Phase lesson (ChrisA? since I am quoting you)

[rebuy]

You can try to line up voice coils for time but I was listening to Andrew Jones talk about this issue

and he said you cannot time align this way because of---The Crossover.

 

He's a speaker designer--I'm not--He knows more than me.

[Travis In Austin]

From an interview of Linkwitz 20 years ago:

 

Dickson: You're most widely known as the developer of the Linkwitz-Riley crossover. Could you explain a few of the characteristics of this crossover?

 

Linkwitz: To answer your question, we need to go back to when I started out exploring the whole speaker issue in the early '70s. Then you could take the grille-clothe off many of the available speakers and see a strange, almost haphazard arrangement of the drivers on the baffle. It really puzzled me and I wondered what was going on. So I asked some of the designers why they were doing this and they said, "Because we've found it sounds better."

 

As I looked further into this issue, I realized that two principal things were not well-understood. First, very little was known at that time about the effects of diffraction from the cabinet edges. Second, and more importantly, very little was understood about how phase-shift with respect to the current passing through the voice-coils of different drivers affected the polar radiation pattern of a speaker. In other words, the interaction between the electrical side of a driver and the acoustical response was not clear at the time. For example, the phase-shift between the current in the tweeter and midrange voice-coils, relative to the placement of these drivers on the baffle, affects the speaker's radiation pattern.

 

Basically, since few drivers are really coaxial, with the difference in physical placement -- that is, if the path lengths between the drivers and the listening point are different, or even if they are the same -- you get a vector addition which is a function of the phase-shift between the different voice-coil currents and the distance between each driver and the listener. So Russ Riley and I began our work, in earnest, to be sure that the drivers were in-phase in the crossover region. This, in essence, is what the Linkwitz-Riley crossover is all about; making sure that you have the same acoustic phase between the midrange/woofer and the tweeter at the crossover.

[Travis In Austin]

The next question and answer in the interview:

 

Dickson: How about the phase relationship outside of the crossover region?

 

Linkwitz: As it turns out, that same phase relationship is maintained at other frequencies as well. This is very much in contrast to the classical Butterworth crossovers that people use in a number of speakers. An inherent property of the Butterworth design, whether these are first-order, third-order, fifth-order, etc., is that the crossovers are always in phase quadrature. In other words, the acoustical signals coming from the midrange and tweeter are phase-shifted by 90^o relative to each other. At its -3dB point, each driver has an amplitude of 0.7, and if you add two 90^o phase-shifted vectors of 0.7, you get unity -- the outputs of the two drivers add to unity on-axis. However, as you move farther away off-axis, one or the other driver will experience more phase-shift as the path-length difference becomes longer, and you'll have either a dip or a peak in the amplitude response off-axis.

 

In any event, the true maximum output of the two drivers will occur someplace off-axis, and this is an audibly bad thing. The peak off-axis response can then reflect from the nearest boundary and combine with the direct sound as added coloration.

 

Now, a first-order crossover can be made phase-perfect at one point in space, but I feel quite strongly that you cannot just look at a speaker's performance at one single point in space. The off-axis response is also very important to a speaker's overall performance in a real room, because the radiation in these other directions will add, through reflected and reverberant interactions, to what you hear. Typically, we don't listen to speakers outdoors or in anechoic chambers.

 

For an ideal Linkwitz-Riley crossover, the amplitude is flat on-axis or at unity, just as it would be for an ideal Butterworth. However, the Butterworth response will have its peak off-axis. In contrast, the amplitude of the L-R crossover will be down in level off-axis, and will never be higher than the on-axis response. The crossover point of a Linkwitz-Riley will also be at the -6dB point, equivalent to an amplitude of 0.5, and only when you add vectors with amplitudes of 0.5 that are in-phase will you get unity. If there is any phase angle between these half-amplitude vectors, their sum will be less than unity.

 

A very important point that people sometimes miss in this discussion is that when we are speaking of a given crossover, we are talking about an acoustic crossover, or what happens acoustically. Now, what I have to do electrically to achieve the correct acoustic response may not look anything at all like a textbook filter design. The actual filter often looks very little like the drawings I may show to explain any given example. This is also true for a Butterworth filter. It is highly unlikely that a textbook electrical Butterworth crossover will produce an acoustic Butterworth response, because the driver's response enters into the picture as well.

 

END OF QUOTE

 

Chris, Mike B and many other have been preaching here for  years that the room, and room treatments are as important as the  speaker, the quotes above explains why that can be the case.  In addition, this quote echos what others have said about the crossover being just one piece of a complex puzzle when it comes to speaker design.

 

He also explains that there is a big difference between the actual acoustic crossover and what is on paper.  Have crossover calculators/programs gotten beyond that, or are they still just getting you in the ball park?

 

At times it seems that it is easier to build a new generation fighter jet than a hi-fidelity loudspeaker. 

[Travis In Austin]

Last quote from the interview, but I thought this was mighty interesting. 

 

Dickson: There is a general misconception in some circles about differential vs absolute phase effects in speakers. Recently, I've heard about some well-meaning but misinformed retailers who arbitrarily reverse the polarity of either the tweeter or midrange hookup wires in all of the speakers they sell that are designed with high-order crossovers, in an attempt to make them "in-phase" -- much to the horror of the original designer. Perhaps you could shed some light on this issue.

 

Linkwitz: If someone were to arbitrarily change the polarity between drivers in a good Linkwitz-Riley crossover, they should get a strong null at the crossover point on-axis. In fact, this is a test I use to see how well I have executed the acoustic crossover. However, making such a change with the idea of somehow making a "phase-coherent" speaker is not correct. It will certainly change the sound, mind you, but is definitely not recommended.

[Coytee]

Chris, to go to your animated sine waves, would it then follow that the closer the signals are in frequency, the more phase is an issue.

 

Meaning, if you have a kickdrum as one of those sine waves and a cymbol as the other sine wave, the frequencies are so far apart that they would be essentially independent of each other.  If however, one was a instrument that (via the crossover) was truncated at 400 Hz and another instrument that began (via the crossover) at 400 Hz, then those two would be potentially more destructive to each other since their playing in/near the same frequency?

 

I forget what it's called, but I recall in elementary school playing with two different tuning forks and you could hear the (second?) harmonic pulsating in/out independent of each fork.  I'm guessing that was a good example of phase addition and cancellation?

[Chris A]

One of the incongruities that I see in certain loudspeaker design circles--especially large, high-priced designs--is the use of 3, 4, and even 5-way designs.  One of the reasons for this I believe is the lack of awareness and attention to the difficulties of designing and integrating high quality crossovers with real drivers. The only good reason for using more ways in loudspeaker designs is almost exclusively due to power handling for commercial systems running at very high SPL--not fidelity.  I find that most of the time, crossovers and driver/horn directivity mismatches are the Achilles heel of these type of designs.

 

Whenever I look at EASE data for a cinema loudspeaker, it's always easy to see the polar disturbances--made visible by amplitude, phase, and driver/horn directivity mismatch irregularities between the paired "ways" of drivers in the crossover bands.  These are very audible and are also difficult, costly and time consuming to get right. 

 

Good crossover design is extremely difficult to do well: it's best to minimize their number.

 

Chris

Edited November 4, 2015 by Chris A

[Bubo]

http://www.rane.com/note160.html

 

Interesting read, well most of it, and for me it had lots of pictures.

 

http://sound.westhost.com/lr-passive.htm

 

Interesting article from Australia DIY supplier, I'm not smart enough to score it.

 

This is what I gleaned.

 

If you need even distribution of sound at all frequencies over a wide and tall area, you are probably talking active crossover, and the Linkwitz Riley is popular for this application. The goal of this crossover, in addition to uniform sound over a broad area is to be perfectly in phase at the crossover point.

 

If you have a favorite chair, the mid drivers at ear level and speakers angled towards your chair, you are in the sweet spot and passive (affordable and reliable) Butterworth crossovers will work just fine. If you move around the room, there are going to be areas that are not the sweet spot, your guests can sit there.

 

The differences between the two approaches is managing the phase alignment of multiple drivers over the desired area. In phase on the same frequency should have an additive effect aka louder and inverse signals on the same frequency should cancel. The effects in between could be observed with two identical drivers and a test tone and something that would allow you variable time delay on one of the drivers. When one is at zero and the other is at 180 there should be silence.

 

The speed with which one driver rolls off and the other turns on would also have it's own set of effects, and there is the issue of not slamming the next driver with full power all at once.

 

With all designs, there are the objectives which may or may not be realistic, then there is what can be done and the trade offs begin. A lot of the trade offs are explored on the bench aka "how does this sound"?

 

My two cents.....

Edited November 4, 2015 by Bubo

[Chris A]

Chris, to go to your animated sine waves, would it then follow that the closer the signals are in frequency, the more phase is an issue...Meaning, if you have a kick drum as one of those sine waves and a cymbal as the other sine wave, the frequencies are so far apart that they would be essentially independent of each other...

 

Think of the sound like stew - not like individual entrees on your plate that the OCD people we know try to keep separated until it hits their stomach.  They all get intermingled in the air and the acoustic space that you're listening to them in.  There isn't a separate channel for each instrument, and each instrument type has characteristic sound that is a complex mixture of sine waves (individual frequencies) that vary in phase and intensity with time with respect to each other.

 

Now there is some forms of music that are extremely simple in terms instrumentation: jazz combos are usually bass, ride cymbal and piano/sax/trumpet.  The spectra of the instruments is minimally overlapping (as I take your question to mean).  Even in those instances, your speakers are playing timbre frequencies from all instruments simultaneously - a stew of musical frequencies that change continuously.  You can see it on the spectrograms within Audacity, etc.  The fraction of the time that there are no other instruments overlapping each other spectrally is extremely small--perhaps a solo instrument or voice playing one "note" at a time for a few moments, but rarely more than that. Hit a drum head with a drum stick - and you have all frequencies at once (a pure impulse).  Percussion and instrument attacks are like that--impulsive and broad band in terms of the combined frequencies.  The human voice is one example of a very wide mixture of frequencies that combine to give each voice its character.

 

Real music is like that - There is almost never separated sine waves at different frequencies with no percussion (i.e.,all frequencies at once).  The notion that our music is composed of separate entities isn't accurate: what you get on the recording is everything mixed together-- from 1 (mono) to 7 (multichannel) locations in the recording space.

 

Relative phase is always an issue--in the loudspeaker crossover regions. Pure or absolute phase isn't really audible to the human hearing system (...this has been demonstrated many times). 

 

I forget what it's called, but I recall in elementary school playing with two different tuning forks and you could hear the (second?) harmonic pulsating in/out independent of each fork.

 

Those are called beat frequencies.

Edited November 4, 2015 by Chris A

[Chris A]

If you have a favorite chair, the mid drivers at ear level and speakers angled towards your chair, you are in the sweet spot and passive (affordable and reliable) Butterworth crossovers will work just fine.

 

Actually, if you listen in an anechoic chamber--and don't move around, then you're okay with Butterworth crossovers if you can get the on-axis levels right. 

 

But anechoic chambers are terrible to listen to music--trust me.  You're sitting there listening to your own heartbeat and breathing, and there are no echos.  It's very unpleasant.

 

Chris

[Bubo]

Chris,

 

Thanks, this brings me to the next question which is how do we adjust for the gear and the room, which is going active.

 

The cheap and easy fix is using something like a Pioneer Elite MACC which EQs and does Phase Adjustments, the various high end Surround Receiver Manufactures all seem to have their own version. McIntosh has their own stand alone unit.

 

So it's running correction over the existing gear and Butterworth Crossover. Unless you are set up for bi-amp.

 

Any thoughts on the effectiveness of just using a surround receiver as the stereo preamp?

 

Some support Bi amping on more than just the Center channel.

 

As to a larger room, and or use with sub woofers, the surround unit does allow for 2.0 just LaScalas, 2.2 LaScalas with subs, and 3.2 LaScalas with subs and perhaps a Heresy in the middle if a blended channel can be derived much like using the Bell between two K-horns. I'm pretty sure I could configure my Pioneer THX Receiver into 3.2 run the MACC and adjust down the volume on the center channel, I would have to play with it.

[Chris A]

Any thoughts on the effectiveness of just using a surround receiver as the stereo preamp?

 

If the receiver has pre-outs for the (5+).1 channels and the sound quality is good, I don't see any reason to not use one.  There also are Pre/Pros out there too that are refurbished for about half their sticker price that work extremely well.  It's an economic decision.

 

The base version of Audyssey that ships with receivers and pre/pros isn't, IMO, good enough to do any customizing or to even see even what it is doing. I recommend another brand of automatic room EQ, or if you have the money, the "pro" version of Audyssey works for the few guys that have sprung for the additional $500+ for that firmware version over the base version.  That seems like a lot of money, though.

 

Some support bi-amping on more than just the Center channel.

 

That would be a good choice if the power amplifiers have good sound quality.  That can save a lot of money over separate EQ units, external amplifiers, and associated cables.

 

Chris

Edited November 4, 2015 by Chris A

[Bubo]

 

Any thoughts on the effectiveness of just using a surround receiver as the stereo preamp?

 

If the receiver has pre-outs for the (5+).1 channels and the sound quality is good, I don't see any reason to not use one.  There also are Pre/Pros out there too that are refurbished for about half their sticker price that work extremely well.  It's an economic decision.

 

The base version of Audyssey that ships with receivers and pre/pros isn't, IMO, good enough to do any customizing or to even see even what it is doing. I recommend another brand of automatic room EQ, or if you have the money, the "pro" version of Audyssey works for the few guys that have sprung for the additional $500+ for that firmware version over the base version.  That seems like a lot of money, though.

 

Some support bi-amping on more than just the Center channel.

 

That would be a good choice if the power amplifiers have good sound quality.  That can save a lot of money over separate EQ units, external amplifiers, and associated cables.

 

Chris

 

 

 

As an experiment, I set up a surround system that literally sits on top of my LaScalas (if I had a bigger room, or house, it would have been 5.2 with Heresys and LaScalas I have the gear) using the Synergy TOTL small B3 as fronts C3 and S3 speakers with 3  excellent Yamaha MX 800s fed by a Pioneer Elite THX Surround Pre Outs with the Center channel Bi Amped. It sounds surprisingly good. In this case the Amps outclass the speakers and probably pull them up a bit.

 

My 1980 Mac Gear has been with me so long that it's hard to connect anything else to the LaScalas, though I did rotate 5 amps one time when the Mac was in the shop, all sounded good, the Yamaha was the best.

 

If I ever get more room a third rack with a circuit switch or patch panel will be set up..........

 

BTW does everyone OEM Audyessy or are there other vendors out there. Emotive uses something different in their new TOTL.

 

 

Thanks,

 

Mark

[pzannucci]

 

If the receiver has pre-outs for the (5+).1 channels and the sound quality is good, I don't see any reason to not use one.  There also are Pre/Pros out there too that are refurbished for about half their sticker price that work extremely well.  It's an economic decision.

 

The base version of Audyssey that ships with receivers and pre/pros isn't, IMO, good enough to do any customizing or to even see even what it is doing. I recommend another brand of automatic room EQ, or if you have the money, the "pro" version of Audyssey works for the few guys that have sprung for the additional $500+ for that firmware version over the base version.  That seems like a lot of money, though.

 

Some support bi-amping on more than just the Center channel.

 

That would be a good choice if the power amplifiers have good sound quality.  That can save a lot of money over separate EQ units, external amplifiers, and associated cables.

 

Chris

 

It seems like a solution to phase and driver overlap can't be done without being able to isolate out the different radiating points for each frequency band.  In other words, in the time domain, each driver has to be driven separately and time/phase corrected to be in alignment.  Additionally we still have to manage the driver overlap and perceived angle that the overlap is most correct (referring to the picture with the ellipse showing the phased centers of the drivers).

In a non-active setup, you can not correct in most horn configurations the phase relationships between say a 24db per octave slope on the crossover and it's time domain modifications.  The higher order filters are good to cut down on overlap and lobing problems but pose other issues.

 

Companies like Vandersteen use multi-way configurations using low order (6db filtering typically) with a 4 or 5 way system to cut down on the time domain issues.  This exacts a price in requiring many drivers/crossover combos and dynamic range due to each driver having to operate without being significantly attenuated outside of it's optimum pass band.  The upside of this type of configuration is a slight baffle slope with a lot of drivers and low order slope crossovers, can achieve close to excellent time / phase relationship at a particular point in space without resorting to multi-amped active setups.  Hard to do with horns and sound paths of 7 feet or more.

 

My opinions of course though it is great to experiment a lot with the multiple configurations and topographies, provided your pocket can handle it.  or maybe not.

[cradeldorf]

Feel free to correct me if I'm wrong but doesn't low frequencys kinda roll towards you like a giant sandstorm in the desert where as high frequencys are like a laser beam shooting at you? Which would tend to upset the time alignment as your distance from the speakers increases?

Edited November 6, 2015 by cradeldorf

[pzannucci]

Feel free to correct me if I'm wrong but doesn't low frequencys kinda roll towards you like a giant sandstorm in the desert where as high frequencys are like a laser beam shooting at you? Which would tend to upset the time alignment as your distance from the speakers increases?

The location of the original sound will still be the same so the alignment in a chamber would be the same.  What the directivity does affect is the way the room interacts with the frequencies causing room modes and reflections either smearing the sound or messing with the frequency response.

 

What bothers me most is horizontal alignment of drivers.  Depending on toe in, the time can change significantly.  You do need to literally sit in one location because the frequency response will many times deviate dramatically depending on the angling of the speakers.  This can be a good thing or bad.  I much prefer vertically aligned drivers.

Edited November 6, 2015 by pzannucci