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Discussion Starter #1
I have a question on REW, after the measurement and when I enter the information under the Target Settting tab for the eq, does this look correct? Specifically the Target Level DB? Is there a certain db level that should be or do I adjust it to where its in the middle of my reading on the left?


Also on the crossover part of the settings are those correct?


I have the crossover on the inuke set to HP 20hz, LP 120hz and the Crossover type its Butterworth qith 12db slope.

 

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Discussion Starter #3

Quote:
Originally Posted by chalugadp  /t/1523218/correct-settings-for-rew-target#post_24500837


Is there a question ? Just posting ..

Lol, For some reason it wasn't letting me post a picture from my ipad.
 

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I thought it was normal to use a steeper slope on the HP. This is for the MARTYcube, right? I thought LTD was recommending 4th order, but maybe I'm making that up.


Why do you need a LP at all - aren't you feeding it from a bass-managed AVR? If you're trying something special for mains integration, that's one thing (adding a filter here, since the filter in the AVR is likely fixed) - but I would start without it.
 

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All good ,

Just hit filter tasks below and match target to filter.

It will create filters , then hit eq filters on the top and there they are.
 

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Discussion Starter #6

Quote:
Originally Posted by chalugadp  /t/1523218/correct-settings-for-rew-target#post_24500885


All good ,

Just hit filter tasks below and match target to filter.

It will create filters , then hit eq filters on the top and there they are.

That is what I thought, but since I had so many peaks and valleys I wanted to make sure I didn't do something wrong. So the 90db target is good or is it good because it puts the target curve right in the middle of my "mountain range"?
 

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Discussion Starter #7

Quote:
Originally Posted by HopefulFred  /t/1523218/correct-settings-for-rew-target#post_24500883


I thought it was normal to use a steeper slope on the HP. This is for the MARTYcube, right? I thought LTD was recommending 4th order, but maybe I'm making that up.


Why do you need a LP at all - aren't you feeding it from a bass-managed AVR? If you're trying something special for mains integration, that's one thing (adding a filter here, since the filter in the AVR is likely fixed) - but I would start without it.

You correct, this is actually an old picture and someone had already mentioned the part about not using the lp filter at all since my receiver is set to 120 already. (I have small mains
)


What is this 4th order you speak of? And as far as the slope you mean to go with something like 24 instead of 12?
 

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Quote:
Originally Posted by cessna1466u  /t/1523218/correct-settings-for-rew-target#post_24500913


And as far as the slope you mean to go with something like 24 instead of 12?
That's what I'm saying, but I'm not sure it's right. I think you need to know what excursion vs power looks like below tuning to know how much needs to be dialed back. I'm sure LTD has posted that somewhere. Of course, even if I had the graph for your driver, box, and amp in front of me I couldn't make a confident recommendation. I just had in my head that he was recommending steeper to protect the driver.


Think about it this way (I'm making these number up), if 1000W gets you to max excursion at 16Hz - a 2nd order BW filter limits you to how many watts at 8Hz? (You'll be one octave down, so that's -12dB. -3db would be 500W, -6dB would be 250, -9 at 125W and -12 at 63W - I think that's right) So how many watts does it take to bottom your driver in your box at 8Hz?
 

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Discussion Starter #11

Quote:
Originally Posted by HopefulFred  /t/1523218/correct-settings-for-rew-target#post_24500996


Looks like I was misremembering - but the premise is important. John says 2nd order should do, but that's a qualified statement (this box, this driver, this amp - not sure how yours compares). http://www.avsforum.com/t/1516724/martysub-flatpacks/90#post_24335690

Ok so when you say 2nd order or 4th order is that the name of if? I am confused, is 2nd order the Butterworth and 4th order the other one like Linkwitz?
 

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Discussion Starter #12
T
Quote:
Originally Posted by chalugadp  /t/1523218/correct-settings-for-rew-target#post_24500977


Your only problem area is the null from 60-68hz. Add a 4-5 db boost at about 63hz and see.

Thanks Donny, as I mentioned that is an older graph that I was using for reference. At your recommendation I did the boost a while back and it helped but I was just checking, its that part of me that always thinks something is missing or wrong.
 

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Quote:
Originally Posted by cessna1466u  /t/1523218/correct-settings-for-rew-target#post_24501098


Ok so when you say 2nd order or 4th order is that the name of if? I am confused, is 2nd order the Butterworth and 4th order the other one like Linkwitz?
Linkwitz-Riley and Butterworth are two (of a few) types of filters, which differ in technical ways relating to phase and amplitude response that I can't describe (I am not a doctor
). First order is 6dB/octave, second order is 12dB/octave, third is 18, and so on. http://en.wikipedia.org/wiki/Audio_crossover#Classification_based_on_filter_order_or_slope
 

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Quote:
Originally Posted by cessna1466u  /t/1523218/correct-settings-for-rew-target#post_24501155


Thank you
You're welcome. and my novice brain is happy to try to explain more fully if you're interested. (I was a high school science teacher in a previous career, so I kinda have a weird freakish soft spot for the technical mumbo jumbo.)
 

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Discussion Starter #16
I knew it! When you ended one of your posts with "So how many watts does it take to bottom your driver in your box at 8Hz?" I told myself, I think this guy is a teacher.



My favorite subject in highschool was science, it wasn't because I liked science, I was actually terrible at it. It was because the teacher was really cool and creative. He wore an airbrushed (1992) lab coat with a mad scientist on it and always had quirky things to say and ways to teach us.


Anyways, I would love to learn more about it. I never ask more than I need only because it seems like its one of those things that you have to of learned either at school or through hands on after many years of dealing with it. It doesn't seem like something that someone can teach you in a forum with a few posts. With that said thought, I have learned more here in the last month since I started my DIY journey that in the last almost 20 years of dealing with electronics and audio equipment. There is a A LOT of knowledge flowing around here. Some of the guys here must do this in their professional life because the stuff they know is insane, like LTD, he seems to be the go to guy here. So I imagine this is his life.


As far as learning, if you are wanting to explain, how about explaining the slope thing? Is it doing exactly what I am seeing it do when I change it on the crossover. It seems to drop down from 40hz or so to 20 more drastically as the order gets higher. So does that mean that as a frequency is being played if it gets close to 20hz which is the lowest my sub will do, it slowly lowers the dbs so that it isnt at regular volume/spl when it gets to 20hz? So if I go with 6db slope it will start to lower closer to that 40hz range than then when I choose 48db slope which seems to lower the volume/spl closer to about 27hz. Does that make sense?
 

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"A filter is not a brick wall" as they say... a perfect speaker or sub would put out the same loudness at any frequency given the same power input - right? 1 watt at 20Hz would produce the same sound pressure level as 1 watt at 50Hz or 5000Hz - of course there is no such loudspeaker. Since each speaker or subwoofer only offers a reasonable approximation of that over a small range of frequencies (bandwidth), it's important to use multiple drivers together to get a good (flat) response over a wider bandwidth. What's more, in some cases, the output can increase dramatically in certain ranges undesirably - with lots of distortion or whatever - we would want to filter that out. So we use filters to limit a loudspeaker or a driver within a multi-driver loudspeaker to only reproduce sounds over the range we have decided is desirable. Each driver only gets a portion of the whole signal - the rest of the signal is filtered out.


Filters have a few characteristics - the simple ones (that I understand) are cut-off frequency (or center frequency in some cases, like for EQ) and slope. The cutoff frequency or f3 is the frequency where the filter has cut the power in half - that is to say, reduced the output -3dB compared to the unfiltered signal. The slope is the rate at which that trend of decreasing power continues.


Let's start with a classic THX low-pass filter: it is called low-pass because it allows the low frequency signal elements to pass - removing the high frequency energy. It's 4th order with an f3 of 80Hz. We apply this low-pass filter to a subwoofer signal, usually in an AVR. The signal will be full strength from very low (ignoring any high-pass filter that may be applied, as in your case with the Martycube, for now) and just before 80Hz the signal will begin to diminish. It might be imperceptibly diminished at 65 or 70Hz, but the slope will be obvious by the time the signal is down 3dB at 80Hz. From that point onward (when tracing the plot to the right, toward higher frequencies) the slope will be constant. And every time the frequency doubles, the signal strength diminishes -24dB. At 160Hz, the signal is down a total of 27dB from where it was at 30Hz or 40Hz (that's the original -3, plus an octave's worth of -24dB). And that's -24dB, because it's fourth order.


Let's consider some of the implications of that LP filter. Say you built a horn sub with a really wild resonant peak just outside the intended bandwidth (this is pretty common for horn loaded subs). The response might be pretty flat up to 100 or 110 Hz, but let's say its up about 30dB at 115Hz. That sort of peak will be really obnoxious, so you will want a high order filter on that kind of sub. If the filter were only 2nd order, the signal would be down by very little compared to the classic THX LP filter. With a second order LP, we should expect the signal to be -15dB at 160Hz (again that's the -3dB from the beginning of the filter, plus -12dB for the first octave. Since the imagined peak in this scenario is at 115Hz, the filter will not have reduced the signal that much between 80 and 115 Hz, so the overall response would still be up 20dB or so at that narrow peak at 115Hz, and we haven't considered the output of the main loudspeakers at that frequency - they'll be adding, too!


That brings us to the next idea - overlaying responses. What happens right at a crossover, where two filtered responses are added together? Let's take the THX example again. The sub signal is LP filtered with a cutoff frequency of 80Hz, so it's reduced by half power (-3dB) at 80Hz. Similarly, the high-pass filter applied to the mains will have f3 of 80Hz, so it's at half-power as well. When we play the two together (two half-power signals) we get back to full power - magic! If we set it up right, the crossover region is smooth. Does the slope of the roll-off of the mains matter below the cutoff frequency? Well, yes - of course. If the mains are rolling off at -6dB/octave, they are producing too much sound - they're wasting power (loosing efficiency - diverting power from the higher frequency sound we intended for them in the first place) and probably muddying in-room response because their position and EQ haven't been optimized for low-frequency playback. What do we want to do? We want to match the slope of the filter used in the subs, so that everything is symmetrical and adds together to get back to the full signal on both sides of the crossover, just like it did at 80Hz. Sticking with THX, that will require a 4th order filter, right? Well, it should - BUT - the mains are going to have diminished output below 80Hz anyway - so that has to be taken into account. A perfectly optimized cabinet for a sealed main loudspeaker will have a natural roll-off of -12dB/octave. That's what THX specifies (at least originally/historically): a sealed loudspeaker for mains and surrounds (satellites) with a natural f3 of 80Hz. So how do we get from 2nd order natural roll-off to a 4th order high pass response? We add an additional 2nd order HP in the processor - so the signal is diminished -12dB/octave, AND the speaker's natural response is diminishing -12dB/octave for a net response that is -24dB/octave rolling off - magic right? yep.


I hope I didn't railroad you. There's a lot going on, I know. Some of the simplest ideas are best seen in graphs, but out of context the graphs are hard to understand sometimes. Much of what I posted here can be read in a different context here .
 

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Quote:
Originally Posted by HopefulFred  /t/1523218/correct-settings-for-rew-target#post_24501365




"A filter is not a brick wall" as they say... a perfect speaker or sub would put out the same loudness at any frequency given the same power input - right? 1 watt at 20Hz would produce the same sound pressure level as 1 watt at 50Hz or 5000Hz - of course there is no such loudspeaker. Since each speaker or subwoofer only offers a reasonable approximation of that over a small range of frequencies (bandwidth), it's important to use multiple drivers together to get a good (flat) response over a wider bandwidth. What's more, in some cases, the output can increase dramatically in certain ranges undesirably - with lots of distortion or whatever - we would want to filter that out. So we use filters to limit a loudspeaker or a driver within a multi-driver loudspeaker to only reproduce sounds over the range we have decided is desirable. Each driver only gets a portion of the whole signal - the rest of the signal is filtered out.


Filters have a few characteristics - the simple ones (that I understand) are cut-off frequency (or center frequency in some cases, like for EQ) and slope. The cutoff frequency or f3 is the frequency where the filter has cut the power in half - that is to say, reduced the output -3dB compared to the unfiltered signal. The slope is the rate at which that trend of decreasing power continues.


Let's start with a classic THX low-pass filter: it is called low-pass because it allows the low frequency signal elements to pass - removing the high frequency energy. It's 4th order with an f3 of 80Hz. We apply this low-pass filter to a subwoofer signal, usually in an AVR. The signal will be full strength from very low (ignoring any high-pass filter that may be applied, as in your case with the Martycube, for now) and just before 80Hz the signal will begin to diminish. It might be imperceptibly diminished at 65 or 70Hz, but the slope will be obvious by the time the signal is down 3dB at 80Hz. From that point onward (when tracing the plot to the right, toward higher frequencies) the slope will be constant. And every time the frequency doubles, the signal strength diminishes -24dB. At 160Hz, the signal is down a total of 27dB from where it was at 30Hz or 40Hz (that's the original -3, plus an octave's worth of -24dB). And that's -24dB, because it's fourth order.


Let's consider some of the implications of that LP filter. Say you built a horn sub with a really wild resonant peak just outside the intended bandwidth (this is pretty common for horn loaded subs). The response might be pretty flat up to 100 or 110 Hz, but let's say its up about 30dB at 115Hz. That sort of peak will be really obnoxious, so you will want a high order filter on that kind of sub. If the filter were only 2nd order, the signal would be down by very little compared to the classic THX LP filter. With a second order LP, we should expect the signal to be -15dB at 160Hz (again that's the -3dB from the beginning of the filter, plus -12dB for the first octave. Since the imagined peak in this scenario is at 115Hz, the filter will not have reduced the signal that much between 80 and 115 Hz, so the overall response would still be up 20dB or so at that narrow peak at 115Hz, and we haven't considered the output of the main loudspeakers at that frequency - they'll be adding, too!


That brings us to the next idea - overlaying responses. What happens right at a crossover, where two filtered responses are added together? Let's take the THX example again. The sub signal is LP filtered with a cutoff frequency of 80Hz, so it's reduced by half power (-3dB) at 80Hz. Similarly, the high-pass filter applied to the mains will have f3 of 80Hz, so it's at half-power as well. When we play the two together (two half-power signals) we get back to full power - magic! If we set it up right, the crossover region is smooth. Does the slope of the roll-off of the mains matter below the cutoff frequency? Well, yes - of course. If the mains are rolling off at -6dB/octave, they are producing too much sound - they're wasting power (loosing efficiency - diverting power from the higher frequency sound we intended for them in the first place) and probably muddying in-room response because their position and EQ haven't been optimized for low-frequency playback. What do we want to do? We want to match the slope of the filter used in the subs, so that everything is symmetrical and adds together to get back to the full signal on both sides of the crossover, just like it did at 80Hz. Sticking with THX, that will require a 4th order filter, right? Well, it should - BUT - the mains are going to have diminished output below 80Hz anyway - so that has to be taken into account. A perfectly optimized cabinet for a sealed main loudspeaker will have a natural roll-off of -12dB/octave. That's what THX specifies (at least originally/historically): a sealed loudspeaker for mains and surrounds (satellites) with a natural f3 of 80Hz. So how do we get from 2nd order natural roll-off to a 4th order high pass response? We add an additional 2nd order HP in the processor - so the signal is diminished -12dB/octave, AND the speaker's natural response is diminishing -12dB/octave for a net response that is -24dB/octave rolling off - magic right? yep.


I hope I didn't railroad you. There's a lot going on, I know. Some of the simplest ideas are best seen in graphs, but out of context the graphs are hard to understand sometimes. Much of what I posted here can be read in a different context here .

Ok, in the above scenario with the sealed speaker having it's drivers crossed over with a 2nd order filter, which would be -12db which would mean that if those speakers have an 80hz crossover then below 80hz would be down by -12db, right? Is this something that is done in the actual crossover inside the speaker, or a crossover inside the AVR? I mean, are we referring to matching up the drivers in a speaker and then building a proper crossover, or inter grating a sub in an ht? Do the same principals apply with regards to crossover and speaker design?


I am curious when it comes to designing speakers and crossovers, what are the differences between the different order types, filter slopes, and things like Butterworth or LR filter types? How do you know how much overlap to give each set of drivers?
 

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Discussion Starter #19
In my case that change is made in the Barringer amplifier I have that has the DSP. As far as the speakers part of it I do not know if my receiver does that as well when I select certain preset parametric EQ defaults like flat, natural, movies. I need to read a little more what he wrote but does this mean that it is lowered by 12 dB. As in the volume itself is lower by 12 dB then save my main would be?
 

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Let me back up a little for some more conceptual issues creeping in here.


First, remember that an octave is a non-linear and relative idea. An octave is a doubling - not a fixed number of hertz, so it's a distance measured relative to a specific "place" not to be laid down at any position like a ruler. (I know that's a rough analogy) From 80Hz, one octave up is 160Hz (a "distance" of 80Hz), but one octave down is 40Hz (a "distance" of only 40Hz this time).


This goes back to what I started with about perfect speakers (see above). When you look at a frequency response plot for a speaker or sub, you're seeing the SPL generated at any frequency for the same power input. Classically, the speaker designer applies a 1W (or 2.83V, those are different, but the difference is not significant to the concept) signal - generally a sweep, like from REW. The power of the signal at 5Hz is 1W - the woofer makes a sound, the mic records the amplitude, and the sweep moves on - the power of the input signal at 100Hz is still 1W - the woofer makes the sound, the mic records it, the sweep moves on. If everything were perfect, the loudness at every frequency would be the same - the graph output would be a flat line from 0Hz ("DC") all the way to 20KHz and beyond. Of course, it's not flat, but that's the goal. That's the goal because that's the reference standard that everyone in the content production chain assumes - the guy recording the music, the guy mastering the disc, the DAC manufacturer, the speaker designer and so on. If everyone in the chain is working from this same basic assumption, we can hear at home what the musician or director or whoever intended (as nearly as possible) - that's the whole premise of hi-fi, in a nutshell.


Drivers don't behave linearly like this, unfortunately. There will be ranges of their functional bandwidths that approach this ideal, and there will be things that designers do for a variety of reasons that will force the performance away from this ideal. One of those things that is done is to place drivers in boxes. By mounting a woofer in a box (an acoustic suspension), you place an artificial limit on its performance. When the woofer cone is driven away from the back of the box, out into the room, it is attempting to draw a vacuum on the volume behind it; just like a piston in an engine, or a syringe - when the volume of the sealed box changes due to the displacement of the driver, the pressure changes inside the cabinet. Meanwhile, outside the cabinet, atmospheric pressure hasn't changed; the result is that there is a net force of pressure pushing the cone back to its neutral position - where the air pressure inside the cabinet equals the atmospheric pressure outside the cabinet. That force puts a very real limit on the effectiveness of the driver - as though the driver were connected to the back of the box by a spring (an "air-spring" in this case). The designer knows this, and accounts for it in the design. In a smaller cabinet, the movement of the driver represents a very large change in volume of the trapped air, so the force returning the driver to its neutral position is very strong, and the limit to the driver's output is high. In a large cabinet (or mounted in an infinite baffle) the change in volume of trapped air is insignificant (or actually non-existent), so there is no pneumatic return force exerted, and the driver can produce more acoustic output at low frequencies. (How's that for a blast back to your high school chemistry class - do you remember Boyle's Law?)


When the volume of the cabinet is optimized to THX standards, the roll-off below f3 is second order, or 12dB per octave. So, if 1W of power at 120Hz produces an acoustic output of 90dB in this THX satellite speaker, at 80Hz, the same 1W power produces 87dB acoustic output. At 40Hz, 1W produces 75dB. At 20Hz, 1W yields 63dB; at 10Hz 51dB; and at 5Hz 39dB. This is natural "acoustic roll-off" of an optimised sealed cabinet. If the volume is excessively large or small, the roll-off will vary (I'm not familiar enough with the details of what to expect to describe that in any detail).


When bass management is not employed, the full range signal is delivered to the loudspeaker, and output below tuning (80Hz here) is limited by the air-spring of the sealed cabinet. When bass management is engaged (speakers are set to "small," and subwoofer is connected) in a THX processor (or probably most home theater receivers these days), an additional 2nd order filter is overlayed with the natural acoustic roll-off, resulting in the 4th order slope I described last night.


This is one of the same set of concepts used to design passive crossovers in multi-driver loudspeakers (the whole process is much more complicated, obviously - whole books written about it are mere primers). Let's look at a different example - something I've seen discussed with SEOS designs. Compression driver tweeters are small plastic or metal diaphragms (just little coin-sized bits of nothing, really) mounted in a way that the signal causes them to vibrate - really much the same way that a woofer does. But in a CD, the diaphragm is brittle and easily damaged by large waves of energy, especially at low frequency - a high power signal at 500Hz or so would cause most 1" CDs to ripple and crack (like the ground opening to swallow cars in movie earthquakes - or at least that works as an image). Each CD has a different natural response based on its material, diameter, thickness, etc. but within its intended bandwidth, the response is very linear. If a full-range signal were applied to a CD alone, the results would be ugly - the output at low frequencies would be basically non-existent, and if you turned up the gain to get enough output to hear it, you'd fry the thing. In this case you need to remove the low frequency signal content (so that's a high-pass filter) before the signal makes it to the driver. A passive crossover is the common way to do this - the details are not important to this conversation and are outside my understanding at this point, but involves a capacitor and a resistor. The order of the crossover and the cut-off frequency are manipulated to protect the CD from accidental overexcursion when powered at high voltage and keep the acoustic output in the intended bandwidth. (Marty, I'll come back in a few minutes if I can and tell you what I know about passive crossover design (shouldn't take too long
)


If a loudspeaker is designed with three different drive types (A 3-way design) the low frequency driver will need a low-pass filter for its signal (it will also probably get an acoustic high-pass filter based on cabinet design). The midrange driver will need a lowpass filter as well as a high pass filter, and the high frequency driver will need a high pass filter only, because there is generally no signal present to cause trouble in the playback in the ultra-high ranges.
 
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