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The Bass System Setup, EQ/Correction Thread


maxmercy

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Do you use REW with HDMI out to DUT?  If so, what do you use for your loopback reference for phase/timing measurements?

 

I no longer use REW to make my measurements as I now use custom software that fits my workflow better.  Yes, I do measure through HDMI to the DUT.  No, I don't use a loopback reference.  Instead I use a trick.

 

The trick is that we don't really care about absolute timing.  What we care about is relative timing between each speaker and sub.  As such, if you run a sweep through two outputs at once, you can find the relative offset between the two by comparing the location of the IR peaks.  This is one of my reasons for not using REW.  On Linux at least, all it will do is sweep the left and right channels at the same time in stereo mode.  My measurement program can sweep through any combination of the 5.1 outputs.  To get responses for each speaker and subs, I set all my mains to "large" in the AVR and the LFE LPF to 250 Hz (maximum).  If I want to see the bass managed response, I switch the mains back to "small" and sweep again.

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JSS,

 

The accelerometer was placed directly on the couch.

 

If I placed the phone on the concrete floor, there would be little doubt that it would not produce any significant reading whatsoever over the noise floor of the app.

 

You are 100% correct in that the couch's resonant frequency was the cause for the vibration. However, that vibration increased closer to the source, given the same SPL.

 

SME,

 

Thanks for your detailed response.

 

You mention that "the level of vibration in the floor and other room surfaces decreases with distance from the source". The question is: what's properties of sound is causing the vibration? It is not SPL alone as my test demonstrates; the SPL of both sub sets were calibrated to ~96db. 

 

My room is on a concrete slab. The couch sits on the concrete. At 96db @15hz, the concrete is not vibrating in any appreciable amount at all and is not transferring any vibration to the couch.

 

If the couch is not being influenced by the concrete vibrating, and the SPL is the same for both distances, then why is the couch vibrating more in the scenario where the sub is closer to the couch?

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I no longer use REW to make my measurements as I now use custom software that fits my workflow better.  Yes, I do measure through HDMI to the DUT.  No, I don't use a loopback reference.  Instead I use a trick.

 

The trick is that we don't really care about absolute timing.  What we care about is relative timing between each speaker and sub.  As such, if you run a sweep through two outputs at once, you can find the relative offset between the two by comparing the location of the IR peaks.  This is one of my reasons for not using REW.  On Linux at least, all it will do is sweep the left and right channels at the same time in stereo mode.  My measurement program can sweep through any combination of the 5.1 outputs.  To get responses for each speaker and subs, I set all my mains to "large" in the AVR and the LFE LPF to 250 Hz (maximum).  If I want to see the bass managed response, I switch the mains back to "small" and sweep again.

 

What measurement program do you use?

 

JSS

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Once a good placement has been figured out (while experimenting with acoustic treatment for upper bass problems that cannot be solved by placement), integration with mains/surrounds takes place.  Measure, measure, measure, adjust positions, delays, move or place more/less room treatment.  EQ is the LAST thing you do.  LAST.  I cannot stress this enough.  If you have not exhausted ways to solve your FR problems with placement and treatment, EQ is not the panacea it is advertised as.  While mixed-phase correction can correct some problems that room treatment can also address, it can usually only do so at one listening position.  Correcting a specular reflection with EQ is not the way to go about it, treat the room instead.  I see people recommending 'fixing' crossover problems in speakers with FIR EQ filters.  Nonsense.  FIX THE CROSSOVER IN THE SPEAKER, DON'T ADD MORE COMPLEXITY AND ADD MORE TO THE SIGNAL CHAIN AND NOISE FLOOR TO FIX IT.  Fix the root problem.

 

But I understand why people would opt not to fix the root problem, when adding EQ will provide a band-aid.  It's easier than taking a speaker apart or going active and needing more amplification.

 

I agree with you on most of these points, especially with regard to measurement.  There's absolutely no way to fix the high frequency part of a specular reflection with EQ.

 

As for crossovers, they are fundamentally broken but are a necessary evil.  The only way to truely fix a crossover is to not use one.  Good luck with that.  All you can do with a crossover is minimize harm.  Using an active crossover provides more flexibility, but the fundamental issues are still there.  For me to rip out the analog electronics in my speakers and run them with an active crossover, I would have to give up a lot of cash and a lot of space in my media cabinet.  That's a lot more than just difficulty or inconvenience.  While I remain open to considering such an upgrade in my future, I'm actually leaning less towards this approach now that I have better room acoustics, which are actually helping to reduce the crossover problems.  And to the extent EQ can be used to improve the crossover response without having to purchase extra components for an active crossover, using it is not a mistake.  It's good engineering.

 

Likewise, for the crossovers between my mid-bass and deep-bass subs and between my mid-bass subs and mains, I don't have the tools to do the crossovers I want natively.  I'd need an additional 5 OpenDRC-AN units plus a separate active mixer to make this possible.  So actually, linearizing the phase response of the crossover after bass management leads to a design that's cheaper, simpler, and functionally almost equivalent.  I'll update here after I implement my first past attempt at this and have had enough time to develop impressions.  I'm actually pretty skeptical that this phase correction wil make much of a difference either way.  In the long run, I may persue a PC-based solution or custom electronics, but those are a long way off.

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Thanks for your detailed response.

 

You mention that "the level of vibration in the floor and other room surfaces decreases with distance from the source". The question is: what's properties of sound is causing the vibration? It is not SPL alone as my test demonstrates; the SPL of both sub sets were calibrated to ~96db. 

 

My room is on a concrete slab. The couch sits on the concrete. At 96db @15hz, the concrete is not vibrating in any appreciable amount at all and is not transferring any vibration to the couch.

 

If the couch is not being influenced by the concrete vibrating, and the SPL is the same for both distances, then why is the couch vibrating more in the scenario where the sub is closer to the couch?

 

Hmm.  Good question!  How close to the couch is the sub?  If the sub is placed adjacent to the couch, as mine are to mine, then part of the couch is probably seeing a much higher SPL than your listeners are.  So the vibrations are being transmitted from your sub to the air to the couch adjcent to the sub and then to your torso and butt.  I still don't think SIL/PVL has anything to do with it, apart from the extent that they correlate with SPL.

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This is just pulling down a peak at the seats using digital PEQ:

 

cd3989bdba6a360acd536db1a3aaed66.png

 

No smoothing on the traces, except for the close mic with no PEQ, which is 1/6 octave smoothed.

 

I'm not sure what I'm supposed to be seeing here.  The response at the seats looks better.  I have no idea what's going on with the close mic because one of the curves has smoothing and the other does not.  I don't doubt that it's less flat, but I don't see that as being a problem except for people listening with their heads right next to the woofer.  In fact if the peak is truely modal, then the correction will likely improve response across the room on average because the peak part of the mode covers more room area than the dip part.  And you get back some headroom, not that it's likely to make much of a diference with a Linkwitz Transform in the chain.

 

Alas, no one noticed the difference as being an improvement and the first octave of response fell by over 6dB.

 

Fair enough.  You could always try to boost back the lost low-end, but this may not be desireable depending on whether the filters have enough precision for such low frequencies and your needs with respect to gain structure and noise floor are met.  This work always involves compromises at some point.

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Care to share it, or do you plan to commercialize?  I remember seeing this a while back:

 

http://andyc.diy-audio-engineering.org/mso/html/index.html

 

Anyone know if this is available to D/L anywhere?

 

Also, what is your technical background?

 

I have not yet decided whether to try to commercialize the software.  I'm a bit torn.  If I didn't need money to pay bills and buy equipment, I'd very gladly make it free.  On the other hand, I'm not particularly satisfied with my current job and have not yet worked out where I want to go from here.  The possibility of going into business for myself is very attractive on the face of it.  For now, the program is a long way away from being really suitable for any sort of mass distribution.  The code is rather specialized for my own setup, and it only runs on Linux.  With more effort, these limitations could be lifted, but at this time, I'm much more interested in improving my own room/system and learning as much as I can from the process.  If someone offered to pay me enough money to do this work that I could quit my day job, I don't think I'd hesitate to do that work and release it for free if I could.  :)

 

Another thing is that I'm not putting much effort into automating filter calculations at this point.  I have studied various optimization strategies in the past, some of which could be applied to audio, but I opted to develop my sub EQs manually to start.  It's not just that automated optimization is difficult, but in the case of audio, it's actually hard to properly define the objective function.  How much weight should be given to the different aspects of the response?  What is the best way to take headroom into consideration?  This latter question is harder than it looks, particularly for "idiot-proof" fully-automated solutions because you don't necessarily know how much headroom you have to begin with.  As an example, people often advise to "only EQ down peaks and leave the dips alone", but when you do that, you end up increasing the overall sub level to compensate for the reduced average level, which is just the same thing as just boosting everything except the peaks you EQed down including response outside the normal bandwidth of the device.  Anyway, I'm not sure how commercially viable my software would be if it requires manual operation by someone with appropriate technical knowledge.  I think I'd rather just make the software free and charge people money to go to their home(s) and run it for them.

 

I did recently use some automation to generate minimum phase FIR filters for my mains.  That was an experiment using a novel weighting scheme that yielded much better results than I expected.  Over the last few weeks, I've been researching ways of automating mixed-phase FIR filter calculations, but even after doing a lot of reading on the subject, I'm not getting warm and fuzzy feelings for any particular approach.  My plan now is to take a hybrid approach where I first identify the particular response abberation I wish to correct and then write code to help me design the filter.  I expect this process to be fairly tedious, but I want to be able to make a connection between the filters I apply, the resulting measurements, and the subjective change, if any.  Likewise in my approach to room treatment, I'm taking the approach of treating a particular problems and then evaluating the results rather than just throwing up a bunch of panels at the corners and first reflections points.  It's a lot more work for sure, but I find the benefits make it worthwhile.

 

As for my technical background, I'm taught myself computer programming and Linux admin while I was growing up.  Later in life, I went to college and opted to study another field figuring I was there to expand my mind and not to take the easy way out.  I studied chemical engineering all the way up to qualifying for a Ph.D., but I dropped out and finished a Master's after I became disenchanted with the whole thing.  I am very interested in computer physics simulation, and I got myself involved in that work.  Unfortunately, in many fields, the ability to generate pretty graphs is more important to success than making software that actually simulates anything correctly.  Despite all the hype, many many physical and chemical problems remain unapproachable using simulations.

 

When I left school, I moved back to the state my family lives in and took a job with a small internet company.  The pay is good, but sadly my education means nothing as I could have gotten it with just my skills.  Unfortunately there's not much chemical industry here other than oil, and being a bit of a hippy, I never had an interest in working for anyone in that industry.  Perhaps some day I will find a place to work that I'm happy with, but in the meantime, I took up audio reproduction and acoustics as a hobby to keep my brain busy.  I've always had a love for music and sound.  My inspiration came from buying an AVR with Audyssey MultEQ XT.  It was the first time I'd heard a digital technology that actually improved sound quality (albeit with caveats that I discovered later), and in fact, the improvement was greater than what I heard listening to many so-called audiophile setups versus my own crappy hand-me-down Bose speakers.  It clued me in to the fact that despite all the snake oil in the industry, excellent sound could be had for a fairly modest cost.  I've been hooked on the subject ever since and have not yet reached the point where I would say I'm seeing diminishing returns.

 

Anyway, this is about as personal as I've gotten on a Internet forum since I can recall, but I guess I've taken to this forum well.  I'm not even a member of the "3-15 Hz club", and my interests in audio span well beyond just the bass.  Nevertheless, I feel more at home on this audio forum than on any other, and I appreciate the surprising depth of knowledge possessed by many of its members.

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I always like this image for this purpose :)

 

attachicon.gifgjmky.jpg

 

That's a good one!  I'm not sure how accurate it actually is, but it's definitely funny.  I know my own eyeballs resonate in their sockets at around 18 Hz.  I recall reading recent research where infrasonic resonance of the eyeballs might account for many reported sightings of ghosts and other paranormal phenomena.

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SME,

 

I am glad you are contributing.  You seem to have a good scientific method to your experience in audio.  I also got really into things when I started measuring what Audyssey did to improve sound, only to find its limits shortly thereafter.  I am spoiled, though,  Dynamic EQ will boost 'chest-thump' frequencies so well, it makes a real difference sometimes.  It is the only part of Audyssey I miss.

 

JSS 

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Hmm.  Good question!  How close to the couch is the sub?  If the sub is placed adjacent to the couch, as mine are to mine, then part of the couch is probably seeing a much higher SPL than your listeners are.  So the vibrations are being transmitted from your sub to the air to the couch adjcent to the sub and then to your torso and butt.  I still don't think SIL/PVL has anything to do with it, apart from the extent that they correlate with SPL.

The subs are about 1 foot away from the couch and ~3.5ft away from the main LP.

 

I didn't measure a foot away from the couch, but if I did, the SPL would more closely resemble the close mic SPL of  the sub. The close mic output would typically be less output than the output further away because the measurement further away would account for room gain, where as the close mic response would be more room gain agnostic.

 

One thing you haven't quantified in your hypothesis is what you're calling "vibrations". These vibrations must be excited by the sound wave. My tests prove that SPL alone is not the sole contributor, so it must be something more...which leads us to the Sound Intensity hypothesis.

 

My test was a simple one.

 

If you rule out the vibration transfer function because of the concrete slab, and calibrate SPL as the same, it must be some other sound property. Given that physics shows SIL = SPL * PVL, it must mean the PVL would be the differentiating factor in this example.

 

I would like to hear alternative views that refute this hypothesis, but have yet to been provided any supporting evidence of the contrary.

 

Is there any additional insight that you can provide as to what's causing this? There aren't any other levers to pull that I'm aware of, which leads us to supporting the SIL hypothesis.

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What measurement program do you use?

 

JSS

If you want to use HDMI out and collect measurements from multiple channels with accurate relative offsets *and* do this on Windows without using some audio device that has a mixer app then I think you can use either jriver or equalizerapo to do this.

 

EqualiserAPO is on sourceforge - http://sourceforge.net/p/equalizerapo/- and it has a delay command and a copy command which you can use to copy the measurement signal to another channel and then add delay to that (maybe 100ms?). This should give you 2 (or more depending on how many channels you copy) separate, clearly identifiable, impulse peaks. You would then be able to see the relative offset between them as the delta between the actual peaks & the delay time you set in equalizerapo. You might then be able to extract these to separate measurements for subsequent trace arithmetic operations by copying the trace (using (A+B)/2 in trace arithmetic), setting a window appropriately to capture the separate impulses and then exporting each impulse. Alternatively, and probably easier, would be to capture the relative offset as above, take separate sweeps for each channel and then manually apply the offset you discovered earlier.

 

To do the same in jriver means using the jriver WDM driver as the target output in REW & then doing the copy/delay etc in some jriver DSP blocks.

 

equaliserapo is free, jriver is about $50 IIRC.

 

Another approach that can work is to apply separate delays to a low pass but equalise the delay to the high passes. HF peaks are easy to align so this gives you a quasi absolute time reference against which to measure the lower/slower LF impulse.

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So EqAPO will take the REW sweep and copy it to other channels prior to sending out via HDMI?  

 

Would another way to do it be to use REW's 'Loopback reference' channel as another active channel (instead of a loopback, just have it sent to another speaker) and see the resulting differences in the impulse peaks?  It sounds like it could work without having to install EqAPO, but the peaks may be too close to eachother.

 

Thanks for the software solution!  It will take a bit to get used to the program, but it can definitely add to REW's functionality for timing/delays.

 

JSS

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Would another way to do it be to use REW's 'Loopback reference' channel as another active channel (instead of a loopback, just have it sent to another speaker) and see the resulting differences in the impulse peaks?  It sounds like it could work without having to install EqAPO, but the peaks may be too close to eachother.

 

The problem is as you suggest, that the peaks are going to be close together, you only really have any delays that can be added in devices downstream (e.g. your AVR or a minidsp). Manipulating the signal on the PC is just quite a bit easier really (once you get it working).

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One thing you haven't quantified in your hypothesis is what you're calling "vibrations". These vibrations must be excited by the sound wave. My tests prove that SPL alone is not the sole contributor, so it must be something more...which leads us to the Sound Intensity hypothesis.

 

My test was a simple one.

 

If you rule out the vibration transfer function because of the concrete slab, and calibrate SPL as the same, it must be some other sound property. Given that physics shows SIL = SPL * PVL, it must mean the PVL would be the differentiating factor in this example.

 

I would like to hear alternative views that refute this hypothesis, but have yet to been provided any supporting evidence of the contrary.

 

 

One factor that I believe is being missed is that your SPL measurements for matching the 2 systems output are taken at a single point in space at the microphone element and in no way take into account the tactile vibrations which are the actual function being looked at. The phone was being set on the couch for the vibration measurement...The couch is a very large piece of furniture occupying a significant volume of space in the room relative to a microphone. In effect the entire couch is the microphone at that point. When you sit on a couch you are now in contact with a very large surface area of the couch. The subs were in very different parts of the room and affected by the room reflections and boundaries much differently. The total amount of energy from each sub system physically imparted into the whole of the couch as vibration will be much much different. This explains the difference in tactile sensation. The near-field sub would have a much more focused and directed energy on the couch than a sub 15 to 20ft away. The 2 subs may have had the same SPL at the spot measured but what about back behind the couch in front of the near sub? Near the floor on the left front corner of the couch, what about the back right corner? Under the couch in the middle? That's what I'm saying you would need hundreds of SPL measurements and a few accelerometer placements to get anywhere close to understanding the total energy impacting the couch from each sub system and calibrating them to each other.

 

A single point SPL measurement where the phone will be placed on the couch simply does not have the resolving power to look into the vibration issue.  In a complex acoustic environment with direct physical contact with other objects you will have the audible component of sound and also the physical sensations. The two are separate issues to a large extent on that we agree. However I'm not yet convinced that it has anything to do with particle velocity or sound intensity.

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FIX THE CROSSOVER IN THE SPEAKER, DON'T ADD MORE COMPLEXITY AND ADD MORE TO THE SIGNAL CHAIN AND NOISE FLOOR TO FIX IT.  Fix the root problem. 

I think SME made the point I was going to make (about how adding DSP can be a simpler and/or more practical and/or more cost effective solution then revisiting the speaker itself or the entire room setup). I also don't understand the resistance to DSP in the audio chain, a speaker is a mechanical device after all so why not use DSP to improve it?

 

In practice though the key issue, with respect to the applicability of DSP to the signal chain, seems to be that modern prepros are just so lamentably weak. The basic design of a prepro hasn't changed 10+ years while the amount of computing power in there has barely changed in that time too. This means add on boxes of varying quality OR accepting the limitations of a PC only approach OR spending a fortune on a trinnov. 

 

I thought this site had some interesting possibilities - http://www.servobass.com/Blog.html- though nothing seems to have come to market yet. The DIY DSP programmable servo controller (for a sub) looked especially promising but I think they pulled that.

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The total amount of energy from each sub system physically imparted into the whole of the couch as vibration will be much much different. This explains the difference in tactile sensation. The near-field sub would have a much more focused and directed energy on the couch than a sub 15 to 20ft away. The 2 subs may have had the same SPL at the spot measured but what about back behind the couch in front of the near sub? Near the floor on the left front corner of the couch, what about the back right corner? Under the couch in the middle? That's what I'm saying you would need hundreds of SPL measurements and a few accelerometer placements to get anywhere close to understanding the total energy impacting the couch from each sub system and calibrating them to each other.

 

A single point SPL measurement where the phone will be placed on the couch simply does not have the resolving power to look into the vibration issue.  In a complex acoustic environment with direct physical contact with other objects you will have the audible component of sound and also the physical sensations. The two are separate issues to a large extent on that we agree. However I'm not yet convinced that it has anything to do with particle velocity or sound intensity.

Let's define the energy that you're referring to.

 

Sound Energy is a form of energy associated with vibration or the disturbance of matter.

 

Sound Power is sound energy per unit time.

 

Sound Intensity is sound power per unit area

 

Therefore, Sound Intensity can be said to be sound energy per unit time/area.

 

Is Sound Intensity not what is causing the vibrations in the couch? I'm trying to understand why SIL would not be a factor?

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I'm not sure what I'm supposed to be seeing here.  The response at the seats looks better.  I have no idea what's going on with the close mic because one of the curves has smoothing and the other does not.  I don't doubt that it's less flat, but I don't see that as being a problem except for people listening with their heads right next to the woofer.  In fact if the peak is truely modal, then the correction will likely improve response across the room on average because the peak part of the mode covers more room area than the dip part.  And you get back some headroom, not that it's likely to make much of a diference with a Linkwitz Transform in the chain.

 

 

Fair enough.  You could always try to boost back the lost low-end, but this may not be desireable depending on whether the filters have enough precision for such low frequencies and your needs with respect to gain structure and noise floor are met.  This work always involves compromises at some point.

 

 

If you're not sure what you're supposed to be seeing in that pic, in reference to your question, then there's not much sense in my continuing with the discussion.

 

Smoothing on a close mic measurement of a sealed sub is irrelevant.

 

The EQ "fix" grossly distorts the input signal, causing problems you apparently choose to ignore. It makes the FR flatter but so does the method Max and myself have mentioned.

 

Butchering the input signal with EQ adds no headroom. Headroom resides at the weakest point, which for any sub is <20 Hz. What happens to your response and headroom if the hard limiters squash the output into a completely different FR every time there is <20 Hz content?

 

Alter the native response of the sub, move the sub, move the seat, treat the room, change the cross point, adjust relative phase... voila!, no EQ needed. Or, skip those steps and just insert and apply ham-fisted EQ and pretend there is no price to pay... that it instead fixes everything. Hey, that's cool with me because what choice do I have in the matter? The majority use quick EQ, tout it's sonic improvement benefits and call it a day. But, I remain unconvinced by that sort of evidence.

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Let's define the energy that you're referring to.

 

Sound Energy is a form of energy associated with vibration or the disturbance of matter.

 

Sound Power is sound energy per unit time.

 

Sound Intensity is sound power per unit area

 

Therefore, Sound Intensity can be said to be sound energy per unit time/area.

 

Is Sound Intensity not what is causing the vibrations in the couch? I'm trying to understand why SIL would not be a factor?

 

 

Sound intensity and sound energy are often very closely linked with SPL (dB) and each other.

 

The entire point of my post was that the "matching" of spl of the 2 subs at a single point in a vast 3 dimensional, complex acoustic space, while having both subs in radically different positions within that space, is a data point of very little relevance for the exercise being undertaken. It simply doesn't have much at all to do with the amount of total energy and vibration being put into the couch by the individual speakers. You can call that energy whatever you want, but the point is that to imply that the SPL levels surrounding and impacting the couch were level matched between the subs so that regular old sound pressure level is removed from consideration as a cause of the difference is not correct.

 

Move the mic 3 feet to the left and the SPL will not be the same between the 2 subs.

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If you place a subwoofer very close to a large coach, the coach will also change the soundfield because it is large and too close to the source.

 

However, @dom's experiment shows an attempt to verify an observation by objective measurement - the coach vibrates more with the subwoofer close up.

It is easy to show mathematically that the soundfield has a larger intensity and a larger velocity, relative to pressure, close to the source. 

 

Practical experiments as in attempts to make an overall improvement in sound quality seems variable.

Some think it is awesome - the coach vibrates more, some find there is no overall improvement.

 

I think this varies a lot depending on room, system configuration.

 

I recently did an experiment with this - placing two subwoofer units very close behind the sofa.

The best result was with units running out of phase with tuned delay.

It did not add any good sensational contribution, as the sofa vibrates more, but not in a good way, simply more of the wrong vibrations.

This is of course very dependent of the construction of the sofa, which will determine its resonance pattern.

It was impossible to tune the complete system into a decent impulse response, so the overall sound was very compromised compared to the current normal set-up.

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Dom, I feel your pain. This thread was created to focus on setup and EQ, not SIL/PV. There should be another thread started just for Dom's subject.

 

The thread OP asked for "details on how YOU set up your system...". In 2004, 11 years ago, I published a setup guide that's still hanging around the interwebs somewhere. The methods of verification have grown in sophistication with computer power, freeware and hardware, but I still use the same steps in the same order, with EQ last, as Max posted earlier.

 

As I've said, I don't want to argue whether other methods are better, especially methods that include specifics that can't be revealed for whatever reason. I just showed my method, reasoning and results.

 

I just conducted a simple experiment to clarify my position on my setup method.

 

Here is a FR at the seats with an obvious problem. I used a simple delay setting adjustment to rectify the problem and have shown the signal and the result. I then injected an outboard digital PEQ and used 2 filters to rectify the problem and have shown the new signal and result, and finally, I've compared the resulting input signal and results measured at the seats to illustrate why I do not use post smoothing EQ.

 

02dbd44e80a2a9c1b9ccd1cdb103613b.png

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If you place a subwoofer very close to a large coach, the coach will also change the soundfield because it is large and too close to the source.

 

However, @dom's experiment shows an attempt to verify an observation by objective measurement - the coach vibrates more with the subwoofer close up.

It is easy to show mathematically that the soundfield has a larger intensity and a larger velocity, relative to pressure, close to the source. 

 

 

 

 

I agree. What I am saying is for the purposes of that experiment there is an elephant in the room that has been overlooked from what I have read so far.

 

Let us say that you have one subwoofer directly behind a couch about 1ft away and firing directly into the back of it. You also have another sub 15ft away from the couch in a front corner. Now lets say that you place a microphone to measure the SPL from each at the headrest position of the middle seat on the couch. You run a 20Hz tone through each sub and set the recorded SPL of each to 100dB. What happens when you move the microphone 4ft to the right and sit it on top of the right corner of the couch and recheck the levels of the two subs? Will it be exactly the same? Most likely not. We can't know for certain without actually measuring it. What if we move the mic all of the way across the back of the couch to the opposing position, or place it near the floor in the front of the couch? We cannot say how the 2 subs relative levels will change without measurements.

 

Now in this same scenario let's say we then move the mic back behind the couch directly in front of the sub firing into the back of the couch and again measure the relative SPL levels. I think we can reasonably guess that the close sub is going to have much higher SPL levels relative to the corner loaded sub across a large portion of the back of the couch. Causing the couch to vibrate more. Yes it will also exhibit higher intensity, PV differences and all that as well, but the SPL is also much higher and more evenly distributed from the near field sub into the back and probably bottom of the couch. We cannot reasonably deduce which effect is causing this at the exclusion of everything else (regular old SPL) with this type of experiment IMHO. There are too many variables.

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I just conducted a simple experiment to clarify my position on my setup method.

 

Here is a FR at the seats with an obvious problem. I used a simple delay setting adjustment to rectify the problem and have shown the signal and the result. I then injected an outboard digital PEQ and used 2 filters to rectify the problem and have shown the new signal and result, and finally, I've compared the resulting input signal and results measured at the seats to illustrate why I do not use post smoothing EQ.

 

You appear to avoid EQ for 2 reasons; the device you're using adds a rolloff & that sub 10Hz response is critical to you, you have enough subs distributed around the room in order to not require much EQ.

 

I would argue this is relatively uncommon set of requirements & would agree that EQ, via an offboard device that adds a rolloff, is clearly unnecessary for you. 

 

This doesn't mean EQ in itself is a "bad thing", it just means it is not appropriate to your use case. 

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You appear to avoid EQ for 2 reasons; the device you're using adds a rolloff & that sub 10Hz response is critical to you, you have enough subs distributed around the room in order to not require much EQ.

 

I would argue this is relatively uncommon set of requirements & would agree that EQ, via an offboard device that adds a rolloff, is clearly unnecessary for you. 

 

This doesn't mean EQ in itself is a "bad thing", it just means it is not appropriate to your use case. 

 

There's more to it than roll off. There's an unnecessary analog =>digital =>analog conversion, the assumption that the DACs don't matter, increase in noise floor, increase in distortion, loss of headroom, the difference in input signal and...roll off. See the thread about clipping and the resultant huge increase in THD, for one example. Anytime you add an additional component, all of these things must be considered and usually are not.

 

Certainly this doesn't mean that EQ is a bad thing, but it's being presented as a cure all by SME while parts of my stated position were referred to as nonsense. I'm inputting the downsides of using EQ as a cure all and leaving it to others to show convincing data to support the contrary claims made.

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