at the listening position or our ears for that matter. This shows that all frequencies are arriving all at once (i.e. time aligned). Further, that nasty maximum phase peak at 28 milliseconds is gone and overall, the response follows close enough to the target for rock and roll




    A Time Domain Experiment


    An experiment I performed is AB’ing two different FIR correction filters, one with time domain correction and one without, but both having the same frequency response. The exact same target and correction procedures were used, except for TTD correction is not enabled, nor is TTD per driver, but all other settings remain exactly the same. This effectively turns the time domain correction off, but has the same frequency correction (i.e. tonal response), so when switching between FIR correction filters in real time while listening to music, one can start to tune into the difference it makes when a system is time domain corrected, versus one that is not, especially with subs. Why? Subs introduce milliseconds of delay and even at low frequencies, we can still hear the overhang or lag in the bottom end.


    Here we go, a new filter procedure, same as the previous filter procedure, but with time domain correction turned off, same target and all other settings identical. Here is the result:








    As one can see, the frequency response is virtually identical to the same frequency response with the time domain correction. Check. Now let’s switch to the time domain and look at the step response over 100 milliseconds:







    Does not track at all to the same target. Look back at the step response with the time domain correction turned on. Quite a difference. Let’s break it down a bit. I zoomed the horizontal scale to 60 milliseconds so we can see the time domain issues better:









    Just like the measured step response at the beginning of the article of the non-time aligned speakers, here in our modeled simulation, we can see the tweeter arriving first, bass cabs second, and the subs some 3 milliseconds later, with a negative going waveform. Of course, the reflection from the left sub is still there, higher in amplitude than the tweeter, but not quite the same height as the woofers from original measurement. Why? Well, we have applied frequency correction, so there is going to be some effect on the timing, but as can be seen, not much compared to the original measurement. This is a great example of how one can’t fix time domain issues with eq. Not only does the timing response resemble nothing like the overall target, but also can’t fix the high frequencies arriving at my ears before anything else and the subs arriving late.


    Given that the two frequency responses are identical, but the timing responses are not, means that Audiolense can adjust the timing response independently of frequency response. This is exactly what happens when Audiolense True Time Domain (TTD) is turned on. All drivers are time aligned and the timing response tracks closely to the target response while taming room reflection issues like that maximum phase peak in this article and reducing group delay at low frequencies as demonstrated in my previous Audiolense article.


    The question… is any of this audible?


    Personally, under blind conditions with my lovely assistant switching filters, I can distinguish between the two every time, even though it does take some concentration. It is not a night and day difference, but rather subtle. For me there are two audible tells. One is that more often than not, the tweeter or high frequencies are the first to arrive. This, to my ears, produces a more forward sound, being a bright brighter in tone, even though the frequency response is the same. That’s because the tweeter is almost always physically closer to ones ears than the other drivers. No amount of “eq” can fix the tweeter arriving first.  Second, one can hear the lag on the other drivers, especially the subwoofer. It does take a while to tune into what is happening when switching between filters, but when you hear it, it is hard to forget about it.


    In addition, this may be what people perceive/confuse as “slow” bass, meaning bass overhang or simply the sub is still outputting, or not begun to output sound, even though the transient has passed and the mains have stopped outputting sound. It is easier to tell with transient material like drums that have a good high frequency cue and low frequency content, like a kick drum for example. Can you hear the click first and then the boom? Or does it sound integrated?


    Given the sophistication and power of today’s audio DSP software, I am hoping that the industry revisits the time domain of speakers in rooms, as it seems to me to be the missing half of what constitutes accurate sound reproduction. If the goal is to accurately reproduce the waveform of what is stored on the digital media to ones ears, then there can be no frequency or time domain distortions added to the waveform. 


    Just like in my previous article where I show the flat frequency, phase response and group delay of my Lynx Hilo converter, I should be measuring the same as the sound arrives at my ears. This ensures, whatever is stored on the digital media is reproduced as close to as possible to my ears with little frequency or time domain distortion. However, there are a few major transfer functions along the way that really mess with frequency and timing responses – mainly non-time aligned speakers of all types, in wildly variable room acoustics of all shapes and sizes.


    This is where Audiolense comes into play as one can design a custom, high resolution digital FIR filter that contains the mathematical convolution of your specific speakers in your listening environment. This custom filter is designed to restore the music signal back to what is actually on the digital media, or as close as possible, again, specifically designed for your speakers in your room.


    It helps if your speakers are time aligned with constant directivity characteristics. It helps if the speakers have been designed and engineered using science like Harman’s spinorama system for example. 


    Lots of controversy over whether to correct frequencies above Schroeder or 500 Hz as an upper limit. In my case, I am trading a more ragged frequency response for having high efficiency (or dynamic speakers). These speakers respond well to eq as verified by the measurements below. If my speakers were Salon2’s for example, they would have a smoother response beyond 500 Hz and may require no eq at all, but they are at least 10 dB less efficient than the JBL’s. Audiolense’s partial correction can be set for any frequency and independently controlled in both the frequency and time domains.



    Verification Measurements:


    As mentioned in my previous article, Audiolense simulations are virtually identical to real world measurements using a 3rd party acoustic measurement software like REW. While there is some variability based on smoothing algorithms used, my book shows in detail that these sophisticated DSP packages produce simulations that are virtually identical to their corresponding measurements.  I can also overlay exactly what the subs contribute versus just the JBL cabs. 


    Frequency response:







    -3dB points are at 12 Hz and 21 kHz and within a ±3 dB tolerance of the target design and better than that over most of the range. 


    Step response:









    All direct sound arriving at the same time and well behaved over 100 milliseconds of sound travel in the room. That nasty peak at 28 ms is gone.


    Group Delay:








    Mostly flat with natural rising delay at the very bottom of the response. A little ripple at 350 Hz.










    Again mostly flat with natural rising phase in the low end. Here we can see a bit of ripple a 350 Hz. Again unlikely it is audible, but I will investigate.


    This Bruel & Kjaer application note on, Loudspeaker phase measurements transient response and audible quality, provides some insight for folks interested in this topic. The one limitation that is overcome with modern DSP software is the ability extract the minimum phase response, correct that, while independently correcting the excess phase response.


    What is interesting is that not only the frequency and timing responses match well between channels, but so does the phase response and group delay. All of which are responsible for a speaker’s ability to completely “disappear”. All one is left with is the stereo illusion presented in 3D with a rock solid stereo phantom center image.


    The most telling improvement is not only showing that the subs integrate seamlessly, both in the frequency and time domain, but how much the subs contribute to extending the bottom end of my JBL 4722’s:








    The red and green traces are with the subs integrated. The blue and purple traces are without subs as measured in the previous article. I used the “bracket” method to determine my target preference for each setup. Even over a span of several months’ between measurements, and target designs, I arrived at virtually the same tonal response, except for the bottom end extension. This was starting with blank target designs in each case. It is interesting to me that I consistently end up with the same tonal response. I know what I prefer


    Finally, a zoomed in scale from 10 Hz to 150 Hz showing the difference it made to my setup integrating subs:








    Listening Results






    Wow, I shouldn’t have waited so long to add subs to my setup. The added weight in the bottom octave really compliments the double 15” impact, giving it the full club or concert sound I have been looking for. I love live music and anything I can do in my home system to give me the feeling like being at the concert or club is all good to me.


    As a side note on the pic above. For critical listening, I move the coffee table out of the way, I took this pic while I was supposed to be working. Since this pic, I installed quiet curtains behind the speakers covering the windows. They do a really good job in quieting my overly live room to fit the upper RT60 spec limit for the size of my room.


    I must say I am surprised how much music material I listen to actually has output below 40 Hz – virtually everything I have has some content below 40 Hz. I can listen with just the subs turned on. I can see why the Rythmik L12’s are recommended by folks with planar or electrostatic speakers. To borrow an Austin Powers or Jake Peralta word, toit!!


    Here is a small sample of music I use to evaluate audio systems and simply enjoy the music. This subgroup contains tunes that have reasonably good dynamic range. But before I do, please allow me to make a short comment on the state of the recording industry:







    Michael Jackson – Thriller is #1 in worldwide sales, and still is today, along with AC/DC – Back in Black, #2 in worldwide sales. Both are DR 12. I want more of this dynamic sound and less of the hyper compressed music that makes up, unfortunately, the majority of my music collection. 


    While I get artistic intent, I feel the DR scales above fairly represent what is good and bad sound from a dynamic range perspective. See the bottom of the DR chart, where DR = punch and impact. Yah, more of that please. There is no excuse for overly compressed music today. It’s just wimpy loud sound.


    If there is one thing we can collectively ask for as music consumers that would make the biggest impact on our sound reproduction systems is to allow the consumer to control the volume. Now back to our regular scheduled programming.








    I could go on about each one, but I think I have gone on enough. Most were a new listening experience for me, discovering for the first time how much low frequency content was on each recording. If there is good low frequency content on the media, it is reproduced unlike what I have heard before from my system. Great fun.


    Aside from a couple of later recordings, all of these recordings, mixes and masters are 16 years old or older. While my daughter bugs me that this is Dad music, it is sad to me that I have to go back 16 years or so to get a decently recorded, mixed and mastered rock album that has some dynamic range (check out the DR column in the playlist above). The more modern music I listen to on a regular basis, most unfortunately, is in the DR8 to DR6 range with too much dynamic range compression.


    To be sure, my sub application is for music. However, the subs are fun with movies too. For example, the Jumanji remake with the drums and rhinoceros stampede shook my house so much my daughter came running out from her room wondering if there was an earthquake going on. Mission accomplished. However, these are light duty home theater subs, Rythmik has several larger subs designed for LFE HT applications. For music, I find these more than loud enough for my particular scenario.






    While several audio DSP or DRC products can smooth the frequency response, Audiolense’s True Time Domain (TTD) correction is something to experience. Not only accurately and precisely time aligning drivers but also taking care of room reflections. I know of no other DSP on the market that can do this with this level of workflow automation. One can set up, take a measure, and be listening to a first good corrected response in under 30 minutes. Fine tuning after that is to one’s preference.


    To me, accurate sound reproduction means the sound reproducing system (including room) is not altering the frequency or timing response arriving at my ears. Meaning a flat perceptual response within a ±3dB tolerance with no phase distortion or excess group delay. I want to hear the music arriving at my ears matching as closely as possible to the content on the recording.


    Most loudspeakers are not time aligned and the timing response (i.e. delay and phase) gets worse when adding subs due to the long wavelengths involved. In addition, room reflections are inevitable due to the physical dimensions of our listening environments. While there are several subwoofer configurations that can help smooth out the bass, Audiolense DSP can pretty much smooth out the response. I show two subs integrating perfectly with my stereo mains and arguably achieving a smoother frequency response than adding more subs alone would do. Having time aligned subs with mains really shows off the transient impact of having the entire music wavefront arriving at ones ears at the same time. I can’t emphasize this point enough.


    If you read JJ’s article linked earlier on, one can learn why we hear what we hear in small room acoustics. Audiolense takes advantage of this knowledge and programs the ability to control these parameters in a software DSP program. Includes user adjustable algorithms like frequency dependent windowing, which based on JJ’s research shows that the spectral balance (i.e. timbre) our ears care about is a blend of room interaction at low frequencies, and mostly direct sound in the mids and top end. 


    Later arriving reflections have an influence on the perceived frequency response, and sometimes quite substantially. Therefore, a more psycho-acoustically correct frequency smoothing technique is used in combination. As a result, this is what Audiolense sees and corrects in the frequency domain. A frequency smoothing based on psycho-acoustic principles leads to a smoothed response that sits high in the comb filter region and avoids overcorrection of dips. 


    These two psychoacoustic features are just the beginning of this very sophisticated and powerful audio DSP software program. At 64 bits of resolution, the calculations and FIR filter adds no distortion of any kind when convolved with the music signal.


    If you are going the whole nine yards with digital XO everything with True Time Domain correction, Audiolense has automated most of the workflow. Once the XO’s are designed and satisfactory, then the workflow is basically the same as if one is working with a passive loudspeaker. It is quite the time saver.


    I can recommend Rythmik subs and Audiolense to anyone looking to get the most out of their two channel or multichannel system. While I have been into DSP for quite a while, I should not have waited so long on adding subs. Using Audiolense, the subs integrate seamlessly with my mains as evidenced by the simulations and verification measurements – and my ears! Those subs are low frequency canons and really add weight below 40 Hz to give a deeper, but “toit” concussive sound quality. Those are Rythmik’s entry level subs. I am really impressed.


    If you can achieve objective measurements similar to what I have shown in this article, I don’t think you would be disappointed with the sound quality. You may find correcting the bass in room is all the partial correction one needs, if the loudspeaker exhibits really smooth on and off-axis mid and high frequency response. The time domain correction can also be independently set for whatever frequency. It can be set to the same frequency as the partial correction above. This will correct the low end in the time domain, like the two examples of reducing group delay in the previous article and controlling reflections in this article. Or you may choose to apply an overall time domain correction, if you can always hear the tweeter arriving first, but just a partial frequency correction to correct below 500 Hz. Experimentation is encouraged.


    Two other Audiolense features to be reviewed in a future article are user defined, mixed phase target design and multi-seat correction. For the latter, some folks feel is perceptually better than a single point measurement used for correction. Let’s see if that is true or not. Until then, enjoy the music!



    Note: Mitch Barnett's previous article titled "Audiolense Digital Loudspeaker and Room Correction Software Walkthrough" can be found via the link below.









     I wrote this book to provide the audio enthusiast with an easy-to-follow step-by-step guide for designing a custom digital filter that corrects the frequency and timing response of your loudspeakers in your listening environment, so that the music arriving at your ears matches as closely as possible to the content on the recording. Accurate Sound Reproduction using DSP. Click on Look Inside to review the table of contents and read the first few chapters for free.







    Mitch “Mitchco” Barnett.


    image15.jpgI love music and audio. I grew up with music around me, as my mom was a piano player (swing) and my dad was an audiophile (jazz). My hobby is building speakers, amps, preamps, etc., and I still DIY today.

    I mixed live sound for a variety of bands, which led to an opportunity to work full-time in a 24-track recording studio. Over 10 years, I recorded, mixed, and sometimes produced over 30 albums, plus numerous audio for video post productions in several recording studios in Western Canada.


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    On 5/13/2018 at 8:48 PM, mitchco said:

    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


    Thanks a lot Mitch!


    Before summing with factor 1 (1L+1R) should the whole signal be reduced by 6dB to avoid digital clipping? 





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    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

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    The Lynx Hilo looks pretty flexible but I have a question that I haven't been able to locate an answer too.  Can you output both an analog and digital signal at once?  The scenario I am thinking about is having a multi-channel digital signal sent to the Hilo, the subwoofer R&L signals converted to analog and sent to the subs, the mid-high R&L digital signals sent to an outboard DAC of my choice and converted there.  Is this a possible scenario?



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    Yes. That’s what I do. But you need to make sure the propagation delay through your R/L DAC is exactly the same every time. This is a big problem with many DACs. You should check with the manufacturer. Many DACs have goofy jitter attenuation that creates a variable propagation delay. These DACs will never work with this setup. My Benchmark DAC works perfectly. You can check the group delay in REW loopback. You need to take multiple measurements to make sure the R/L DAC isn’t drifting. That’s the only way to make sure. 

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    8 hours ago, dallasjustice said:

    Yes. That’s what I do. But you need to make sure the propagation delay through your R/L DAC is exactly the same every time. This is a big problem with many DACs. You should check with the manufacturer. Many DACs have goofy jitter attenuation that creates a variable propagation delay. These DACs will never work with this setup. My Benchmark DAC works perfectly. You can check the group delay in REW loopback. You need to take multiple measurements to make sure the R/L DAC isn’t drifting. That’s the only way to make sure. 

    thanks for the response!  Not sure if this is the scenario I would use but it is an option!

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    I stopped reading your article when I saw your crossover sweeps and delays/speaker. I'm also wondering how your tri-amping because I just can't see enough hardware in the pic. When truly actively bi or tri amping, one bypasses all of the internal passive crossover networks --going straight from the amp to the speaker--nothing in between. Maybe that's the problem with your 15" bass response. I have a set with dual 12" woofers and when I employ them, they will shake the house and then some--2600 sq ft above ground. I only use the subs when I have my dual 8' woofer speakers set up. I like to swap them in and out. I'm not putting the author down, I just don't see the hardware, speaker wire that one would normally see with actively bi or tri amped speakers. 

     I have never used a digital crossover, so maybe I'm wrong here. But I have been truly tri-amping speakers with electronic crossovers for decades. I vertically bi amp the midrange and tweets and horizontally bi-amp the woofers. Sometimes, i switch that and vertically bi-amp the woofers and mids and then horizontally bi amp the tweets with a smaller amp.


    And then, I add a fourth class A/B amp (380 watts RMS @ 4ohms) per channel to drive my 12" passive subs (JBLs in 1" thick MDF with a 2" thick baffle). You'll never get good, tight bass from a class D inboard sub amp--never. I also elevate the subs so they aren't picking up shit from the floor.


    I don't think you can ever get this truly right without using a real electronic xover which sums for the subs and outputs to each amp individually. But, my listening room is a lot larger than yours.


    So, I use a real xover and SPL meter and it takes about two hours to do using a tone generator. Then again, I only employ fully discrete analog amps. the way one goes about doing it is to set each one individually to hit say 98db from ten feet away. Everything has to be equal to make it sound right--but you should know this given career.

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    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

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    19 hours ago, mitchco said:

    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

    Yes--I have all of those things and my Rane Xovers--including the "howto" for time alignment (which really matters in a large room), setting SPL to match the speakers, etc. It comes with the box. Ranes  can do time align--they can go one step further with moving a jumper or two and time align giant woofers for use with short horns or silk dome tweeters. I prefer discrete circuits--they sound better.


    I've read up on digital xovers. I put them in the same box as five amp integrated amplifiers. They can't do real summing for subwoofers. You ain't gettin what you think you are.  Is that how you're bi-amping the JBLs? With an all in one surround sound kit? One of my friends has a Rotel like that. Not impressed--but he is always impressed with my gear. Real, fully discrete amps which suck twice as much juice as they put out. Back to what I said before, the D class amp in a subwoofer will not put out audophile grade bass. They're boom boxes for theater effects. Try boosting them off the floor and see if the bass gets tighter.


    Since all of my gear is analogue, I only require a DAC to go from the Mac to the pre amp.



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    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!

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    On 6/4/2018 at 11:09 AM, dallasjustice said:

    Yes. That’s what I do. But you need to make sure the propagation delay through your R/L DAC is exactly the same every time. This is a big problem with many DACs. You should check with the manufacturer. Many DACs have goofy jitter attenuation that creates a variable propagation delay. These DACs will never work with this setup. My Benchmark DAC works perfectly. You can check the group delay in REW loopback. You need to take multiple measurements to make sure the R/L DAC isn’t drifting. That’s the only way to make sure. 

    By chance, anyone know if there's a variable PD from an RME Babyface (Gen 1) to a Crane Song Solaris? I know the RME has this Steadyclock feature, and the Solaris I would imagine is quite solid.

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    On 5/10/2018 at 1:22 PM, dallasjustice said:

    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. 




    Where can I learn more about this 4 stereo cascaded sub array and “source/sink” array?

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    hi @mitchco - thanks again for all your contributions. i'm trying to monkey-see monkey-do but not faring too well :)


    i'm quite uncertain about my phase results - i made a few a short video to show where i'm at below.


    donny is out of his element here but that looks quite awful to me and quite unlike the results you got, is that a huge amount of pre ringing or am i misunderstanding? 


    i'm using the default TTD filter procedure as shown below but if i were to play with a custom procedure, i'm unclear how to adjust the windows based on the measured phase response, i'm wondering how you made the decision in your first article:


    "First, let’s try a shorter time correction window in the bass, so now I have 2 cycle window both in the low and high frequencies as entered in the True Time Domain Subwindow:"


    is there something in the measured data that can guide one on how to adjust the time windows? especially since i have huge peaks from the L sub in the phase response as far out as 45ms and 67ms! i do see in the simulated response that those peaks have disappeared... so that appears to be a good sign at least?



    just to add a little bit of background:


    • my room is similar to yours, i'm against the long wall left of the center, on the right rear it opens up to the kitchen
    • 2 x Kef (semi) full range speakers driven by a power amp (channels 1 & 2)
    • 2 + 1 x JL Audio subwoofers
      • right channel is two cascaded subs, both should be receiving the same signal (channel 3),
      • left channel is a single sub (left of the center of the room) (channel 4)
        • anywhere here is just terrible, both in freq response and phase response - i really want to find a reasonable location for the left sub somewhere here but it just haven't been able to yet
    • RME ADI2-Pro in multichannel USB mode, channels 1/2 goes to power amp L/R and channels 3/4 goes to 3 subs
    • i've manually adjusted the phase knob on the left sub which is closer to the seating area so that the first peaks of the sub impulses lines up (they go out of phase later...) - i'm sure AudioLense can correct for that but figured might as well since these subs (and speakers) are used in a home theater system too (Audyssey isn't the best:) 


    here's my speaker setup, i've gone quite a different way than you have (since the mains have integrated crossovers - unfortunately). coincidentally, going this way i have noticed i cannot enter a low pass value on the third tab for the subs, i believe AudioLense just uses the high pass value for the mains as low pass for the subs (judging by the XO graphs), i would have liked to play with them overlapping a bit more. 


    thank you!!




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

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    oh also, wondered what does the measured impulse response say about the polarity of the mains vs subs and if that's a problem?

    speaker setup1.JPG

    speaker setup2.JPG

    speaker setup3.JPG

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    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.



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    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...

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    Thanks Mitch!


    BTW, my search on their forum did not turn up any hits on magnep or maggie

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    Thanks so much @mitchco - played around a fair bit and found the default TTD filtering works best for me, as far as I can tell at least, I'll share another video soon :) 

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    If you haven't installed bass traps in the corners of the room, you should. Makes a huge difference in the quality of the both the bass, mids and highs. Corners include the wall to ceiling points as well.

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    Thanks for another superb article. As a beginner thinking of going down this route in a 'conservative' manner I'm thinking of 2 or 3 steps:

    1. Buy a calibrated microphone and measure the main speakers and sub separately. Adjust (single) sub crossover frequency and volume to integrate with main speakers guided by frequency response curves and remeasure combined sub and mains. Try out various test tracks with and without the sub - schematic would be as in 'Full Range' picture.


    2. Buy some software (HQPlayer/Accourate/Audiolens/Audirvana/JRiver) or use freeware foobar2000/Sox/REW and create convolution filters treating the sub and mains as one full range system. So calculate convolution based on the system set up in step (1). Evaluate system with and without DRC. Probably try LF range alone first since mains and sub overlap roughly 50-100Hz.


    3. Progress to splitting sub as separate output as in 'Split Range' picture. If I'm spending money on software I'd like to include upsampling in the feature requirements. Since convolution needs to be handled in the computer, so must the master volume control so the sub and mains maintain integration. So I need a music player such as JRiver which handles convolution and sub output. This could be connected by line out or by a computer sound card/onboard DAC. My DAC is 2 channel and I wouldn't want to spend more on a multi channel. Since volume control is software controlled the preamp is unnecessary, and the DAC could be set at a safe max volume (for speaker protection).


    So the questions - my concerns are volume control and which software package to buy. Is my reasoning correct and is this a workable plan? Does Audirvana have the required features to replace JRiver as used by @mitchco? I believe HQPlayer could do the job but not sure about mains + sub output via separate devices as seen by the computer.



    DRC Scheme - Full Range.jpg

    DRC Scheme - Split Range.jpg

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    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...

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    I've now bought a miniDSP UMIK-1 and using REW measured my Tannoy Kensington SE's and Wharfedale SW150 HT subwoofer. It's apparent I have a room node at 40Hz which main speakers and sub are exciting. The sub is mainly for HT and I was hoping would supplement the Tannoy's but they don't go low enough and just reinforce the 40Hz node.


    So my next consideration is a better sub. I've noted @mitchco is using Rythmik Audio F12 (about $1100 inc. shipping) but couldn't find  anything for @dallasjustice or @3ll3d00d who also seem to have invested in hifi audio sub set ups. Another option is maybe 2 x BK's XXLS400-FF Subwoofer 400W RMS 12" but this is non-servo controlled.


    Any comments/recommendations from folks that have tried/demoed subs in this budget range i.e. 1 or 2 subs totalling <£1000?

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    Hey Mitcho, 


    I have for the last 3 years continuously been reading your articles over and over again. Its always been a pleasure.


    I actually have similar set up to yours, of course I've been emulating your set up for several reasons. We have talked about this on the ASR forum. I have been convinced on how a set up like yours might hit the spot with audio nirvana. I have to say that the difference between my 3 way Troel 3WC with subs vs my 15inch/compression driver doesn't provide a significant difference. Your experiment(article) comparing the bookshelves with your JBL's was fascinating. This isn't a sign of disappointment the sound is larger with the 15 inch and I am still yet to upgrade to a decent 2 inch compression driver with a better horn/waveguide. So the potential is there, and of course the sound is better but not a huge difference which was noted in your other article.


    Now, I have noticed a few things and I was wondering if you could back it up or refute it with some theory so I know which way to go according to science. I am aware they are amatuer questions but knowledge on them might direct me to a smoother transition with my set up. There just some observations which of course are all related to my particular system and can yield no definite answer but curious ultimately what you think?


    1. I know this question is always asked but I am curious if you feel the same way? Adding two subs 12 inchers(.43 QTC) so there good subs, doesn't add too much and at times I think it better to have all the bass come out from the 15 inches(Deltalite). But it can get messy down low with some types of music. I am curious if I do add subs whats a decent crossover point? I found 70hz was better than 140hz proving my point that more out of the 15 is better. My cabinet with the deltalite is a SBB4 alignment with an f3 at 51hz and tuned at around 40hz. Any recommendations?



    2. I also found that crossing over the Deltalite with the DE250 at 1600hz sounded better. I know that the 15 inch beams at that Hz and a crossover at around 800hz is better and that I do need a 2 inch compression driver to meet it more smoothly. I am just curious if it could be any reason other than the fact that a 1 inch compression driver can't play well at a lower crossover point at around 1000hz? 


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