When are multiple side surrounds necessary? - AVS Forum
Forum Jump: 
Reply
 
Thread Tools
post #1 of 43 Old 01-29-2013, 09:57 AM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
I haven't been able to find much about this, so I thought I'd ask here. When is it recommended to use multiple side surrounds (i.e. a side surround pair for each row)?

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
Sponsored Links
Advertisement
 
post #2 of 43 Old 01-29-2013, 10:49 AM
AVS Special Member
 
4DHD's Avatar
 
Join Date: Aug 2004
Location: sierra ecuadoriana
Posts: 5,811
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 0 Post(s)
Liked: 62
I would say one pair will do for 2 rows, maybe even 3. But 4 rows or more, then you would want 2 pairs.
4DHD is offline  
post #3 of 43 Old 01-29-2013, 11:07 AM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
I assume that's true even when rows are at different levels such as each row being on a different level riser?

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #4 of 43 Old 01-29-2013, 11:50 AM
AVS Club Gold
 
Dennis Erskine's Avatar
 
Join Date: Jul 1999
Location: Near an airport
Posts: 9,141
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 10 Post(s)
Liked: 46
Anytime you have multiple rows of seats, don't want di/bipole speakers, and have some distance from the wall to the first ear on each side of the room.

Dennis Erskine CFI, CFII, MEI
Architectural Acoustics
Subject Matter Expert
Certified Home Theater Designer
CEDIA Board of Directors
www.erskine-group.com
www.CinemaForte.net
Dennis Erskine is offline  
post #5 of 43 Old 01-29-2013, 12:03 PM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
That certainly makes sense from an intuitive standpoint. Hard to believe it's that simple smile.gif Thanks, Dennis.

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #6 of 43 Old 01-29-2013, 02:39 PM
AVS Special Member
 
HopefulFred's Avatar
 
Join Date: Mar 2007
Location: Atlanta, GA
Posts: 2,758
Mentioned: 1 Post(s)
Tagged: 0 Thread(s)
Quoted: 134 Post(s)
Liked: 194
Quote:
Originally Posted by Dennis Erskine View Post

Anytime you have multiple rows of seats, don't want di/bipole speakers, and have some distance from the wall to the first ear on each side of the room.
Would you make an exception to this statement for any particular speaker? (exceptionally well-controlled directivity and wide dispersion, for instance - or only di/bipoles)

I'm not asking for a particular recommendation, just, is such a design possible or does it exist?
HopefulFred is offline  
post #7 of 43 Old 01-29-2013, 02:54 PM
AVS Club Gold
 
Dennis Erskine's Avatar
 
Join Date: Jul 1999
Location: Near an airport
Posts: 9,141
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 10 Post(s)
Liked: 46
No exceptions and wide dispersion direct radiators (assuming great off axis response) is something you could do in the meanwhile; but, when you then go to the array your speaker positions would change. Understand, to do this correctly aside from the two additional speakers and 2-channels of amplification, you need a very good DSP (aka QSC DSP322, 922 etc) and professional calibration.

Dennis Erskine CFI, CFII, MEI
Architectural Acoustics
Subject Matter Expert
Certified Home Theater Designer
CEDIA Board of Directors
www.erskine-group.com
www.CinemaForte.net
Dennis Erskine is offline  
post #8 of 43 Old 01-29-2013, 09:57 PM
 
cybrsage's Avatar
 
Join Date: May 2007
Location: Harrisburg, PA
Posts: 8,074
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 0 Post(s)
Liked: 147
Due to the shape of my room, I went with two multiple side surrounds. Here is a picture to show why:



When you look at the right side of the picture, you will see a large "bump in" which makes the rear row 3 feet less wide than the front row. That bump in would block the sound from the side speaker on that wall if I did not use two of them. Also, I vastly prefer direct radiating sound vs the dispersed sound caused by bi/dipoles...but in my setup bi/dipoles would have the same problems as direct radiators have with that bump in.


(As a note, my camera has an issue where it angles straight lines if I do not hold the camera perfectly level - so the room looks more cramped than it actually is, etc).
cybrsage is offline  
post #9 of 43 Old 01-30-2013, 07:27 AM
Member
 
AaronN's Avatar
 
Join Date: Jul 2011
Location: Baltimore, MD
Posts: 128
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 1 Post(s)
Liked: 10
cybrsage, thanks so much for posting that. I have a two-foot bump-in on one side of my room and was trying to figure out what to do about the side surrounds. Can I ask how you wired them from the receiver? Did you just split the side surrounds?

__________________________________________________ ______

http://www.avsforum.com/t/1415464/aa...sement-theater
AaronN is offline  
post #10 of 43 Old 01-30-2013, 08:51 AM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
You have to be careful with how you wire multiples like that. You need to present the receiver with a load that it wants to see. For example, if you wire two 8 ohm speakers in parallel, the impedance becomes 4 ohms. OTOH, if you wire them in series, the impedance becomes 16 ohms. Back to what Dennis mentioned above, IIRC from what I've read in Toole's book, you also want to decorrelate the two sets of surrounds. I've also read that this can be accomplished by adding the correct signal delay for each speaker (hence the pricey DSP required).

On another note, I'm not sure why adding a time delay will "decorrelate" the two signals. It would seem that the two signals would still have a strong correlation, only shifted. Perhaps the term is used in a different context here, or maybe I just don't remember my signal processing as well as I would like smile.gif

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #11 of 43 Old 01-30-2013, 09:51 AM
AVS Special Member
 
HopefulFred's Avatar
 
Join Date: Mar 2007
Location: Atlanta, GA
Posts: 2,758
Mentioned: 1 Post(s)
Tagged: 0 Thread(s)
Quoted: 134 Post(s)
Liked: 194
I don't think you're having trouble with your recall, I think the issue is that a simple delay is not the optimal solution. I think it's improved over no signal change, but not what Dennis would do if he had a QSC processor in the signal chain. I feel like I read that a phase shift (like from an all-pass filter?) is a better choice - but having had no signal processing education, I'm less well-informed than you are, probably.
HopefulFred is offline  
post #12 of 43 Old 01-30-2013, 10:38 AM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
Assuming a uniform phase shift, wouldn't that be the same as a time delay (e.g. an 90 degree phase shift of a sine wave is a cosine)? If you are doing some sort of manipulation of the phase across the spectrum, I don't know what that would do.

IIRC in my signal processing studies, decorrelation is also referred to as "whitening." Which is a reference to what you are doing to the frequency content of the signal. When you decorelate it, you pass the signal through a matched filter, and the result is a signal with a uniform frequency spectrum. Like you would see with white noise, hence the term "whitening." However, I don't think white noise from the second row surrounds would get the job done rolleyes.gif

Keep in mind that I'm referring to things I haven't studied in years. So I'm afraid there are little truths spread amongst erroneous information. I'd probably be better off keeping my hands taped up and away from my keyboard wink.gif

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #13 of 43 Old 01-31-2013, 02:23 PM
AVS Special Member
 
HopefulFred's Avatar
 
Join Date: Mar 2007
Location: Atlanta, GA
Posts: 2,758
Mentioned: 1 Post(s)
Tagged: 0 Thread(s)
Quoted: 134 Post(s)
Liked: 194
You've left out the frequency variable: delay=phase shift at only one frequency. 1/T is not constant.

I'm just not sure if that's the full story (about how to decorrelate properly), if that's even the best choice or the right way to go about it.

And I have the same feelings you do about keeping my fingers off the keyboard. I just can't resist sometimes; I feel like people ask a reasonable question and get an answer that gives them a direction to go but no appreciation for the complexity that leads to the recommendation. Even though I can't explain the full reasoning, I can't resist pointing out some of the pertinent principles. What I'd like is to get in on some HAA classes or something, but that stuff's expensive for a hobby.
HopefulFred is offline  
post #14 of 43 Old 01-31-2013, 04:03 PM
Member
 
AaronN's Avatar
 
Join Date: Jul 2011
Location: Baltimore, MD
Posts: 128
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 1 Post(s)
Liked: 10
Wow, that's a lot more complex than I anticipated. I guess I'll stick with 7.1 and just make sure I get a front row seat.

__________________________________________________ ______

http://www.avsforum.com/t/1415464/aa...sement-theater
AaronN is offline  
post #15 of 43 Old 01-31-2013, 06:37 PM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
I've been thinking about this more, probably more than I should, and this is what I've come up with. If you look at the Fourier transform (or really just it's properties), a time delay results in a phase shift. The amount of that phase shift depends on the frequency. That does not affect the correlation, and the correlation coefficient is still 1 between the two signals. Now, what happens if you apply a 180 degree phase shift to ALL the frequencies? Think reversing the polarity on your speakers. You get a negative correlation coefficient of -1 (same signal just opposite polarity). Finally, if you want to perfectly decorrelate these signals, you essentially randomly change the phase of the signal without manipulating the magnitude of the frequencies in it. Ths will give a correlation coefficient near zero.

How do you add all that random phase to your signal? Apparently that's called reverb.

Edit: I didn't get to finish my thought earlier. My guess is that calibrating a setup like this would add a delay to get the signals aligned in time at the LP, and then a little reverb to decorrelate the signals. How much reverb is probably where all the calibration experience comes into play. From what I've read, this is the same process they use to create a stereo signal from a mono signal and to simulate concert hall effects.

Here's an interesting paper that discusses most of this.

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #16 of 43 Old 01-31-2013, 07:22 PM
AVS Special Member
 
HopefulFred's Avatar
 
Join Date: Mar 2007
Location: Atlanta, GA
Posts: 2,758
Mentioned: 1 Post(s)
Tagged: 0 Thread(s)
Quoted: 134 Post(s)
Liked: 194
Right.

If the signals are left totally unchanged, the result is a comb filter. The notches will change based on the path length difference between each of the sources and the listener, such that the change in the comb filter is readily apparent as the listener moves. This, I think (surmise), is the most noticeable sign of highly correlated signals.

If one signal is delayed - a fixed but small delay - I don't think that has done anything to diminish the comb filter, but it has moved the nodes in the interference pattern some fixed distance in space. I suppose if the content is normal and reasonably active/dynamic (not sustained pure-tones or simple harmonics) then one might imagine a delay that allows the signal content to change enough during the applied delay that the interference is rendered psychoacoustically meaningless.

If one signal is phase shifted, what's the result in the interference pattern and comb filter?

This is where my brain breaks.
HopefulFred is offline  
post #17 of 43 Old 01-31-2013, 09:31 PM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
You may be right regarding the comb filtering. However, I think the phenomena that we are trying to replicate is much more complicated. The paper I linked to gives a good example of this. In a large concert hall, we hear a direct sound from the stage that is nearly perfectly correlated. The sound originates from the same spot and traveled the same distance to our ears on either side of our head. However, there is also sound reflecting from other surfaces in the concert hall, and some of those arrive at our ears a little later. The sounds arriving at the left ear have interacted with different stuff as well as traveling a different distance than the sounds arriving at the right ear. So both of these waves have a different "character" (or internal phase structure) than one another, and have a near 0 correlation. We've learned over the centuries to interpret this information and decide we are in a large reverberant space. Meaning if you were blindfolded and dropped in an concert hall, you would be able to identify that space as large and reverberant based on the sound signature alone.

For our small listening rooms (relative to that concert hall) we don't have the benefit of of all that space to allow our sound to interact with stuff as well as be delayed, so we try to mimic that with our discrete sources. The original mixer will create a track for that single side surround by adding the sound effects that he/she likes, and by adding the appropriate amount of reverb to get the spaciousness that they want (I'm inferring that at this point). We want to essentially mix our own discrete channel, so we add a little delay to account for the path difference as well as add some reverb to further decorrelate our sounds. Again, we're trying to reproduce or predict that "character" or internal phase structure that would have occurred in the concert hall.

It may very well boil down to comb filtering, but that paper suggests that decorrelation reduces the perception of comb filtering. It may be semantics, however.

BTW, I'm using the term reverb, but that may not be the right thing to call this randomization of the phase structure. I'm fairly confident that's what we're trying to emulate, though.

Wow. Talk about diarrhea of the hands. I just can't stop myself today smile.gif I agree that some HAA classes would be awesome. Maybe we could get a group discount smile.gif

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #18 of 43 Old 02-01-2013, 08:34 PM
AVS Special Member
 
HopefulFred's Avatar
 
Join Date: Mar 2007
Location: Atlanta, GA
Posts: 2,758
Mentioned: 1 Post(s)
Tagged: 0 Thread(s)
Quoted: 134 Post(s)
Liked: 194
I wasn't able to read the article yesterday, but I got through some of it tonight - honestly, I think a fair amount of it is not directly pertinent, so I skipped through a little. There seems to be two sections that directly relate to this conversation.

First, section III.: Technique for Creating Decorrelated Signals,
Second, section IV. Part A: Elimination of the Perception of Destructive and Constructive Interference

Unfortunately, the description of the techniques for creating decorrelated signals is cursory at best. It gives us a few details: the convolution is accomplished via FIR filters; the filters should not change frequency response (duh!); phase changes should be constructed from combinations of random number sequences.

This paper is, as explained in the introduction, intended to provide access to unpublished ideas that require further work before publishing, so it seems like we shouldn't expect it to explain the fundamentals of the convolution - but I sure wish it did. There don't seem to be any references from this section that could lead to more informative reading.

Section IV, Part A describes the application of the decorrelation to sound reinforcement (the use of multiple loudspeakers in an array to distribute sound over a larger area). Honestly, I was surprised to read this first paragraph and find that it expresses pretty exactly what I was trying to convey yesterday - though my use of terms doesn't match theirs exactly.
Quote:
"Multiple loudspeakers create interference patterns that can be heard especially clearly when the listener is moving in relationship to the loudspeakers." "The composite magnitude spectrum will exhibit spectral peaks and notches that results form the constructive and destructive interference of the acoustic waves. The frequency of these peaks and notches is dependent on the difference in arrival times of the acoustic signals at the measurement position."
The bits later in that section describe the important characteristics of the interactions between the decorrelated signals.
Quote:
"Both coloring and combing can be eliminated when the delayed signals are decorrelated from the leading signal. When the decorrelated signal has random phase changes spaced more closely than critical bands, the resulting composite magnitude spectrum will exhibit spectral peaks and notches which are narrower than a critical band and the smoothed spectral envelope is much more likely to retain its original shape. Combing itself is impossible with decorrelated signals because the decorrelated signal is smeared in time and the temporal periodicity between the original and delayed signal varies with frequency. It is interesting to note that the constructive and destructive interference is still present, but the perceptual effects are eliminated."

So here's what I've come to:
  • Simple filters and delays are not the ideal treatment for an additional side surround loudspeaker signal
  • An optimal solution is a digital FIR filter based on a random number sequence, which adjusts phase and delay based on frequency.

What I'm unclear about:
  • Is there some pre-programmed filter that can be set up based on positions and dimensions and distances in a listening room, such that the calibrator need only know what measurements to make and how to adjust the filter within the device, or is this done in a more iterative or intuitive/organic way by the calibrator, by applying a number of different filters and measuring and listening to the result?


Much of this is new to me, but I've become more-or-less familiar with most of the technical jargon by now, so I think I've got it straight. I'd be happy for anyone's feedback about my reading and conclusions.
HopefulFred is offline  
post #19 of 43 Old 02-03-2013, 02:58 PM
AVS Club Gold
 
Dennis Erskine's Avatar
 
Join Date: Jul 1999
Location: Near an airport
Posts: 9,141
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 10 Post(s)
Liked: 46
Here's another read http://dafx04.na.infn.it/WebProc/Proc/P_280.pdf
http://dafx04.na.infn.it/WebProc/Proc/P_280.pdf
...and the Dolby patent... http://www.google.com/patents/US8015018

Dennis Erskine CFI, CFII, MEI
Architectural Acoustics
Subject Matter Expert
Certified Home Theater Designer
CEDIA Board of Directors
www.erskine-group.com
www.CinemaForte.net
Dennis Erskine is offline  
post #20 of 43 Old 02-04-2013, 12:22 AM
Senior Member
 
JonasHansen's Avatar
 
Join Date: May 2008
Location: Denmark
Posts: 424
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 3 Post(s)
Liked: 11
This is a very interesting topic for me, as I am in the middle of a construction using two pairs of side surrounds. I have DSP/amp for each speaker, but I actually thought it would be "enough" to delay the signal. I guess it is more complicated than that.

My DSP is a BSS London Soundweb (identical to JBL Synthesis SDEC4500) which JBL uses for multiple side surrounds. So unless JBL does things the wrong way, the DSP should be able to handle it right. I will have to study the manual.

EDIT: I can insert white noise, summation and EQ in each output path for each surround speaker. I would guess that would decorrelate the signals, but I still dont know how to vary the noise output in relation to the signal level to make it inaudible. Inputs are appreciated smile.gif

JBL ScreenArray Cinema | 9.4 Surround | 136" Scope
JonasHansen is offline  
post #21 of 43 Old 02-04-2013, 04:55 AM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
I'm not sure adding white noise is the ticket. That would certainly help to decorrelate the signals, but you would change the spectral content. The idea is to keep the same spectral content but to change the phase content.

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #22 of 43 Old 02-04-2013, 06:14 AM
HOME THEATER CONTRACTOR
 
BIGmouthinDC's Avatar
 
Join Date: Jan 2003
Location: Northern VA
Posts: 20,700
Mentioned: 2 Post(s)
Tagged: 1 Thread(s)
Quoted: 225 Post(s)
Liked: 579
Quote:
Originally Posted by Dennis Erskine View Post

No exceptions and wide dispersion direct radiators (assuming great off axis response) is something you could do in the meanwhile; but, when you then go to the array your speaker positions would change. Understand, to do this correctly aside from the two additional speakers and 2-channels of amplification, you need a very good DSP (aka QSC DSP322, 922 etc) and professional calibration.

So what techniques do these units use to avoid the problems of multiple side surrounds? From my quick read of the marketing materials, it is just balancing and delays.
BIGmouthinDC is offline  
post #23 of 43 Old 02-04-2013, 07:21 AM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
Dennis, very informative paper. Thanks for the reference. I haven't made it to the patents yet, but I'm interested to see what's in there.

BIG, hopefully Dennis will elaborate on that, but from looking at the spec sheets for the DSP it has extensive filtering capabilities, one of which is an all-pass filter. Presumably, this filter will allow the phase variations necessary to decorrelate the signals. That really seems to be the key here.

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #24 of 43 Old 02-04-2013, 10:52 AM
Advanced Member
 
Nyal Mellor's Avatar
 
Join Date: Aug 2010
Location: SF Bay Area, California, USA
Posts: 926
Mentioned: 3 Post(s)
Tagged: 0 Thread(s)
Quoted: 29 Post(s)
Liked: 70
Yep an all pass filter will work - keeps the frequency response, changes the phase. I don't have any hard and fast rules on how to set this up but if you have access to an acoustic measurement system then with a bit of trial and error and playing with delays / phase shift you will be able to minimize comb filtering between the two rows of surrounds. Other DSPs I know have all pass include Lake (now Lab Gruppen, $$) and Xilica e.g. XP-2040.

Master of Minions, Acoustic Frontiers. We specialize in the design and creation of high performance listening rooms, home theaters and project studios for discerning audio/video enthusiasts.
Nyal Mellor is offline  
post #25 of 43 Old 02-04-2013, 11:36 AM
AVS Club Gold
 
Dennis Erskine's Avatar
 
Join Date: Jul 1999
Location: Near an airport
Posts: 9,141
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 10 Post(s)
Liked: 46
The QSC will do just dandy ... more to the point, it is the only DSP where, regardless of the number/types of filters to each output, the latency on all channels is exactly the same. Makes life easier.

Dennis Erskine CFI, CFII, MEI
Architectural Acoustics
Subject Matter Expert
Certified Home Theater Designer
CEDIA Board of Directors
www.erskine-group.com
www.CinemaForte.net
Dennis Erskine is offline  
post #26 of 43 Old 02-04-2013, 06:30 PM
AVS Special Member
 
Elill's Avatar
 
Join Date: Dec 2007
Location: Sydney, Australia
Posts: 1,436
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 9 Post(s)
Liked: 19
FYI - the Xilica Neutrino series has a fixed latency of 3ms, regardless of the number of filters (I've been looking at them a little while now and I just confirmed it with them).

I'm not sure about the US, but the pricing differential in Australia is ~ half compared to the QSC (note I currently use the DSP 30 (x2) and DSP 4)

Peter the Greek

Downunder Theatre MkII
Redefining snail pace construction
"what is worth knowing is difficult to learn"

Elill is online now  
post #27 of 43 Old 02-05-2013, 04:56 AM
AVS Club Gold
 
Dennis Erskine's Avatar
 
Join Date: Jul 1999
Location: Near an airport
Posts: 9,141
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 10 Post(s)
Liked: 46
To avoid patent problems, the Xilica went to a fixed latency rather than cross swords with QSC on their approach.

Dennis Erskine CFI, CFII, MEI
Architectural Acoustics
Subject Matter Expert
Certified Home Theater Designer
CEDIA Board of Directors
www.erskine-group.com
www.CinemaForte.net
Dennis Erskine is offline  
post #28 of 43 Old 02-05-2013, 06:50 AM - Thread Starter
AVS Special Member
 
J_P_A's Avatar
 
Join Date: Dec 2008
Location: L.A. - Lower Alabama
Posts: 4,109
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 234 Post(s)
Liked: 253
Quote:
Originally Posted by Nyal Mellor View Post

Yep an all pass filter will work - keeps the frequency response, changes the phase. I don't have any hard and fast rules on how to set this up but if you have access to an acoustic measurement system then with a bit of trial and error and playing with delays / phase shift you will be able to minimize comb filtering between the two rows of surrounds. Other DSPs I know have all pass include Lake (now Lab Gruppen, $$) and Xilica e.g. XP-2040.


Quite helpful! Can this sort of calibration be done with REW, or is more sophisticated analysis equipment required?

Dude, are you made of leprechauns? Cause that was awesome!

The Plains Theater Has Begun
J_P_A is online now  
post #29 of 43 Old 02-05-2013, 07:31 AM
AVS Special Member
 
localhost127's Avatar
 
Join Date: May 2009
Posts: 2,277
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 0 Post(s)
Liked: 24
Quote:
Originally Posted by J_P_A View Post

How do you add all that random phase to your signal? Apparently that's called reverb.

just to be clear, reverb in the form of an FX is merely a decay. it is not the same behavior as reverb in terms of a reverberant sound-field. im sure you're aware, but the terms really do get mixed (no pun) and used interchangeably far too often (probably as a result of amp mfg'rs adding the "reverb knob" to represent decay FX on their products so many decades ago - the word has been dumbed down to mere slang).

and reproducing a recording or generation of a reverberant sound-field in your room does not imply the energy emitted from your speakers is magically diffuse/random-incidence - the FX signal generated is still at the mercy/physics of a small space (focused specular reflections).
localhost127 is offline  
post #30 of 43 Old 02-05-2013, 07:33 AM
AVS Special Member
 
localhost127's Avatar
 
Join Date: May 2009
Posts: 2,277
Mentioned: 0 Post(s)
Tagged: 0 Thread(s)
Quoted: 0 Post(s)
Liked: 24
Quote:
Originally Posted by HopefulFred View Post

Right.

If the signals are left totally unchanged, the result is a comb filter. The notches will change based on the path length difference between each of the sources and the listener, such that the change in the comb filter is readily apparent as the listener moves. This, I think (surmise), is the most noticeable sign of highly correlated signals.

If one signal is delayed - a fixed but small delay - I don't think that has done anything to diminish the comb filter, but it has moved the nodes in the interference pattern some fixed distance in space. I suppose if the content is normal and reasonably active/dynamic (not sustained pure-tones or simple harmonics) then one might imagine a delay that allows the signal content to change enough during the applied delay that the interference is rendered psychoacoustically meaningless.

If one signal is phase shifted, what's the result in the interference pattern and comb filter?

This is where my brain breaks.

what you're referring to is polar lobing due to summation or superposition of two or more direct/indirect signals. comb-filters do not exist in the real world. 3D spatial polar lobing is the physical phenomenon of which results in an interference pattern manifested within the 2D frequency response referred to as "comb-filter".

the polar lobes and polar nulls are what physically exist, and the interference pattern will change based on your location in 3space with respect to the polar lobes/nulls, as well as wavelength and source spacing.

changing phase merely modifies the propagation of the 3d spatial polar lobing and thus the resultant location (in 3space) of the lobes and nulls.
localhost127 is offline  
Reply Dedicated Theater Design & Construction

User Tag List

Thread Tools
Show Printable Version Show Printable Version
Email this Page Email this Page


Forum Jump: 

Posting Rules  
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Trackbacks are Off
Pingbacks are Off
Refbacks are Off