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Bass punch threshold


TTS56A

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Amount of tilt also depends on room acoustics - more live rooms will have and also require more tilt.

 

Working on a new room now, and here is how the bare room with very little furniture and no acoustic treatment performs.

The speaker itself is flat - as can be seen on the 1m graph, the fall-off at lower freqs here is due to boundary interference, which is not eliminated even this close to the speaker.

At the listening position the response is tilted, and for this speaker and lp position there is a huge suckout in the mid bass range, so no bass.

 

Reflections destroy tactile feel, we know that, so it is expected that the tactile response from this set-up is lacking.

However, as tactile feel also relates to spl, you can just turn it up, and then you get solid tactile feel from upper bass up far into the midrange, but then it is way too loud.

 

The solution is of course to fix the room, so that reflections and decay is in control, and also add a decent bass system as the speakers are designed to work from 60hz and up.

 

 

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That 2013 Fireworks recording is clipped.

 

JSS

Yup, just like his other fireworks.  IIRC, the worst cases happen at about the same time into both tracks.  They are still amazing tracks to experience with a lot of headroom.

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Awesome post, Kvalsvoll!  That suck-out from 55-75 Hz is a real drag.  It's harder to address frequencies that low with room treatments, but you'll probably want to if you are going to cross at 60 Hz.  Big nulls like that usually come with steep phase changes which make crossovers all the uglier.

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I think group delay is another big deal... Bass frequencies are always deleyed compared to highs, Even in well damped sealed box, lower frequencies come a bit later. Let's look to a 100 Hz 24db/oct LR crossover, wich introduces a not negligible quantity of GD... Then kick foundamental will be time delayed related to its armonics

 

You can see, 20-80 Hz range is delayed by almost 6 ms, that's of course independent from the speaker position!

 

fr.jpgfree image hosting

 

gd.jpgurl immagine

 

A closer looks to the step, the first impulse comes from tweeter, the last one is the woofer... Remember, a perfect phased step looks like a square wave.

 

step.jpgurl immagine

 

What if, if we try to compensate for delay? Then delaying main speaker about 6 ms. Well i think it's not possible without compromising the frequency response at crossover point. 

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Of course FIR filters don't introduce phase shift, but pre-ringing become a fact...

there's certainly a tradeoff (between pre and post ringing), most of the aforementioned tools provide ways to achieve that so that the result has no meaningful preringing (or more specifically, the amount of preringing that you want to allow).

 

http://www.bodziosoftware.com.au/pre_post_ringing_ir_and_pulses.pdf(by the guy behind Ultimate Equaliser & Soundeasy) might also be interest for another view (on relative audibility)

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Thank's for the pdf  :)

 

Conclusion 2 Typical crossover implementation: Low-Pass and High-Pass components will cancel each other’s pre-ringing, if the filters are of the same type and slope. This is illustrated on Figure 20.

 

However this conclusion is not quite right, because high and low pass components will totally cancel each other's pre-ringing only if there is no minimum offset between drivers, and if the lp is prefectly on axis. This is quite hard to get anyway. Then in real world pre-ringing cannot be totally eliminated.

 

 

This is an example, the same 24LR but linear phase with no offset (looks like a perfect step)

 

step1.jpghost immagini

 

 

 

Then the same but with woofer offset about 0,1 ms wich means less than 1,5 inch in deeper

 

step2.jpgfree image hosting

 

As you can see, pre-ringing is about 20% of the initial impulse. Of course this may or may not be audibile, however it can't be totally avoided.

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.. That suck-out from 55-75 Hz is a real drag. 

 

Yes, and it is difficult to fix on this room.

 

Here this is caused by the position of the listening position, moving speakers does not fix it.

 

It is not possible to add absorption to the back wall - it is already partly open, the width of the room is too narrow to allow really thick side wall absorption, and it is not practically possible to move the listening position out of the null.

 

The bass system must be on the front wall behind the main speakers, so no help from there.

 

In a more normal room it would be possible to fix this by adding absorption to the back wall.

After adding front wall absorption and ceiling and side wall absorbers, I will see how it performs, and then there are some tricks that can be done with the bass system.

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Yes, and it is difficult to fix on this room.

 

Here this is caused by the position of the listening position, moving speakers does not fix it.

 

It is not possible to add absorption to the back wall - it is already partly open, the width of the room is too narrow to allow really thick side wall absorption, and it is not practically possible to move the listening position out of the null.

 

The bass system must be on the front wall behind the main speakers, so no help from there.

 

In a more normal room it would be possible to fix this by adding absorption to the back wall.

After adding front wall absorption and ceiling and side wall absorbers, I will see how it performs, and then there are some tricks that can be done with the bass system.

 

Your room sounds about as awkward as mine - the best layout for living is not the best layout for sound!

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Your room sounds about as awkward as mine - the best layout for living is not the best layout for sound!

 

Yes, practical functionality is always #1, if not, nothing makes sense.

 

But it will have great sound when it is finished, even with the given constraints. 

Good bass may not be possible with full-range mains only, but that is what we have bass systems for.

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...

What if, if we try to compensate for delay? Then delaying main speaker about 6 ms. Well i think it's not possible without compromising the frequency response at crossover point. 

 

Good post.

 

In practical situations, though, you will find that it may indeed be possible to get  a reasonable response after compensating for this delay, and it will sound better, even with some minor faults in the steady-state freq response.

 

Also, due to both room and speakers, the actual roll-offs will be different.

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Great thread.  Kvalsvoll, no way to add subs to the back of the room to help cancel the worst nulls?

 

I have played around some with biquad implementations of BW active crossover filters on transient signals and what the phase response does to the signal itself after re-summation.  6dB/octave crossovers do not appreciably change the signal at all.  LR2 is just slightly worse, LR4 is not nearly as good as LR2 and BW1.  LR8 is a demolition of the original waveform.

 

I can only tell the difference between the different x-overs when played back-to-back in succession with headphones on in brief testing.  Have not tried the test on the big system.

 

JSS

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Great thread.  Kvalsvoll, no way to add subs to the back of the room to help cancel the worst nulls?

 

No way to fit subwoofers on the back, due to room layout and a fireplace.

And it would not necessarily help, in this particular case.

With some tricking with delay and phase, it would help, though.

 

In the other room I have 2 units at the back, to cancel the suckout around 50-60hz, with proper delay set individually for all subwoofer units, this works.

But that room has a sealed back wall.

The trick here is to set delays properly so you don't cancel the sound intensity/particle velocity at low frequencies.

 

But wait and see, now.

The front wall will be finished tomorrow, then I just need to build some absorbers.

If the acoustic treatement can fix the room for say 100hz and above, this will sound great, and allow pleasant listening at louder volumes because the decay and resonances are fixed.

 

This is quite similar to what most people have to live with - the room is what it is, you just don't knock down walls or place your listening position in a doorway just because it will give better sound, you have to work within some reasonable compromises.

But if you do it the right way, the end result will be very good.

 

The frequency response pre-fix seems like it is heavy in the low end - too much bass/lower midrange, and the impulse response measures better than most studios - must sound rather dark, and almost anechoic?

But it does not sound like that.

At lower volumes, it is tolerable, and the 3d presentation is almost like everything is real - voices are not only clear but also appear like physical objects.

Drums and transient instruments are physical objects placed in a 3d landscape, it is like you could get up and walk around the different instruments.

Attack and dynamics and tactile feel in the midrange is great.

And, there is no bass - not talking about missing the 20hz and below, this IS no bass, below around 80hz.

But the bass that is there, sounds great, kind of like you can feel it - tactile - before you hear it.

 

But when you turn it up, to get more of this, it gets too massive, like there is too much midrange and higher freqs, and all the resonances gets really annoying.

And, there IS NO BASS:

 

Here are some more measurements of this - as it was, before the fix.

And yes, this is a great thread.

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post-181-0-18662000-1460424285_thumb.png

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..

I have played around some with biquad implementations of BW active crossover filters on transient signals and what the phase response does to the signal itself after re-summation.  6dB/octave crossovers do not appreciably change the signal at all.  LR2 is just slightly worse, LR4 is not nearly as good as LR2 and BW1.  LR8 is a demolition of the original waveform.

 

I can only tell the difference between the different x-overs when played back-to-back in succession with headphones on in brief testing.  Have not tried the test on the big system.

 

JSS

 

This is an interesting experiment.

I have done tests for phase distortion, with low frequency transient signals, and found that phase shift at low frequencies can be audible.

But if I remember correctly, you would observe a visual change of the waveform long before it is audible.

 

If the phase shift - which is a delay - is large enough, it will obviously be audible.

But where is the limit.

And this limit is likely to be frequency dependent.

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No tricks, mostly doing it like described in the "How to.." article, but this room is different from the simpler examples, so the solution is a little different.

I took measurements, and pictures, for an article, all will be presented. 

 

I intend to show in detail what and how everything is built, even detailed construction drawings you can take to a local shop/carpenter.

 

The tricks will be the bass system, but that will be later.

 

And of course I should have done the velocity measurements, but I did not think of that for the room.

Would have been very interesting, actually, but it is too late now when the new front wall is up.

But I will make those measurements from now on, thanks for reminding.

 

This article will not be written before everything is done and completed, excluding the bass system.

But I can post some measurements and things, so that we have something to discuss, while at it.

 

Yes, this is the before waterfall.

Reveals a nasty 100hz resonance - not difficult to hear.

You see that it generally gets worse as frequency goes lower, but there is also a power-response problem in the upper midrange, there is too much energy in the decay around 1k-4k.

The radiation pattern and power response of the speaker is what makes it as good as it is, keep in mind there is no absorption in the room, and little furniture.

 

I think I have a picture.. just don't look at the speakers, they are not presented yet, it is a prototype with no grills and lacking some of the design features.

 

post-181-0-79860100-1460432267_thumb.jpg

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I'm currently running a 24 dB/octave cross for subs vs. MBMs. at 50 Hz, in addition to 24 dB/octave crossovers at 80-110 Hz.  So I basically have more crossover-induced group delay than is typical.  I'm also in here talking about how awesome my system slams.

 

I'm not saying that this group delay doesn't matter.  I have actually been planning to implement linear phase crossovers and doing some A/B work eventually because I want to see if I notice a difference.  I'm thinking it might be audible but only just barely.  And in fact, the difference might be unnoticed in a more typical listening environment where there are likely much more serious response problems in the crossover range.

 

That may seem to contradict earlier arguments I've made about ensuring more energy is concentrated in the initial impulse, but it's not exactly the same.  If that initial impulse is skewed in time just a bit, that's a lot better than if the initial impulse is weak and the short-time gated SPL is low but the un-gated SPL is still high because of reverb and/or modal build-up.  In the latter instance, the initial impulse is weak and there is a lapse in time before the bulk of the energy arrives.  That's undesirable.

 

OTOH, the impulse being skewed in time may not be a big deal at all.  Why not?  Simply put, we are already well habituated to it.  How much content do we all consume that likely has bass boosting shelves applied using minimum phase filters?  These introduce some delay for the lower frequencies but we still happily listen.  It gets more interesting.  The speed of sound is actually frequency dependent.  High frequencies travel faster than low frequencies.  According to this Wikipedia article, the change is most substantial over the 10-100 Hz range over which the speed increases by about 0.1 m/s.  Indoors, that's completely irrelevant, but when outdoors at great distance, this phenomenon could add a lot of delay to the deep bass.  When watching fireworks at say, 1 km away, the delay would be about 30 ms.  Do you notice a "lack of slam" from that distance?  I sure don't.

 

But of course, curious minds want to know if group delay matters.  I'll surely post something when I get around to trying linear phase crossovers.

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Yes, this is the before waterfall.

Reveals a nasty 100hz resonance - not difficult to hear.

You see that it generally gets worse as frequency goes lower, but there is also a power-response problem in the upper midrange, there is too much energy in the decay around 1k-4k.

The radiation pattern and power response of the speaker is what makes it as good as it is, keep in mind there is no absorption in the room, and little furniture.

 

I think I have a picture.. just don't look at the speakers, they are not presented yet, it is a prototype with no grills and lacking some of the design features.

 

attachicon.gifrom2 pre-fix med F2.jpg

 

Forgive me for indulging my inner geek here for a bit, and of course note that as far as advice is concerned, YMMV.

 

When I saw your IR data, I knew right away you were running horns.  This picture confirms it.  (I know, I know, don't look at the speakers!)  The IR or ETC is very useful as a first pass for identifying reflections.  However, it also emphasizes the high frequencies much more than the low frequencies.  In fact, for assessing low frequency reflection points, it is pretty much worthless.  I actually use the spectrogram function in REW a lot for assessing room acoustics.  Start with a small window size like 2 ms or even 1 ms.  Set the display time range to about 10 times the window size and look at the results.  Then increase the window size and reset the display time.  Larger window sizes give more insight into lower frequencies and longer ringing resonance in the high frequencies.  Smaller window sizes give you more time resolution, which makes it easier to see very early reflections and better pinpoint the time of arrival of the reflection.  Once you get up to a window sizes of 25-50 ms, you'll have a nice picture of your power response decay.  I like the spectrogram for this more than the "RT60" feature.  In fact, I don't use the RT60 feature much at all because it feels quirky and unstable to me.

 

I can definitely see that space giving you a lot of trouble.  Is there any possibility of employing a compact near-field sub or MBM, disguised as an end-table?  I know, I know.  The answer is probably "no", so I will continue.

 

One major issue is that the room height and width appear to be almost the same.  Not will the modes overlap but the early reflections will be stronger.  Some of these may cause "extra" (i.e. non-minimum phase rotations, which sound like trash and totally evade treatment with minimum phase EQ.  Having the woofers are opposing sides of the wall will help manage the width-wise mode.  However, the side-wall reflections from both woofers will still cause suck-out at frequencies somewhat above the modal one.  Assuming a room about 9-11 feet wide, this is very consistent with the nulls you see.

 

As far as treatments, it doesn't look to me like you need much, if any high frequency absorption.  This is to be expected using horns.  I wouldn't give too much concern to the RT60 plots.  I suspect the tool is unreliable.  You have to ask yourself why you should be seeing blooming upper midrange if your speakers don't have that kind of power response.  Does your room somehow selectively absorb high and low but not upper mid frequencies?  The answer is, probably not.  Perhaps some low frequencies in the 50-500 Hz range may escape through the thin wall(s) and windows but not a whole lot.  Above there, the walls are probably largely reflective until you push up into the highest octave.  Knowing your speaker, and your power response, do *you* think you should be seeing too much midrange?  Now, go look at the spectrogram like I suggested and see if you agree with me.

 

I haven't seen your room spectrogram, but I'm thinking you'll want to mainly use absorbers with membranes.  Mass loaded vinyl will deflect high frequencies and enhance low frequencies absorption, which is crucial in such a confined space.  On the ceiling, go with thicker lower density treatment, spaced some distance from the wall, if at all possible.  How much space do you have on the one sidewall?  This is trick because you have the window on the other side and you may not want to break the left-right symmetry for mid-range frequencies by putting absorbers only on one wall.  Instead, you may consider trying to acquire tuned membrane-based traps from a reputable vendor (admittedly tough to pull off for you in Europe) to try to absorb as much from the 50-80 Hz range as possible.  The benefit of these traps is that they can be made to a smaller depth than broadband traps with similar capability.

 

One more question is whether you could provide "corner-adapting"  end-tables for the speakers that might help to enlarge their baffles and better integrate them with the walls.  A benefit I see with this is better low mid-range directivity and a reduction of the accumulation of higher order reflections between the side-walls by directing more of the sound toward the rear part of the room.  Then you could mount bass traps above the speakers, leaving little nooks.  :)

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I can only tell the difference between the different x-overs when played back-to-back in succession with headphones on in brief testing.  Have not tried the test on the big system.

 

JSS

 

I did a lot of experiments like this, and what i can say is that phase shift is most noticeable with in-ear headphones wich have a very flat phase response over the band. Though I always heard more difference when reproducing high dynamic contents... However to be honest i never noticed audibile difference with 12 and 6 db/oct, rather LR4 and higher become quite audibile.

 

I also noticed that the maximum shift is near the xover point, then if the kick foundamental is close to xover point, will be more smeared. That's the same for the snare, if the xover is tuned a bit higher (in the 200 hz range).

 

Here an example of well dry recorded punchy track.

https://www.song365.biz/track/yellowjackets-dewey-a-tribute-to-miles-davis-377579.html

 

And this is the waveform of the kick drum. You can see the orginial waveform on the L channel, and then the same through a LR4 crossed at 80 Hz on the R channel.

 

kick.jpghost image

 

The effect of phase shift is delaying the foundamental, and the intensity of the attack is spreaded over time. But as i said this will come udibile under specified conditions, for example absence of reverb, no steady waves, and no phase shift through the entire band. Well i think a few rooms in the world can meet these requirements.

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That may seem to contradict earlier arguments I've made about ensuring more energy is concentrated in the initial impulse, but it's not exactly the same.  If that initial impulse is skewed in time just a bit, that's a lot better than if the initial impulse is weak and the short-time gated SPL is low but the un-gated SPL is still high because of reverb and/or modal build-up.  In the latter instance, the initial impulse is weak and there is a lapse in time before the bulk of the energy arrives.  That's undesirable.

 

I agree with this! However i think punch is more related to the thiny part of the mid-bass wich needs to be perfecly time aligned with the rest of band, rather than slam. As you go in deep more delay is allowed, it's common to get 10-20 ms < 40 Hz, but it will be badly if such delay is going around 100 Hz.

 

I never noticed that sound speed changes with frequency, that's seems to be a bit strange, anyway i'm going to inform me better  :)

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