The official Dirac Live thread - Page 16 - AVS Forum | Home Theater Discussions And Reviews
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post #451 of 481 Old 05-23-2020, 12:49 PM
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Originally Posted by flax View Post
Usually we focus on frequency (magnitude) response only... while it's important and defines the tonal balance, there is much more than frequency response, this video explains why:

https://youtu.be/T3nO5KxgRIE
Hmm...some of the statements made here appear to go against other research conclusions, such as those of Floyd Toole. Referring to the statements made starting from 13:00 (disregarding the comments about ported subs for now): For example, Toole showed that impulse response does relate to frequency response. He states they are simply different ways of looking at the same phenomenon. In fact, you just do a Fourier transform to get from one to the other. This means changes in impulse will result in changes in frequency amplitude response, and vise versa. One could put A and B together and realize that other systems that supposedly do not alter impulse response, for example Audyssey, must be by adjusting amplitude. Being that the sources of resonances in your system are going to be drivers, cabinets, and the room itself, any EQ system that flattens frequency amplitude response is going to by extension flatten the impulse. To put it simply, reducing the magnitude of a peak reduces the magnitude of its ringing and the amount of time it takes to reduce to an inaudible level. Think of ringing a bell - the only way you are going to stop the ringing after it starts is by applying energy to make it stop. Otherwise, the only option you have to reduce the ringing is to just hit it with less force. It is because of this that it becomes hard to distinguish what exactly Dirac is doing to set itself apart here. I would like to see some comments that speak to this from those more knowledgeable than myself.

To be clear, I have used both Dirac and Audyssey on my system and the audible differences are obvious. I would just like clarification on the above.

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post #452 of 481 Old 05-23-2020, 01:03 PM
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Only way to find out what's causing these shifts is to do investigative measurements with REW. Unfortunately no one seems to be willing or able to do such measurements.
I will say in my case, the shift seems to be more than 1 dB. It’s all a weird shift in that I don’t think it’s just an amplitude difference, but phase as well. I should also note that I don’t use a center channel, so image precision is way more apparent to me than someone who uses a center channel.

My speakers are 9 feet part. An image vocal that should be dead center, is instead to the right and almost halfway the distance between center and the right speaker. That’s with 2.5xx. With 2.3 it’s centered, but not nearly as focused as with Dirac disabled. Just another reason why I don’t think it’s an amplitude issue.

I sent project files to Dirac. One with version 2.3xxx, the other 2.5xxx. Unfortunately they couldn’t find anything.

Is there a way you can do a correction on all channels except bypass front left and right the way you can with audyssey?

Also I wonder why it only allows you to set the height channels as Dolby enabled and not a ceiling mounted height channel. Dirac is disabled if you do. With Dirac, the atmos soundstage is too high. It would be nice if they let you override the distances and dB differences in speaker config. That would probably let us fix the imaging and soundstage issues ourselves. With Audyssey I would always zero out the differences for the front left and right for best imaging. Same for even the TacT RCS 2.0, that’s how far back I go with room correction lol.
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post #453 of 481 Old 05-23-2020, 01:07 PM
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I have the ability to measure in REW, but I decided to invest my time in re-calibrating with 2.5.3 and using the focused imaging arrangement instead of wide - and guess what, now everything is as it should be. Imaging is spot on again, and the response is back to what I remembered it was pre-2.5.2. Sorry I don't have any quantitative data for you, but I'm happy now and we can get this thread back on track
What did you do different? I only do measurements in focused and still have the image shift. I even use a tape measure to make sure front speakers are equidistant to all walls and to the listening chair and microphone. I did multiple measurements.
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post #454 of 481 Old 05-23-2020, 02:11 PM - Thread Starter
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I will say in my case, the shift seems to be more than 1 dB. It’s all a weird shift in that I don’t think it’s just an amplitude difference, but phase as well. I should also note that I don’t use a center channel, so image precision is way more apparent to me than someone who uses a center channel.

My speakers are 9 feet part. An image vocal that should be dead center, is instead to the right and almost halfway the distance between center and the right speaker. That’s with 2.5xx. With 2.3 it’s centered, but not nearly as focused as with Dirac disabled. Just another reason why I don’t think it’s an amplitude issue.
What does the impulse response look like? Did you do measurements with REW?

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Is there a way you can do a correction on all channels except bypass front left and right the way you can with audyssey?
No.

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Also I wonder why it only allows you to set the height channels as Dolby enabled and not a ceiling mounted height channel.
You need to define speaker type in the AVR first then run DL. What AVR?

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post #455 of 481 Old 05-23-2020, 02:14 PM - Thread Starter
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What did you do different? I only do measurements in focused and still have the image shift. I even use a tape measure to make sure front speakers are equidistant to all walls and to the listening chair and microphone. I did multiple measurements.
Do your L and R speakers form an equilateral triangle with the main listening position?

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post #456 of 481 Old 05-23-2020, 02:55 PM
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Not yet. It is a planned feature but it isn't available now.

Ok, missed that.

Thinking about this a bit more, it seems like the signal processing would get pretty complex.

For consistent summing of all of the bass reproducers, they would need to start with the same bass signal, which would seem to mean that the mains' low freq portion would need to be bass-managed.



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Think of ringing a bell - the only way you are going to stop the ringing after it starts is by applying energy to make it stop.

While you could do that by applying a forcing function that's the inverse of the ringing, normally you just wait for the internal damping to convert the kinetic energy to heat, or speed up the process with some form of additional damping.

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post #457 of 481 Old 05-23-2020, 04:35 PM
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or speed up the process with some form of additional damping.
Which is what I meant by applying energy - you hold it with your hand to stop it.

That's why Dirac's claims are curious to me. I don't think you can alter impulse without changing the FR, and vise versa.

To your first point - do you think Dirac is applying a signal to counter the impulse, using your amp(s) like active servos? I don't think their videos are truly revealing how and why Dirac works. Maybe that is proprietary.

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post #458 of 481 Old 05-23-2020, 05:31 PM
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Which is what I meant by applying energy - you hold it with your hand to stop it.
That's applying damping, not energy.

Maybe you think it's energy if you're squeezing it, but resting any inert object with higher internal damping than the bell against it would do the same.


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To your first point - do you think Dirac is applying a signal to counter the impulse, using your amp(s) like active servos? I don't think their videos are truly revealing how and why Dirac works. Maybe that is proprietary.

I agree, quite vague.

My memory isn't trustworthy, but some years back I believe it was Dirac who said more about the case of the rear wall reflection whose timing would be shared over more than one seat, and that it would indeed generate the inverse of the reflection with the proper timing to cancel it.

Oh and this would be destructive interference, not damping.

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post #459 of 481 Old 05-23-2020, 06:37 PM
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That's why Dirac's claims are curious to me. I don't think you can alter impulse without changing the FR, and vise versa.
This is actually a pretty interesting question. What's often loosely called the "frequency response" is the Fourier transform of the impulse response. One property of the Fourier transform is uniqueness - if you know the "frequency response", you know the impulse response, and vice versa. So, given this uniqueness property of the Fourier transform, how can you change the impulse response without changing the "frequency response?" You can't.

But this misses something. For physical systems, the impulse response is a real-valued function of time. That is, for any value of time t that you pick, the impulse response at that time t is just a real number. But the "frequency response", that is, the Fourier transform of the impulse response, is a complex-valued function of frequency. This means that for each frequency f that you choose, the "frequency response" is a complex number.

This complex number can be interpreted much like an ordered pair of numbers { x0, y0 } in an x-y plane (Cartesian coordinates). You can also think of such an x-y pair in terms of polar coordinates by picturing a line segment drawn from the origin to the point { x0, y0 }. This line segment has a length, or magnitude, and an angle. The magnitude is easily calculated from x0 and y0 using the Pythagorean theorem, and the angle can be calculated using an arctangent formula. In an analogous way, a complex number can be thought of as having a real and imaginary part (like the Cartesian coordinates of an x, y pair), or as a magnitude and angle. So at each frequency, the "frequency response" has a complex value, which can be expressed as a + jb, where a and b are real numbers, and j is the imaginary number sqrt(-1). Now, the frequency response magnitude in dB is 20 * log10 (sqrt(a*a + b*b)) by using the Pythagorean theorem and the definition of dB. But the magnitude is only half the story! There is also the phase to be considered. Since the "frequency response", that is, the Fourier transform of the impulse response, is a complex-valued function, if you alter its phase vs. frequency without changing its magnitude, you have nonetheless changed it, and you have therefore changed the impulse response too.

So when we loosely say, "FR", this could be stated as "20 times the base 10 log of the magnitude of the Fourier transform of the impulse response." But again, the magnitude of a frequency response is only half the story.

How do you alter the phase vs. frequency of a "frequency response" without changing its magnitude? With an all-pass filter. An all-pass filter can be as simple as a single biquad, or can get as complicated as one likes.

I hope I didn't get too crazy with this explanation.
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post #460 of 481 Old 05-23-2020, 06:53 PM
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This is actually a pretty interesting question. What's often loosely called the "frequency response" is the Fourier transform of the impulse response. One property of the Fourier transform is uniqueness - if you know the "frequency response", you know the impulse response, and vice versa. So, given this uniqueness property of the Fourier transform, how can you change the impulse response without changing the "frequency response?" You can't.

But this misses something. For physical systems, the impulse response is a real-valued function of time. That is, for any value of time t that you pick, the impulse response at that time t is just a real number. But the "frequency response", that is, the Fourier transform of the impulse response, is a complex-valued function of frequency. This means that for each frequency f that you choose, the "frequency response" is a complex number.

This complex number can be interpreted much like an ordered pair of numbers { x0, y0 } in an x-y plane (Cartesian coordinates). You can also think of such an x-y pair in terms of polar coordinates by picturing a line segment drawn from the origin to the point { x0, y0 }. This line segment has a length, or magnitude, and an angle. The magnitude is easily calculated from x0 and y0 using the Pythagorean theorem, and the angle can be calculated using an arctangent formula. In an analogous way, a complex number can be thought of as having a real and imaginary part (like the Cartesian coordinates of an x, y pair), or as a magnitude and angle. So at each frequency, the "frequency response" has a complex value, which can be expressed as a + jb, where a and b are real numbers, and j is the imaginary number sqrt(-1). Now, the frequency response magnitude in dB is 20 * log10 (sqrt(a*a + b*b)) by using the Pythagorean theorem and the definition of dB. But the magnitude is only half the story! There is also the phase to be considered. Since the "frequency response", that is, the Fourier transform of the impulse response, is a complex-valued function, if you alter its phase vs. frequency without changing its magnitude, you have nonetheless changed it, and you have therefore changed the impulse response too.

So when we loosely say, "FR", this could be stated as "20 times the base 10 log of the magnitude of the Fourier transform of the impulse response." But again, the magnitude of a frequency response is only half the story.

How do you alter the phase vs. frequency of a "frequency response" without changing its magnitude? With an all-pass filter. An all-pass filter can be as simple as a single biquad, or can get as complicated as one likes.

I hope I didn't get too crazy with this explanation.

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So can your saying yes you can alter the phase without changing the FR " Fourier transform" with a all pass filter
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I hope I didn't get too crazy with this explanation.
Andy, that was bat-**** crazy!
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post #462 of 481 Old 05-23-2020, 07:11 PM
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So can your saying yes you can alter the phase without changing the FR " Fourier transform" with a all pass filter
An all-pass filter has a flat FR in dB, but a funky phase response. They are very easy to create, and very useful. The full description of a system in the frequency domain (which is the Fourier transform of its impulse response) includes both magnitude and phase of response. By applying an all-pass filter to a system, you're altering its phase response vs. frequency without changing its magnitude. This alteration in the phase response will also affect the impulse response of the system, even though its magnitude in dB hasn't been changed.
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post #463 of 481 Old 05-23-2020, 09:11 PM
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So can your saying yes you can alter the phase without changing the FR " Fourier transform" with a all pass filter
Phase is a comparison of timing. Almost all filters affect phase. It's an unfortunate side effect. And that includes the most commonly used filters: high pass and low pass, typically referred to as a "crossover" when those two filters are used together.

An unfortunate side effect of applying these filters to subwoofer signal and speaker signal is that it affects the phase; i.e., some frequencies in the signal end up delayed compared to other frequencies. Phase is comparison of timing. Even though the subwoofer signal and speaker signal should sum smoothly, they sometimes don't due to the filters messing with the timing. So you end up with a notch instead of a smooth splice between the sub and speaker.

Since the cause of the problem was timing, the most common way to fix it is to change the timing: i.e., play with the subwoofer delay until the dip smoothens out. Another way to fix the phase is to use a filter, since almost all filters affect phase. Since we're not trying to change the frequency response, we don't need a high pass or low pass filter but instead a filter that will allow all frequencies to pass intact: an all pass filter. This way, only the phase is affected.

Since the problem was caused by a pair of filters messing with the phase in the crossover region, it's nice to have a solution that uses a filter to fix the phase in the crossover region (instead of a solution that delays the entire subwoofer signal). Make sense?
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Really interesting stuff, even if I'll need to read it again several times to comprehend. Thanks!
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Really interesting stuff, even if I'll need to read it again several times to comprehend. Thanks!
It’ll take me more than several times. I felt like Ralph Kramden on The Honeymooners when he was reading about taxes....😂😂😂

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This is actually a pretty interesting question. What's often loosely called the "frequency response" is the Fourier transform of the impulse response. One property of the Fourier transform is uniqueness - if you know the "frequency response", you know the impulse response, and vice versa. So, given this uniqueness property of the Fourier transform, how can you change the impulse response without changing the "frequency response?" You can't.

But this misses something. For physical systems, the impulse response is a real-valued function of time. That is, for any value of time t that you pick, the impulse response at that time t is just a real number. But the "frequency response", that is, the Fourier transform of the impulse response, is a complex-valued function of frequency. This means that for each frequency f that you choose, the "frequency response" is a complex number.

This complex number can be interpreted much like an ordered pair of numbers { x0, y0 } in an x-y plane (Cartesian coordinates). You can also think of such an x-y pair in terms of polar coordinates by picturing a line segment drawn from the origin to the point { x0, y0 }. This line segment has a length, or magnitude, and an angle. The magnitude is easily calculated from x0 and y0 using the Pythagorean theorem, and the angle can be calculated using an arctangent formula. In an analogous way, a complex number can be thought of as having a real and imaginary part (like the Cartesian coordinates of an x, y pair), or as a magnitude and angle. So at each frequency, the "frequency response" has a complex value, which can be expressed as a + jb, where a and b are real numbers, and j is the imaginary number sqrt(-1). Now, the frequency response magnitude in dB is 20 * log10 (sqrt(a*a + b*b)) by using the Pythagorean theorem and the definition of dB. But the magnitude is only half the story! There is also the phase to be considered. Since the "frequency response", that is, the Fourier transform of the impulse response, is a complex-valued function, if you alter its phase vs. frequency without changing its magnitude, you have nonetheless changed it, and you have therefore changed the impulse response too.

So when we loosely say, "FR", this could be stated as "20 times the base 10 log of the magnitude of the Fourier transform of the impulse response." But again, the magnitude of a frequency response is only half the story.

How do you alter the phase vs. frequency of a "frequency response" without changing its magnitude? With an all-pass filter. An all-pass filter can be as simple as a single biquad, or can get as complicated as one likes.

I hope I didn't get too crazy with this explanation.
As I understand it Dirac takes these extra variables into consideration while a lot of simple EQ system does not. I.e. using "complicated biquads" to optimize phase and therefore IR instead of just looking at the FR and leaving it at that. Would that be a correct statement?

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As I understand it Dirac takes these extra variables into consideration while a lot of simple EQ system does not. I.e. using "complicated biquads" to optimize phase and therefore IR instead of just looking at the FR and leaving it at that. Would that be a correct statement?
"Regular" DL does a mixed phase optimization. It allows a small amount of pre-ringing to improve the impulse response.
DL "Bass Control" adds allpass filtering on top of it to align phase of stereo speaker pairs and sub(s) so their combined response follows the target curve more closely. Furthermore DLBC is able to optimize multiple subwoofers to reduce seat to seat differences.
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@sdurani @andyc56

Thanks

Very Interesting, I am starting to understand the concept that's been explained, but in the video and perhaps he explained it but went over my head is this pre-ringing. In real life we don't have this but we do with speaker to the listener, what is the cause of the pre-ringing? where in the chain is this? cause it seems that Dirac concentrates most on this. Perhaps Andy you were explaining this in your third paragraph but that was one where I lost you
Also how does this compare to Audyssey, does Audyssey work with Impulse as well, in fact would not all EQ software work with impulse to some degree, but Dirac takes it that extra mile
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@sdurani @andyc56

Thanks

Very Interesting, I am starting to understand the concept that's been explained, but in the video and perhaps he explained it but went over my head is this pre-ringing. In real life we don't have this but we do with speaker to the listener, what is the cause of the pre-ringing? where in the chain is this? cause it seems that Dirac concentrates most on this. Perhaps Andy you were explaining this in your third paragraph but that was one where I lost you
Also how does this compare to Audyssey, does Audyssey work with Impulse as well, in fact would not all EQ software work with impulse to some degree, but Dirac takes it that extra mile
The impulse response and the frequency response is the same thing just different representations. One can be transformed into the other using the Fourier transform.
When you transform the impulse response into the frequency response you end up with a) the magnitude response and b) the phase response. A lot of confusion stems from the fact that the term "frequency response" and "magnitude response" is used interchangeably. It is NOT the same thing.

Pre-ringing is caused by a filter that has output at the "left side" of its impulse response peak. We quickly perceive it as a pre-echo if certain thresholds aren't maintained. Linear phase filters introduce pre-ringing for example. Audyssey MultEQ is minimum phase only. There is no pre-ringing with minimum phase filters.

I've added a paper by Mathias Johansson to post 1 which explains room equalization, filter types/implementations and perception in greater detail.
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I probably will never completely understand the underlying audio principles, even though Andy, Sanjay and Markus have done a laudable job in explaining them. However, my expectation is that I really don’t need complete understanding—I only need an advanced product like DLBC to take care of the configuration complexities for me automatically. At least that is what I am hoping for.
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However, my expectation is that I really don’t need complete understanding—I only need an advanced product like DLBC to take care of the configuration complexities for me automatically.
Agreed, no need for complete understanding. But if you can grasp enough of the concept, then it will temper your expectations of what the room correction technology can do. That's why I tried to make my post as non-technical as possible. If your only take-away was that the crossover splice involves getting subwoofer and speaker phase matched (phase is a comparison of timing), then you will understand that that timing relationship can't be the same at different listener locations.

Just understanding that part means you won't be expecting the auto-blending feature to create an optimal splice at listener locations outside the main listening position. Multiple seats, let alone multiple rows of seats, cannot all be the same distance away from any particular speaker. Nothing the technology can do about it. Such is the nature of phase (such is the nature of timing).

BTW, I'm using "you" in this case in the general sense, not specifically Jerry (I know your set-up doesn't have more than one listener location).
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post #472 of 481 Old 05-25-2020, 10:29 AM - Thread Starter
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Just understanding that part means you won't be expecting the auto-blending feature to create an optimal splice at listener locations outside the main listening position.
Going from this...



...to that...



...is an improvement in ALL locations, not only the main listening position.

Having said that there's certainly room for improvement like prioritizing certain locations.
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post #473 of 481 Old 05-25-2020, 01:59 PM
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Quote:
Originally Posted by markus767 View Post
Going from this...



...to that...



...is an improvement in ALL locations, not only the main listening position.

Having said that there's certainly room for improvement like prioritizing certain locations.
Are we supposed to be looking at the black line when comparing these two rather busy measurement graphs?
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post #474 of 481 Old 05-25-2020, 02:18 PM
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Both the blue and black lines show a great improvement in the crossover region.

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post #475 of 481 Old 05-25-2020, 02:36 PM - Thread Starter
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Originally Posted by AustinJerry View Post
Are we supposed to be looking at the black line when comparing these two rather busy measurement graphs?
Black is the average of all other measurement points (blue). It shows main + subs. Target curve flat.

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post #476 of 481 Old 05-25-2020, 03:22 PM
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What are you all's thoughts on Dirac Live 2.0 vs REW correction filters for 2.1 music listening?
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post #477 of 481 Old 05-25-2020, 05:56 PM
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Both the blue and black lines show a great improvement in the crossover region.

I wonder why there's essentially no improvement in the peaks/dips in the mid-20's.

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post #478 of 481 Old 05-25-2020, 07:43 PM
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I wonder why there's essentially no improvement in the peaks/dips in the mid-20's.
Ya I was wondering this to, perhaps this graph is more to show the difference on phase alignment between sub and mains where the big difference is and not so much sub to sub alignment, or it is possibly only one sub in which case very little could be done to improve the mid 20s

I'm wondering also how many seats were covered here and the amount of measurements taken

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post #479 of 481 Old 05-25-2020, 10:54 PM - Thread Starter
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Quote:
Originally Posted by noah katz View Post
I wonder why there's essentially no improvement in the peaks/dips in the mid-20's.
Quote:
Originally Posted by Jimmy2Shoes View Post
Ya I was wondering this to, perhaps this graph is more to show the difference on phase alignment between sub and mains where the big difference is and not so much sub to sub alignment, or it is possibly only one sub in which case very little could be done to improve the mid 20s

I'm wondering also how many seats were covered here and the amount of measurements taken
The graph shows the crossover splice optimization. The sub cluster itself is already optimized. 4 subs were used. The individual sub responses at 21 measurement points looked like this before the optimization:



Source: https://register.gotowebinar.com/rt/8043050009869528846
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post #480 of 481 Old 05-25-2020, 11:48 PM
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Quote:
Originally Posted by noah katz View Post
I wonder why there's essentially no improvement in the peaks/dips in the mid-20's.
Probably because the room is not good enough to correct any better than that. You could optimize and get a flat response in the sweet spot but then the errors in other places would be a lot bigger.

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