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This question has probably been asked before, but I must have missed the answer.
When installing a horn on top of an enclosure, do you position the voice coils in alignment or as the Altec documentation states, align the cone and the horn opening?
Thanks.
For precision;
(Re)wire the HF driver backwards. (Leave the XO in place, everything but the HF wired normally).
Play a test tone the same frequency as the crossover point.
Measure (software/mic, RatShack dB meter, or ears if that's all ya got) a ~meter away (not critical exactly how far). Height halfway between the drivers vertically;
Have a helper slide the HF horn forwards and back, while you monitor the sound level for the lowest reading.
Leave the horn where the lowest reading was observed and rewire the HF back to normal.
(Why not just look for the highest output without the rewire? Because cancellation is more powerful than addition, and therefore more obvious).
Added
Then have your helper fetch a beverage while you cue some tunes.
Very helpful, thanks bfish. I'll try this with my 828 cabs with top mounted 808's
and 421's. And the beverage idea is good too! :)
Actually the beverage comes first!
Maybe thats my problem...:laffingassoff:
You're welome. I should add that the higher the XO frequency, the smaller the sweet spot becomes.Quote:
Originally Posted by rontec
Using an "L" pad the balance in volume can be achieved. I was more concerned with the timing. I have heard different opinions regarding the positioning of the horn for timing reasons.
1, Align voice coil with diaphram.
2, Align horn flare opening with cone.
The Altec documentation I have suggests #2.
I have also heard #1 from a reliable sorce.
Opinions???
Mustang Marvin and Altec Best: Please re-read bfish's response. He describes an excellent method for aligning a two-way loudspeaker. The acoustic center (source of the sound) on a woofer is not the voice-coil or the edge of the frame, and the acoustic center of a HF horn/driver is not the voice-coil or the edge of the horn. bfish's method is described in old Altec literature for theater system set-up . . .
Another way to align a two-way loudspeaker is to measure the time difference using an analyzer. A passive system is then physically aligned (fore/aft position of the source). You can adjust the alignment in a by-amp system using digital delay. Delay the closer acoustic center to the distant acoustic center. These methods require special equipment (expensive), highlighting why bfish has described an excellent method for most of us to use.
An AES white paper that I assisted on offers this;
The closer the acoustic center is to aligned, the more phase damage, until they are perfectly aligned and in-phase. The converse is that the farther out of alignment the less damage occurs. Aligning the mouth of the horn with the front edge of the woofer can sound better than lining up the voice-coils!
Thanks G!
I tried to be simplistic, that's usually pretty easy for a simpleton like me...
As I understand this situation (may be a bit faulty here and there) the "acoustic center" or virtual source of sound can be considered to be at the plane of the voice coil on a high frequency compression driver due to the very light moving mass. On a woofer with many times the moving mass the acoustic center is somewhat behind the plane of the voice coil due to inertia slowing the response time. This helps somewhat on a coaxial driver like a 604, at least partially compensating for the distance between low and high frequency voice coils.
Bfish's method of reversing polarity of one driver is a good one, as it creates a cancellation at certain points in space due to one driver moving out while the other is moving in. At points in space where the distance between acoustic centers of the two drivers are equal the cancellation is most complete and the null is greatest. Assuming vertical alignment of high and low frequency drivers, one can visualize how the locations of greatest cancellation would form a straight line (or beam) whose angle vertically would vary as one driver is moved closer or farther from the listening position while the other is held constant. Find that point where the beam coincides with the listening position and the alignment is optimal.
There is more to this Humpty Dumpty job of putting the audio signal back together, at least when it comes to creating a playback which is most transparent to the source, preserving localization (imaging) information in a stereo recording. Second order passive crossovers require the high and low frequency drivers to be connected in opposite polarity. This preserves even power response of the system through the crossover region, though imaging flies out the window. Audio waveforms are highly transient and asymmetric, and a system with drivers connected in opposite polarity cannot reconstruct such waveforms. First and third order networks permit the drivers to be connected in the same polarity. First order networks can sound terrific though they offer very little protection from low frequencies for the high frequency driver. Third order networks offer great protection, though at the cost of a high parts count and possible sonic effects due to the signal having to fight its way through all those parts. No free lunch!
doesn't the "quote" go something like this?Quote:
Originally Posted by bfish
Quote:
“Any fool can make things complicated, but it requires a genius to make things simple.”
Guilty. :)
For several years we have been measuring loudspeakers and components for design programs like "Ease". These measurements have shown that the acoustic center of a horn is a couple of inches up the throat. The exact distance depends on the horn and throat. In addition, the acoustic center of a constant directivity horn changes slightly with frequency. This is not a real concern when working with one frequency at the crossover point. Your description of the woofer is interesting. Not exactly what we have measured, but interesting. It is true that woofers will lag behind HF drivers due to their mass. The larger the woofer, the more they lag. This also means that smaller woofers have little lag and not really a consideration.Quote:
Originally Posted by Steve Schell
Alignment of a loudspeaker is about aligning the components. Although the listening position is critical for stereo imaging, once the loudspeaker components are aligned, they are aligned at 2, 20 or 200 feet away. Home hi-fi folks are encouraged to align at the listening position.Quote:
Originally Posted by Steve Schell
Where to start here . . . Yes, audio waves can be reconstructed with the components connected with opposite polarity. Please try and review "polarity" versus "phase". These are not the same. Connecting the drivers with the same polarity does not necessarily put them in-phase. In-phase, or aligned, means the signal generated by the two components reaches the listener at the same down point in playback time.Quote:
Originally Posted by Steve Schell
Most Altec loudspeakers with passive crossovers have the HF polarity inverted to put them in-phase. This correction is needed because there is a phase shift introduced by the crossover (all filters cause a phase shift and a passive crossover is a filter). Add this phase shift to non-aligned components and inverting the polarity often puts the system closer to in-phase.
Not sure what power response has to do with phase and alignment. Certainly as more and more power is applied to the components they will begin to react in a non-linear fashion (lagging behind). This problem is not automatically the crossover's fault.
It is a common belief in some circles that crossovers are a bad thing. This concept leads some folks to running their systems with bare minimum high-pass filters on the tweeter and letting the components overlap. Starting from this concept can get you to this statement; ". . . possible sonic effects due to the signal having to fight its way through all those parts. No free lunch!" I suggest that the comb-filtering caused by the overlap is much more damaging then the crossover.
Whitebroncoii I offered my thoughts and opinions with the caveat in the first sentence. I would be happy to discuss these concepts further with you, though hopefully without the overlay of condescension and schooling. I assure you that I fully understand the differences between polarity and phase, and I chose my words carefully.
First, could you explain how high and low frequency drivers connected in opposite polarity (i.e. one moves in and the other moves out on a positive signal) might properly reconstruct an asymmetric waveform (i.e. a snare drum hit) at the listening position? If there is a way to do this I am all ears. I am of the opinion that most audio signals are not simply periodic but rather highly transient.
I also tried to choose my words carefully, not knowing who the audience might be. Can a HF and a LF component reproduce a signal at crossover with one of them reversed in polarity without horrible things happening? Yes, when one component is half a wave length away physically or electronically. Am I going to be able to convince you? Probably not.Quote:
Originally Posted by Steve Schell
Take the A7 for example. When the sources are physically aligned, the 2nd order XO (electronically) offsets the HF 180 degrees/half wave, requiring HF polarity reversal for correction.
Whitebroncoii you seem to be refusing to consider my point, that a highly asymmetric, transient waveform cannot be reconstructed by drivers connected in opposite polarity. I never said that "horrible things" would happen, those are your words, though I do sense that you are slipping out a side door with such vague generalization instead of responding to my question. I contend that fidelity to the signal is degraded by this approach compared to what is possible. I have experimented with this phenomenon for years, with many Altec and earlier Lansing systems using stock second order filters. Connecting the drivers in intended (reverse) polarity yields smooth response through the crossover region, at the cost of muddled imaging and degraded transients. Reversing one of the drivers results in a hole in the response at the crossover frequency, but provides far superior soundstaging and transients, where the system becomes far more invisible, or "immediate" if you will. This is especially true if the system polarity agrees with that seen by the microphone(s) at the time of the recording.
Quote:
Originally Posted by Steve Schell
I think you hear improvement the way some folks hear improvements from expensive speaker cables. Yes, I am refusing to get into a pissing contest with you.
Yes, they can depending on the amount of phase shift imparted by the low-pass and high-pass sections of the crossover and the amount of physical displacement of both drivers. Adjusting one to suit the other is the goal.Quote:
Originally Posted by Steve Schell
Phase, polarity and time-alignment are all separate but interrelated parameters.
A few variables here, the first of which you do not detail: the physical displacement of both drivers in relation to each other...Quote:
I have experimented with this phenomenon for years, with many Altec and earlier Lansing systems using stock second order filters. Connecting the drivers in intended (reverse) polarity yields smooth response through the crossover region, at the cost of muddled imaging and degraded transients. Reversing one of the drivers results in a hole in the response at the crossover frequency, but provides far superior soundstaging and transients, where the system becomes far more invisible, or "immediate" if you will.
Also, the type of filter alignment (Butterworth, Bessel, Linkwitz-Riley, etc.) will have an effect on the response because of the amount of phase shift, the amount of attenuation at the crossover point and of course the aforementioned physical relationship between the two drivers; the response (assuming the drivers are flat at and beyond the crossover frequency) is shaped the the constructive and destructive interference that occurs between the two as a net result of all of the above.
Personally I'm not convinced about the audibility of absolute phase on complex waveforms though I do keep track of it during classical recording. I thought I could discern a difference on woodwinds though that was through sighted (not blind) evaluation which is essentially useless.Quote:
This is especially true if the system polarity agrees with that seen by the microphone(s) at the time of the recording.
BobR
Well I've always assumed that time alignment can be achieved regardless of crossover type (1st,2nd,third order) and regardless of HF, LF wired in same polarity or reversed polarity.
It seems that time alignment is achieved most easily by physically positioning HF driver with respect to LF driver. (A7)
Why not then connect the HF and LF drivers in the same polarity and adjust time alignment by positioning the HF driver. Theoretically then when the signal begins the cones and diaphragms begin moving in the same direction. And even though a 2nd order crossover introduces a delay, can't it be compensated for by physically moving the HF driver? (phase alignment)
Probably missing something here so feel free to enlighten me.:confused:
(I'll have to go to biamp and active next):)
It looks like whitebroncoii has bowed out with an ad hominem, but we shall motor on. Joyspring, I will attempt to address the very valid issues you have raised.
First off, my knowledge of the various filter types and their properties is limited. I have spent the most time with parallel first order passive filters as they sound the best in the ways that matter to me. I realize that no relative placement of the two drivers' acoustic centers will ever be perfect, as the path lengths vary at different locations in the listening field. For any given placement, the phase error will vary with both listening position and frequency. I do try to optimize things for the center seated listener, and go to lengths to align the acoustic centers of the drivers for the best result there. Relative placement of high and low frequency drivers' acoustic centers is best achieved through impulse measurement of each driver, and making physical adjustments until the leading edges of the impulses align when measured at the listening position. I used to do this with the DEQX unit, as its calibration made the adjustments electronically. More recently I have been using passive crossovers and aligning things by measurement with the ARTA program. Adjusting left and right channels for symmetry is aided by a stretched string tied to a button in the center of the listening chair. I used to try and eyeball things until I realized that this often resulted in errors of several inches.
I remain puzzled at the notion that high and low frequency drivers could properly reconstruct a transient when they are connected in opposite polarity... with any crossover configuration. After all, a sharp drum hit would create a steep wavefront that bears no resemblance to a periodic waveform like a sine wave. One driver moves toward the listener, the other moves away from the listener, and the result when observed with a microphone and oscilloscope is frankly chaotic. Monitoring the original event in similar fashion would yield a clean spike and some ringing after, but the recombination of drivers firing in opposite directions bears no resemblance to this in my experience.
Joyspring I cannot always determine the best polarity of a given recording for playback. The more microphones used the more confused things get, and most pop recording made in a studio are fairly hopeless in this regard. I have made a fair number of recordings using a pair of Blumlein configured figure eight ribbon mics though, and with these it is usually easy to pick the polarity that offers the best system imaging and transient response. I find it useful to place a DPDT switch at the input of passive crossovers to reverse system polarity with the flip of a switch.
I may have stepped on some toes by offering my opinion of the limitations of second order filters i.e. most stock Altecs. This was not my intent, but we learn as we go. I revere the engineers such as Harry Kimball and Jim Lansing who chose these filters long ago, and I understand their reasoning. The audio world had not yet evolved to such fussy wussy aspects as proper image localization. Jim Lansing, in his last crossover design for the late 1940s Lansing Sound two way system chose a third order network, and I have to think that he had become aware of the limitations of second order networks with their requisite polarity reversal.
Rontec, your post appeared while I was composing my last post. As I understand things, the changing phase of a second order filter is in opposite directions for high and low frequency sections, and is 90 degrees out of phase at the crossover frequency in each section. Changing the physical placement of the drivers relative to one another will not correct this. If the drivers were connected in the same polarity (i.e. both moved the same direction with the signal) the there would be a net 180 degree out of phase condition at the crossover frequency... perfect cancellation if the acoustic centers were aligned. Connecting the drivers in opposite polarity provides constant power response from the system through the crossover region. Unfortunately this results in a situation where, with transients, one driver sucks while the other blows... so to speak.
If we were dealing with periodic (i.e. perfectly symmetric and repeating) waveforms like sine waves, then physical placement could correct any errors. With the highly asymmetric, transient waveforms of real recorded signals, the drivers must be in the same relative polarity and aligned correctly to stand a chance of reconstructing them properly.
Thank you, mah. This article will require considerable study, but looks to be worth the effort. I sure like the term "transient perfect" and the various square wave responses look a lot like what I have heard, i.e. clean vs. scrambled. If I understand the author correctly, the "2nd Order Transient Perfect" crossover connects the drivers in the same polarity with aligned acoustic centers and fills the response hole at crossover by overlapping the responses somewhat. Nice!
"Perfect" replication of a broadband point-source acoustic impulse by any two non-coincident tranducers is simply not possible. There are just varying degrees of a reasonable facsimile. A percussion impulse originates from a single pinpoint location.
Two bullets on converging paths won't become one when they collide.
WOW!
A lot of interesting thoughts here. Since I have the ability to change polarity as well as distance of the 2 drivers in my set up, I think I will just experiment and choose that which sounds best to my ears. Theory, papers and opinions are great but in the end isn't what sounds best to the listener what we are all trying to achieve?
After I complete my project I will post my conclusions.
Thanks,
MM