Solid State & Tube Bi-Amping

I am picking up an Outlaw Audio 950 processor and 770 (7 x 300 in 4 ohms) multichannel amp. I will be using VMPS Ribbon Monitor 40's as mains, VMPS LRC for center and VMPS ribbons & dipoles for surrounds. Right now, I would be bi-wiring the setup (L/R/C channels). I am using my setup for movies AND music.

Would it make that much of a difference if I purchased a stereo tube amp and bi-amped my front speakers, using the tube amp for just the highs and mids?

Being a musician, all I know about tube amps is from my guitar amp gear. I use Marshalls that have that el34 sound and Mesa/Boogies that have the 6L6/5881 sound.

Could I get away with just the SS Outlaw?

To check out specs on my speakers go here (http://www.vmpsaudio.com).

Thanks!

To do real bi-amping, you need to replace the internal passive crossover network in your speakers with an external electronic crossover. The problem with passive bi-amping like you propose is that both amps will be presented with full bandwidth. The tube amp in particular will still be required to reproduce the power-intensive bass frequencies, which could cause it to clip, potentially destroying the tweeter.

Usually, passive bi-amping requires two identical power amps for the low and high end drivers so that there are no power limitations and the gains are identical.

Personally, I think passive bi-amping is not worth the trouble (nor is bi-wiring).

You are absolutely right about the tube amp sound quality however. An alternative perhaps would be to use two stereo tube amps, with one channel of each driving the woofers and the other channel driving the tweeters. At least this would satisfy the equal gain problem. You might still run up against power limitataions however unless the tube amps have over approximately 40 watts, or unless your speakers are very efficient. I actually use a 5 watt tube amp for my tweeters in an active electronic crossover setup, but they are extremely efficient at 106db/watt.

sbcgroup1,

I have read about VMPS speakers for years, but I've never had a chance to listen to any.

I'd love to hear what you and fellow Outlaw's think about them.

Regards.

Jeff Mackwood

id agree that bi-amping is a waste, but disagree about bi-wiring. bi-wiring isn't going to dramatically improve your system, but its nice to have the highs and lows running along their own cables.

The flaw in that logic is that with bi-wiring, the signal travelling down each cable is identical. This is also true of passive bi-amping. The only way to make it so that different signals travel down each speaker cable is with an active system using an external electronic crossover to split the signal into a low frequency range and a high frequency range. These must of course be amplified by two seperate amplifiers. This is active bi-amping.

The only thing you "replace" with bi-wiring is that jumper that resides on the back of your speaker that shorts the two inputs to the passive crossover - the frequency split is done inside the speaker by this crossover network.

With bi-wiring, you are only moving that "jumper" from the back of the speaker system to the output terminals of the power amplifier - there is no electrical difference other than the effective gauge of the speaker cable is doubled. The same electrical result can be had by using a single run of larger gauge speaker cable.

At least with passive bi-amping the amplifier used for the high frequency section does not have to supply current in the low frequency range. An argument could at least be made for the benefit of this, although the audible benefit is questionalbe. This arrangement is also wasteful of resources, especially for the high frequency section.

Question: is it absolutely necessary to have two separate amplifiers? Can you do something like use a 4-channel amp (or 4 channels of a 5 channel amp) and get the same thing as a pair of stereos (or a quadrant of monos)? I've heard before that you want to have separate power supplies (or torroids at least) for this. How much of a difference does that make?

Ok, so there was more than one.

The amplifiers can be two channels of a 4 channel amp, or a 7 channel amp or whatever is most convenient. Any of these combinations will work, however completely seperate monoblock amps would totally avoid any power supply interaction. An amplifier with seperate power transformers per channel will act like monoblocks.

The next best thing would be to use two channels that share a common power transformer (and associated power supply components) for the left and right woofers, and two other channels that share a common power transformer for the left and right tweeters. This avoids interaction between the low frequency amps and the high frequency amps.

The world will not come to an end however if you use, say, an amplifier that has one power transformer for 4 channels. Unless you really crank the amplifier to the edge of clipping, the interaction will likely be below audibility.

There is another benefit to active bi-amping you should be aware of. You can use a much lower powered amplifier for the high frequency drivers since the power demands are nowhere near those of the low frequencies. If you have a 100 watt amplifier for the woofers, you could probably use something in the neighborhood of a 30-50 watt amp for the highs (depending on the crossover frequency - the lower it is, the more power you need in the high frequencies).

This also opens up the door for using something like a class "A" amplifier (or a class "A/B" amp that runs rather more into the "A" class than the "B" class) for the high frequencies. Tube amps are ideal for the highs because the power demands are less, the damping factor requirements are less, and class "A/B" tube amps by their nature run very rich bias - they run essentially class "A" up to a significant amount of their rated power. This makes these amps practically immune to crossover distortion. A single-ended triode tube amp would run pure class "A" and there would be no crossover distortion since there is no crossover, but this type of amp is probably not warranted unless you have extremely sensitive speakers like I do (or else you simply like SET amps).

Whether you go solid state or tubes for the highs, an active bi-amp system will sound worlds better than using speakers with passive crossovers.


[This message has been edited by soundhound (edited May 12, 2004).]

http://www.st-andrews.ac.uk/~www_pa/Scots_Guide/audio/biwire/Page1.html
page three says there can be differences between bi-wiring and standard wiring.

also, feedback is generated by your speakers, so the high frequency feedback is going to go along its cables and low frequency feedback will go along its cables. i know for a fact that is the way my speakers are designed, with the feedback not going back through the crossover, but instead straight out the high and low frequency inputs.

"Tube amps are ideal for the highs"- id disagree with that as well, most tube amps are not "ideal" at all for anything other than door stops. the goal of a system should be accurate reproduction of a source, nothing more and nothing less.

i think you are missing the point here, bi-amping is only worthwhile for non-passive speakers. are yours capable of not utilizing your internal crossovers? if they must use them, THEN YOU ARE WASTING YOUR TIME.

I'd really like to know if you have ever implemented or even heard an active bi-amping system - let alone one that used tubes for the high frequencies? Have you ever even heard a well thought out and implemented tube system at all?

If you could cite some specific expamples and the perceived deficiencies you heard in such a system, I think it would be a bit easier to take what you are saying seriously.

That article is all well and good, however the damping factor of the power amplifier is what deals with back EMF - not the cables. If the cables are large enough and have a resistance that is significantly lower than the internal impedance of the power amplifier (fairly easy to do), their contribution is insignifcant.

In any event, active bi-amping makes the cable issue a moot point. Direct connection to the speaker drivers with no intervening passive components presents the power amplifier with a more benign load. Additionally, it is the inductor that exists in all passive crossover networks, and which is in series with the woofer, that is by far the largest contributor to the deterioration of effective damping factor, and which makes the effects of back EMF from the woofer more serious. The elimination of this inductor allows the damping of the amplifier to do it's work.

distortion is the enemy of audiophiles, tube amps distort. i am sure not every single one does horribly. but, in college we tested several different types of amplifiers (different brands) and anything with a tube in it typically showed more distortion. we found this by using an oscilliscope (sp?), controlling the input wave forms and measuring the output wave forms, then used some computer program to calculate the differences. after our testing, it was clear tube based amps had much higher distortions on average compared to others. i know that some people love tube amps, and they do so because of their sound, but that doesnt mean that they sound accurate, merely "good" to the user. i dont settle for good, i want it the way it was intended to be or as close as i can afford to get it. :-)

Oh brother

also tube amps need maintenance, and more and more maintenance means more and more potential for errors/damage to be done.

So do cars, trucks, boats.....that doesn't stop people from using them.

I believe that the engineers/scientists at B&W have a legitimate purpose for including the bi-wire/bi-amp option (other than sales). Since they design and build all the drivers in their speaker lines, they have detailed analysis of how the voltage/impedance characteristics at the speaker terminals can result in a form of "harmonic crosstalk" between the drivers themselves. Because advanced driver design (and analysis) has been a cornerstone of the B&W mark, their I believe their claims are based on sound science.

Of course, the active crossover that SH promotes would solve this problem also. The B&W crossovers are also designed to flatten the response of each driver, which might be difficult to achieve with an active crossover network. For those who have the analysis tools, EE skill set, and highly advanced knowledge of drivers and their acoustic performance characteristics; "go for it." While I DO EQ, active crossovers are out of my league.

Allan

From the B&W website:

Bi-wiring and bi-amping:
Most B&W speakers are provided with two pairs of speaker terminals; this allows you to either bi-wire or bi-amplify them. The aim of both these techniques is not to simply get the customer to spend more on cables and electronics (although no manufacturer objects to this spin off) but to improve the resolution of the speakers.
A multi-way speaker contains a crossover network that not only divides the incoming signal into different frequency ranges, appropriate to the working range of each drive unit, but also equalizes each driver’s response to be flat (raw driver responses are usually anything but flat).
There are two different basic types of crossover – series and parallel. Series crossovers have each filter section wired in series between the positive and negative input terminals. It is impossible to treat each filter section individually – each interacts with the others – and such crossovers are not suitable for bi-wiring or bi-amping. But by far the most common type is parallel. Here, each driver has its own filter wired between it and the input terminals. If there is only one pair of input terminals, the inputs to all the filters are connected in parallel to that one pair of terminals. If, however, you have more than one pair of terminals, you can completely separate the inputs to each filter. Why on earth would you want to do such a thing?
In the case of bi-wiring, the answer lies in the cable connecting the speaker to the amplifier and the fact that the amplifier is a voltage source but the speaker is a current driven device (force on voice coil = magnet flux density x length of conductor in the magnet gap x current).
Firstly, all cables are a compromise. Some types of construction work better at low frequencies and others at higher frequencies. Providing separate inputs to the speaker allows you to use different cable types, each optimized for the frequency range of use.
Secondly, consider that the cable has an impedance that causes a voltage drop along its length. Now consider the current flowing along the cable. Assume for the argument that the amplifier delivers a perfect voltage waveform to the cable and the cable itself adds no distortion. However, each driver has a non-linear impedance (for example, the inductance of the voice coil alters depending on its position in the magnet gap) that causes the current to be non-linear. This non-linear current through the impedance of the cable causes the voltage drop along the cable to be non-linear and thus the voltage across the speaker terminals is also non-linear, even though it is linear at the amplifier end.
If we were just concerned with one driver, things would not be too bad. But that non-linear voltage at the speaker terminals may contain harmonics within the frequency range of one of the other drivers and that driver will reproduce them, albeit at low level. If, however, you separate the inputs to each driver filter, each driver’s distortion is kept to itself and the total system distortion goes down. We are talking small changes here, but the resolution of some modern drivers is now so good that small improvements like this are readily detectable by keen listeners.
Many people ask us whether the load on the amplifier is different if you bi-wire. It is not. As far as the amplifier is concerned, it matters not one jot whether you parallel the inputs to the filter sections at the speaker end or the amplifier end of the speaker cable.
Bi-amping takes advantage of all this and adds some benefits of its own. Like with cable, you can choose different amplifiers that excel in different frequency bands. You may, for example, be keen on valve (tube) amplifiers. But even the most die-hard of aficionados would be hard pushed to claim that they are any good at keeping good control of the bass. Bi-amping enables you to combine a valve amplifier for mid and high frequencies with the control of a solid-state device at low frequencies.
Unlike bi-wiring, the load to each amplifier is different from that using a single amplifier full range. The voltage demands on each amplifier remain the same (each is still fed a full-range input and gives a full-range output), but the current demands are reduced. This of itself can improve the amplifier’s ability to deliver the signal to the speaker.
Be careful when bi-amping that the gain and polarity of each amplifier are the same, otherwise you will compromise the frequency response of the system.

I am familiar with B&W's argument, and I just don't buy it. I have found in my own experience, and also while conducting experiments while working at Altec Lansing, that given sufficient gauge wiring, it does not matter at all audibly whether the junction of the two (or more) sections of the crossover are joined at the back of the speaker or several feet away at the power amplifier's terminals. I have always used good amplifiers with sufficient damping factors when experimenting: lacking this, and large enough gauge wire, of course alteration of the signal is possible and all bets are off.

Consider the source too. B&W is after all a speaker company, and they are in business to make money. The way they voice their speakers is a very significant part of their market identity. If this were to be eliminated, they would be basically marketing "drivers in a box".

The audio world is very heavily marketing driven. Aligning themselves with popular audiophile ideas is almost required from a marketing perspective. Saying that things like bi-wiring and passive bi-amping are not important would be as suicidal as a CD player manufacturer saying that "16 bits is enough".

B&O has just introduced a very expensive speaker system for consumer use that uses active bi-amplificaiton in addition to other DSP functions which are only possible with such a design. I believe as more speakers incorporate things like self-equalization to compensate for room interaction and various forms of DSP, active technology will become more the norm. Speakers like the Mackie HR 824 simply could not exist were it not for the flexibility afforded by active crossovers which in addition to the basic frequency division function, contour the frequency response with more precision and detail than any passive crossover could ever manage.

By the way, don't sell yourself short! Implementing an active system is not nearly difficult as you seem to think it is - no EE required! The crossover networks and equalizers can be bought off the shelf from firms like Rane and Behringer. Software based RTAs like TrueRTA are extremely powerful. Behringer makes a calibrated microphone suitable for measurement for around $80. With an inexpensive USB audio interface, a laptop, TrueRTA and the microphone, you have an extremely powerful sound analysis system.

[This message has been edited by soundhound (edited May 13, 2004).]

Actually, I was thinking about that. I have never heard a tube power amp in an audio sound system, only in my recording studio and my guitar amps. I notice that if I applied too much voltage or "tube sound" on my microphone preamp tubes, it was distorted and the solid state stuff sounded just the same and could be run at much higher levels throught the gain spectrum w/o the "tube sound"..ie distortion. And as for the guitar amps I have (my Marshall uses EL34's and my Mesa/Boogie uses 6L6/5881's), those have incredible overdrive characteristics. Ultra clean (more precise, but way less organic), is done with solid state stuff. I just don't know how this all applies to stereo systems or HT... I think we may be on the right track of thinking...? 'Cause wailin' on the guitar = distortion & organic cleans!

With musical instrument applications, you are producing a sound rather than reproducing sound. There is a huge difference. The distortion in a tube guitar amplifier is designed in and is a vital characteristic of the sound. There have been solid state guitar amplifiers made, and they have never supplanted tube units.

Solid state amplifiers typically use large amounts of global negative feedback in order to produce low amounts of distortion and acceptably low output impedance. Tube amplifiers use much less - from zero in the case of an SET, to about 20db in the case of an ultra-linear amplifier.

Additionally, the spectra of the distortion matters as much as the static measurements of level. It is a complex topic that cannot be reduced to a simple statement that distortion measurements are everything.

This is the thing that I think is seldom appreciated. Electrons go SO fast, eh?



Bingo!

I've played a bit with active crossovers and EQ; My next system is going to have 18-24 channels of amplification. Big passive components are a pain to fiddle with and not cheap anyway.

Geez! And I'm only running 14 channels of amplification (if I count a bridged stereo amp as a single channel - and each powered sub as one as well) with passive "factory" internal components.

I somehow feel unworthy.

Jeff Mackwood

Having an quality active crossover allows you more felxibility to control how your signal can be splited. You can split signals in various ways to assure precise sound by using the optimal frequency range of each driver in your loudspeaker. It also governs your sound quality sometimes even your dynamic range.

Talking about bi-amping, there are also lots ways and combinations you can use to split your signal by using external Xover. The choice is yours, really, for example.

You can do it in stereo, full-range, 2-Way bi-amping. You split the signal into 2 frequency bands per stereo side ( High and low without mid ). It costs you four amplifier channels; two for the tweeter and the other two for the woofer. This is probably mostly used and very cost effective. Since it is full range, therefore no subwoofer needed. However, you have to use speakers that are really capable of "full-range"

You can do it in stereo, 2-Way bi-amping, plus mono-subwoofer. You split the signal into 2 frequency bands per stereo side ( High and low without mid ). In this mode, the low bass is summed out to the subwoofer. You use 2 amplifier channels to push high and low and another bridged mono amp to push the subwoofer for low bass. It is very similar to the bass management style found in home theater who use satillite speaker.

You can do it in stereo, full-range, 3-Way bi-amping. You split your full-range signal into three frequency bands per stereo side ( High, Mid, and Low )and use 4 amp channels to push the speakers. Two amplifier channels for High and Mid and the other two for low woofer. It sounds great, better than the 2-way bi-amping.

You can do it in stereo, full-range, 3-Way tri-amping. You split your full-range signal into three frequency bands per stereo side ( High, Mid, and low ) and use 6 amp channels to push the speakers. It is very popular these days for large spaces and it greatly reduced intermodulation distortion, but not very cost effective.

You can even go futher to run it in stereo, full-range, 5-Way fifth-amping, plus 1 mono subwoofers. You split your full-range signal into 5 frequency bands per stereo side ( High, High-mid, Mid, Low-mid, and low ) and use 10 amp channels to push the speakers. And then use 1 bridged mono amp to push another additional mono subwoofer. Nice.



[This message has been edited by theendofday (edited May 14, 2004).]

i recognize that there are a FEW good tube amps out there, that MAYBE would meet or exceed a standard for low distortion (if there was such a thing), however IN GENERAL they create that "sweet/good" sound due to the characteristics of their distortion. that is why they make different brands/types/etc. tubes to distort the sound differently. this point is painfully clear to everyone, so there needs to be no more discussion about it.
as far as bi-amping or bi-wiring goes. there is absolutely no need to bi-amp passive speakers, and if given the choice one should bi-wire.

But I can’t help it george, especially because I’m not an audio expert.

I’ve stumbled into an article about music reproduction via electronics devices. It states that live sound of musical instruments cannot be exactly duplicated with electronic devices. Expert gunslingers, is this true? If so, that means the tube or solid-state, neither one can pull off the magic. Then the logical thing to do is to pick a device that one prefers. If that doesn’t suite him or her, then they should go to the concerts or if they can afford it, have the performers come to them.

Correct me if I’m wrong but some gunslingers seem to suggest that solid-state gear CAN pull off the magic of duplicating live sound.

It can get close enough for practical purposes.

It's not an issue of solid state, or tubes, or CD, or SACD, or DVD-A, LP or even Cassette. Any of the current technologies can capture tonal accuracy reasonably well. The issue is much more basic - that of current recording capabilities, and this has little to do with solid state vs tubes.

The reason reproduced music cannot exactly duplicate the experience of live music is the scale and dimension difference between sound coming from two or more small sources (speakers) and many, many real instruments. Bt this I mean the physical size and the way the sound eminates from individual instruments cannot be duplicated acoustically by any means other than the sound coming from the instruments themselves - they way they disperse the sound into the concert all.

It is also because the natural acoustics of a real concert venue and "canned" acoustics (reverb) from a recording have vastly different spatial characteristics coming from either 2 or 5.1 channels verses coming from all around you from an infinite number of directions.

Binaural recordings can come extremely close to what a person would hear if transported to a concert hall. These recordings are made by "dummy" heads with microphones placed in the "ears". The resulting stereo recording is listened to through headphones. You should try to listen to one of these types or recordings because they can get scary-close to the virtual sound of actually being there. You can do a Google search of "binaural recordings" and turn up the limited number of available sources of these recordings. The method never really caught on - it was popular in the early 1950's when stereo was a new medium, but people didn't take to having to listen to their music through headphones exclusively, and binaural recordings sound terrible through speakers.

You can do an interesting experiment to prove to yourself the importance of the dimensional aspect of standard recordings. Play a recording of any number of channels, and listen to it from outside the room through an open door. This removes all the "dimentional" aspect of the sound presentation since it is coming from a single open door - a single source, just as it would be if musicians were playing in the room live. You will find that with many recordings, it actually will sound like the musicians are inside the room playing. If you enter the room, the illusion is destroyed to a degree because you are hearing the soundfield from just two (or 5.1) sources instead of from live musicians with their sound coming from many sources (their instruments).

The organ demo CD that I have circulated on this board is a good example. It is physically impossible with any practical number of channels to exactly duplicate the sound swirling around the listener inside that gigantic stone church with a pipe organ that has ranks of pipes in front and in the back of the listener. With even 5.1 channels this is impossible - with 100 channels it would be closer, but never exactly like being there. I've listened to this recording on every conceivable type of system, and none of them (even mine) come anywhere close to the totality of the soundfield created inside the gigantic chruch where I made the recording.

There are other reasons that a recording will not sound like live musicians, such as the fact that the reccording microphones are never placed where you would normally listen to the performance - they are always placed much closer. In some cases, the microphones are placed within inches of the instruments - you would never listen this close. All this is done to make the recording subjectively sound like a good representation of what you would "expect" to hear. Recordings made "literally" with the recording microphones placed in the concert hall where you would normally sit will sound very dull and lifeless when played on a system in your home.

There is also the issue of the frequency response of the microphones which usually have a reasponse peak in the high end to give the sound more "sparkle" than it would otherwise have (this is more true of microphones used to record popular music however). Again, if you were to not record this way, the recordings would sound very lifeless and dull in your home.

Recording is an art form in it's self, and the object is to give the subjective illusion of reality, rather than literal reality. There is really no such thing as a totally "purist" recording - there is always manipulation somewhere in the chain (either by microphone placement or by electronic processing and/or microphone choice) to transform the sound of the musicians and the recording space into a form that sounds "real" in your home, in a much smaller and acoustically different space than where the music was originally performed.

[This message has been edited by soundhound (edited May 14, 2004).]

I bi-amp my passive crossover speakers (B&W N803's) because:

1) I process the bass frequency signal (under 400 Hz) coming from my pre-amp with a parametric EQ. The room/speaker interaction has been analysed with the same True RTA program that SH has refered to in many posts. The EQ adjusted signal has to be the single most important improvement to the sound quality of my system.

2) The EQ'd bass signal is also limited to a 600 Hz ceiling (low-pass filter) so the amplifier is dedicated to low frequency signals only.

3) The amplifier serving the mid-high frequency speaker driver circuit is fed the full (non-EQ'd) signal, so the internal crossover does all the work. True RTA has shown that I do not need EQ for the mid-high frequency range. Since the midrange and tweeter transducers provide most of the spatial and subtle tonal information, I have chosen not to include a processor in this signal path. I could pass the signal through a high-pass filter, so the amplifier does not need to boost the full bandwidth, but I have concluded that this is not as critical. The actual current demand for the mid-high circuit is relatively small.

I am not against active crossovers , as I have used them in live concert venues. This is the way to go when professional musicians are plugging in to a complex chain of processors and mixing boards in a live environment. Typically live music processing is quite different, and does not have the same spatial information as the studio engineered sound found in recordings.

Many who work in a studio environment (SH is certainly one) prefer to use monitors that emulate those found in live sound venues. I am not sure, but I would guess that movie sound editing would benefit from systems that are based on those found in large theaters.

There are, however, a very significant number of highly regarded professional studios that use the passive crossovers found in the monitors themselves. The studio space and audio systems are all professionally analyzed and balanced by the recording engineers to provide as "neutral" an environment as possible for the type of music (varies for classical, jazz, rock...). This list includes Tellarc, Deutsche Grammophon Gesellschaft, Abbey Road Studios, etc... There is no expense spared in these studios and control rooms.

[This message has been edited by AGAssarsson (edited May 15, 2004).]

>>>Many who work in a studio environment (SH is certainly one) prefer to use monitors that emulate those found in live sound venues. I am not sure, but I would guess that movie sound editing would benefit from systems that are based on those found in large theaters.<<<

I am only involved in music, not sound effects or dialog, and the vast majority of the music I work with is completely orchestra based, usually a minimum of about 50 players. The monitors I use must perform well with music, not sound effects. The monitor speakers used on music scoring stages are not in any way related to "public address" or "live" types - those are a completely different type of speaker intended for a completely different purpose - to project audio great distances at great volume in large spaces, even outdoors.

The final mix of a movie is performed in a movie theater enviornment, and the speakers used in these "dubbing" stages are exactly what you would find in a very good movie theater. These speakers are more like "public address" speakers than not because they must play extremely loud and be rugged enough to not fail, which would cause expensive down time.

The requirement of a final movie mix that includes the sound effects and dialog is to make the film sound right in a typical movie theater - "accuracy" is not even on anybody's radar. The point of a music recording session is to record music as accurately as possible - the reqirements of the dialog and sound effects are not taken into consideration at all. The monitor speakers used for the music mix do not have to fill a movie theater sized room, and are about as far from "public addires" speakers as you can get.

[This message has been edited by soundhound (edited May 15, 2004).]

Odd that you would spend all the extra time and money to bi-amp and change the signal when once it arrives at your speakers IT IS GOING TO GET CROSSED OVER ANYWAY! seems like a waste to me. bi-amping just sums the power inputs, you could achieve that with one, more powerful amp (if passive speaker network). i guess if you like it thats cool for you, but there is no way to get around your passive crossover unless you take it out. so regardless of what you do to the signal prior, it is going to end up the same in the end. i used to do some manufacturing analysis (not really audio design, but how to more efficiently make speakers get them to retailers, etc.) for polk audio and while on that project i had the chance to ask them about why would one bi-amp and then said its pointless for passiver crossover speakers. you could instead of buying two lesser amps for price A, instead buy one better one for price A and be better off. obviously i asked them more questions about it than that and they said more than that, but that sums up what they said. YOU ONLY BI-AMP ACTIVE CROSSOVER NETWORKS (NOT PASSIVE SPEAKERS), UNLESS YOU JUST WANT MORE POWER AND WANT TO UTILIZE EXISTING AMPS.

if you are only getting low frequency signal of 400Hz then why is it necessary to further "limit" it with a 600Hz low pass filter that is doing nothing since the freaking signal is already 400 or below. that is just plain dumb. btw i wouldnt consider 400-600Hz as bass, perhaps you should label it more correctly as low frequency sound, or mid-range. "bass" i.e. what one would expect from a subwoofer is WAY lower than 400Hz...
[This message has been edited by curegeorg (edited May 15, 2004).]

[This message has been edited by curegeorg (edited May 15, 2004).]

I don't quite understand how you can say something like this:



yet still say something like this:



It sounds like you're saying bi-amping doesn't do a thing with the passive crossovers in place yet bi-wiring does? I would venture to guess that what you mean is that bi-amping isn't worth the time over just bi-wiring.

A big thing that might help some of us understand the issue better is to know how the passive crossovers receive the signal when it comes from two different lines be it via bi-amp or bi-wire. How does the crossover handle two input signals instead of one? Does it do anything different?

I see what curgeorg is saying about the passive crossover still being in the way, but it is absolutely imperative to answer the question I just posed before being able to answer that. Maybe it was and I just didn't see it? There is a LOT of information here.

[This message has been edited by JT Clark (edited May 15, 2004).]

yes i mean it is no better than bi-wiring, however it is not as good because it costs a lot more (from that standpoint at least). i am not exactly sure how a signal is dealt with as soon as it enters the speaker, i.e. is the input combined if you connect high and low frequency or what... it seems to me like it has to be since it is only going through one crossover, however i think feedback is fed back through each individual input (high or low). another advantage to bi-wiring is the removal of the cheap p.o.s. jumper used to connect the two posts, granted you could just use a short segment of wire.

Not all jumpers are cheap POS's. Mine are big and strong enough to tear a nasty hole in someone. Not that I have any experience with that.

My car speakers have the ability to be bi-amped. They have two inputs and have a switch for a single input or bi-amp/bi-wiring. I doubt they would spend all that time and money putting it in for nothing. That's what led me to think the passive crossovers might do something different with 2 signals entering vs 1. I haven't tested out how effective it is though. It just doesn't seem very wise to say they don't do anything without actually testing it. Those without a switch may have some sort of automatic detection built in. I don't know.

It is not rocket science to make a 4" long jumper bar to short together two sets of terminals on the back of a speaker. Just how "expensive" does such a bar have to be? Pure gold perhaps? Plated with silver and platinum? Blessed by the Pope? Maybe 4" long by 3" thick, and pure silver - then blessed by the Pope?

That bar does nothing but join the two sections of the crossover network - the section that feeds the woofer, and the section that feeds the tweeter (or midrange and tweeter combined in a 3 way system). That connection could just as easily be done inside the speaker cabinet and routed to a single + and - terminal set on the back of the cabinet. Before the bi-wiring fad reared it's ugly head, this is exactly how all speakers with passive crossovers were configured. The bar just gives you the "option" of feeding the two crossover sections independently from two amplifiers or one amplifier. Either way, the electrical configuration is identical as far as the crossover network is concerned, and the performance will be identical whether you bi-wire or not.




[This message has been edited by soundhound (edited May 15, 2004).]

If someone understands enough about active bi-amping to convert their passive system, I’m sure that person would know better to open up the speakers and rewire it to bypass the stock crossover circuit. They can always reattach them if passive crossover is desired. Only possible compromise (cosmetic) may be drilling two extra holes on the back for additional pair of binding posts if the speaker only came with 1 pair.

So if the you can feed the two crossover sections independently that means that if you had dedicated wiring for each section there would be a benefit. i.e. the inputs are not summed and then split up, but instead that the high terminals connect to the mid/tweeter assembly and the low terminals connect to the woofer assembly? that cannot be what you are saying...

For the sake of simplicity, let's just say each speaker size has it's own input terminal available, ie biwire a 2 way or triwire a 3 way. That way whether you're talking about a 2-way setup or 3-way speaker it wouldn't matter. No partial garbage going on, like figuring out biwiring a 3-way. Or just stick with a 2-way only.

georg, I don't quite understand what you're getting at?

[This message has been edited by JT Clark (edited May 15, 2004).]

I bi-amp my mains by removing the straps which allows you to feed each section of the passive crossover independently. Bi-amping allows me more control over the sub section of the speaker by using the right type of amp with it. If you use an integrated sub amp you can effectivly tune the sub section by being able to adjust volume, phase and bass rolloff independently of the mids and tweet section. The amp I use also allows you to high pass the signal thru to the amp supplying the upper range. My next step will be to remove the woofer passive crossover and control it completely form the sub amp.

As for biwiring I see no advantage at all unless you are stuck with a lot of wire at to small of a gauge and you want to use it up. SH's logic and explanations (numerous) are perfect in explaining this, I just don't understand the reluctance of some to see it.

I am not sure when the Bi-Wire Fad started, but speakers in the early 1960's had straps connecting the Low and High crossover networks; example AR2Ax. And I know for a fact if you remove the strap only the woofer/midrange will prouduce sound if the low terminal is fed from the amp, and only the tweeter will produce sound if the high terminal is fed from the amp.

I'm not sure about the AR speakers, but several other brands of the time had bars that were moved to select different electrical taps that adjusted the volume level of the tweeter verses the woofer. Bi-wiring was not even on the radar screen at the time, nor were boutique interconnects - they were called "patch cords" then.

[This message has been edited by soundhound (edited May 15, 2004).]

that doesnt answer the question of whether or not each terminal feeds its respective frequency range or not.... if high frequency terminals only go to high frequency sound and low to low then there are obvious advantages to both biwiring and biamping. so this cant be true.

Soundhound:
Thank you for your detailed explanation of your work and how your system works to that end. My intent was not to inaccurately define your work, but to affirm that there are other professionals using monitors with passive crossovers. You have my utmost respect.

on the other hand...

[QUOTE] Originally posted by curegeorg:
if you are only getting low frequency signal of 400Hz then why is it necessary to further "limit" it with a 600Hz low pass filter that is doing nothing since the freaking signal is already 400 or below. that is just plain dumb. btw i wouldnt consider 400-600Hz as bass, perhaps you should label it more correctly as low frequency sound, or mid-range. "bass" i.e. what one would expect from a subwoofer is WAY lower than 400Hz...

Yeah... just plain dumb... but my mama told me that middle C is 262 Hz... and that's OK...

But the system configuration I described works very well. Since the passive crossover point for the bass driver circuit in my speakers is 400 Hz, I set the low pass filter on the EQ feeding the bass circuit to 600 Hz so as to provide some headroom for roll-off in the crossover design. The parametric equalizer I use has a low-pass filter, and I use it to limit the signal that the amplifier sees to the range it needs to be supplying this bass driver circuit. The passive crossover inside the monitor will further trim the signal sent to the bass drivers as per it's designed profile. I use an Outlaw 770 amp (200 watts, and >800 damping factor) for the bass circuit. The EQ also has a gain control which allows balancing of the bi-amp circuits.

Since I have used various amplifiers for the mid-high frequency driver circuit, it is useful to bi-amp. These amplifiers have included, monoblock valve (tube) amps, stereo valve amps, and the Outlaw 770 again. This experimentation with different amps has been easy, fun, and educational to me. When my friends come over (during visiting hours), we compare notes on the sounds we hear, and how different amplifiers can make us happy or sad .

As for what mentally challenged persons call bass... or midrange... or high freak'n frequencies... it really depends on what medication we are using at the moment. We often believe that the signal sent to the woofers (aka bass/mid drivers for a two-way speaker) is most easily referred to as "the bass signal" even though it might extend above 2500 Hz. How deluded is that? 2500 Hz is more than three freak'n octaves above middle C. Stupid is as stupid does.


[This message has been edited by AGAssarsson (edited May 16, 2004).]

Why not? If biwiring did anything, I could take a great big amp with some fat cable and put a splitter halfway down the line that split it into two smaller gauges going to each terminal set (hi and low) and be better off than splitting it right out of the amp.

Now as to whether the signal is summed internally and then split to each part of the crossover, you'd have to ask the speaker manufacturer or take them apart yourself. If the terminals can take jumpers externally, then the summation probably doesn't happen. If it does not take jumpers then the speaker either must be bi-amped or the jump is done internally thereby summing the signal and splitting it.

We should disregard monetary aspects, as in for the money. That can be worked out after getting the theory straight.

There is no "summing" of anyting in a speaker/crossover/amplifier/cable interface. Quite simply, there is a low pass crossover network usually consisting of a series inductor shunted on it's output (and across the woofer's input terminals) by a capacitor. This yields a 12db per octave low pass filter for the woofer that eliminates the high frequencies going to that driver.

Then there is another (completely seperate) network consisting of a series capacitor that is shunted on it's output side by an inductor (across the terminals of the HF driver). This yields a 12db per octave high pass filter that eliminates the low frequencies going to the tweeter driver.

There are also sometimes other components to change the frequency and phase contour of the signals being fed to the individual drivers. These fall into the category of components used in order to "voice" the speaker.

The two "ends", the inputs to these two independent crossover sections are tied together - this is really like a "Y" adapter. This constitues splitting the signal, not "summing" it. Huge difference.

That tying together point can be at the back of the speaker via that bar that connects two sets of input terminals to the two crossover network sections inside the speaker. Those two sets of terminals can be fed either by removing the shorting bar and using two seperate amplifiers, or by two seperate wires. Or you can leave the shorting bar in place and feed the whole schbang with one wire and one amplifer.

The woofer is fed a signal that goes basically from DC (let's hope not though, as that means you have a fried amplifier channel!) up to the crossover point - anywhere from around 250Hz to around 3kHz. It is the crossover network that limits the high end extension - the upper limit of frequencies being fed to the woofer. The tweeter crossover network works in a similar manner except that it filters out all frequencies below the crossover point and feeds the remaining signal to the tweeter.

An active crossover network just does this frequency division with active electronics and usually also contains components like equalization and phase compensation (at least in some crossovers) to perform the "voicing" task. In these systems, the passive crossover components inside the speaker are eliminated or bypassed, and the output terminals of the high and low frequency power amplifiers are connected directly to the input terminals of the actual speaker drivers, without having to pass through the original passive crossover network.

[This message has been edited by soundhound (edited May 16, 2004).]

Soundhound:

Excellent explanation of the crossovers. My experiments with the AR speakers and reading of the User's Manual for my NHT ST-4s agree competely with your post. The ST-4 manual states that bi-amping config provides greater dynamic range and allows the subwoofer section to become a powered subwoofer. The sub statement I can buy, I think the dynamic range thing is debateable.
The user manual simply sez that some people beleve that bi-wiring can bring about soun improvements.

You can also buy the raw drivers and an active Xover to bi- or tri- amping. As long as you sent the correct frequency to the appropriate drivers. For example, you can buy 1 horn with 1" or 2" compression driver for high, a 12" woofer for mid and another 15" woofer for low to full-range tri-amping.

sh, that clears it up and is what i thought you were getting at earlier on.
aga, i do not see how so much modification of the signal is ever going to make it more realistic. obviously you and i differ on our opinions of what the purpose of audio gear is, however kudos to doing what it takes to make yourself happy. i know i do.

On the contrary, I would think that we probably agree on the purpose of audio gear. For instance, most purists in search of "realistic sound" use stereo bypass mode for two channel recordings such as CD's, LP's tapes, etc... and we dimwits do too.

Listening environments (room, studio, hall, etc...) have characteristics and limitations that work against the goals of "realistic sound". Most rooms have significant modes of reinforcement and cancellation at specific frequencies. These occur commonly at the low end of the audio spectrum (less than 400 Hz). Without a cure, this very unintended and "unrealistic sound" is often described with words like "boomy, muddy, or sucked-out." The cure for such maladies could include moving speakers, acoustic treatments, bass traps, and other modifications to the physical properties of the space. The goal is to achieve a flat room/speaker frequency response, which gives us that elusive "realistic" sound. In fact, I am not aware of any listening environment that could not benefit from some form of "room correction" as it is often called.

SH has brought up the value of using "Real Time Analysis" (RTA) of the room/system configuration as a diagnostic tool. Programs such as True RTA can resolve response curves to 1/24 of an octave, and are relatively inexpensive to employ. All acoustical engineers use some form of RTA (and other forms of analysis) to carefully assess the room characteristics and evaluate alternate solutions. Generally, you don't need RTA to recognize the problem, you can hear it, but it helps you dial in the solution.

In many cases, it is not practical or not possible to cure the "unrealistic sound" problem without the introduction of sound processors. There are many types of processors, but the most commonly used for room correction are band equalizers, or a parametric equalizers. A good parametric equalizer is the most precise and "tweekable" type of sound processor. It is used commonly in all kinds of spaces for this purpose. These processors are of high sonic quality, and do not introduce unwanted distortion or noise in to the signal path.

So... in review... Rooms have acoustic properties that make for "unrealistic sound". To overcome this, we employ forms of "room correction". Parametric equalizers are commonly used for this purpose. The goal is to achieve a "realistic" room/speaker frequency response. It is not in any way comparable to selecting the stadium or hall mode on a gimmick loaded receiver. You can still select stereo bypass mode on your pre-amp and expect a clean, detailed and balanced sound.

Some pseudo-audio gear heads will argue that the introduction of equipment such as sound processors are, by definition, distorting and personalizing the sound at the expense of the true intent of the recording. They do not understand the fundamental nature of the problem being solved, nor the purpose and effect of the equipment. Don't hire a philosopher to fix your car; and don't let a generalist talk you out of valuable discoveries that are beyond their curiosity or depth.

yeah, i understand what parametric eq is, it happens to be intergrated into my new receiver and was something i liked about it. however i knew about it beforehand, its not like it suprised me. you seem to be implying that you are just correcting room acoustics by manipulating the signal the way that you are, i would counter that there are better ways to do so. you mentioned some in your post. i think that people should be trying to allow the signal, in turn the sound, to get as close as possible to the way it was recorded. if you are going away from that, then you are doing an injustice to yourself, you certainly seem like you think you are going closer to it. it doesnt seem like you are to me. dont waste my time with lengthy explanations of something that is common knowledge for everyone here as a poor attempt to conceal what you already said.

I will be brief. Based on your comments, I would guess that you don't actually know what a PEQ does or the legitimate reasons why people chose to use them. You state that you like your receiver's PEQ feature, but in the next breath you discount others who would use PEQ in their systems. What's up doc?

Why don't you explain to "everyone here" what it is that I am concealing. WMD? Are you really that paranoid? What is your agenda "in here" anyway? Do you think that the intellectual elite are pushing separate audio components?

FYI, at the CES 2004 in NYC, Denon has introduced the AVR-3805, AVP-1x, POA-1x, and DVD-1x. The AVR-3505 is a AV Receiver with an 8 band PEQ. Interestingly, the other products are separates in their "Studio Reference" line. Just imagine how foolish and stupid those people who purchase Denon's higher priced separates will feel when they find out they could get the same stuff in one box; smaller, lighter, and less expensive. And it is also DENON, so it's got to be just as good. Sure glad we know better.

Wouldn't bi-wiring to a speaker with a passive crossover network inside of it work out to be the same or similar to simply running a larger diameter cable between the amp and speaker? If this is true (just asking) then wouldn't the end result be less resistance on the wire? And isn't this a good thing?

I seem to remember reading a Canare, I believe that it was Canare, blurb about their 4S11 wire stating that each of the four strands was 14 awg and that if you were to attach two of those strands each to the positive and negative terminals on the speaker, then you would effectively have an 11 awg speaker wire. Would this not be an advantage to bi-wiring?

Jeff

While bi- or tri- amplification will bring some benefits, I find virtually no benefit to bi- wiring – unless I was making a profit selling cable. If I was given a choice, for a given distance between amp and loudspeaker, of bi- wiring two cables, one for mid/high frequencies and one for low frequencies, giving a round trip resistance of two ohms each, versus one cable that gives a round trip resistance of half an ohm for all frequencies, I would choose the lowest resistance option. Remember, whether bi- wiring or not, the same passive crossover elements are still in the path and still interacting electrically.

i guess it depends on what you consider an advantage to be...

yes it will give you a lower guage aggregate which in turn could lead to less resistance, but that does not mean that the sound will change...

you would be better served to use multiple strands of higher guaged wire, than one low guage piece of wire. i am pretty sure the skin effect is not detectable in wire 18ga or thinner, but it could be 17. either way, 2 18s yielding a 15 combined would be better than one 15ga, assuming all else is equal and there is no short or anything...

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This post has been brought to you by curegeorg, thanks for reading.

Oh....never mind....

[This message has been edited by soundhound (edited September 23, 2004).]

?

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This post has been brought to you by curegeorg, thanks for reading.

Why verbalize until the upper frontal externality displays a reflectance tinge wavelength approaching 475 nm?

“… the better part of valor.”



[This message has been edited by bestbang4thebuck (edited September 27, 2004).]

Don't want to start anything but; this article has some interesting points. It discusses how a tube amp differs from a transistor. I use a combonation of both in my system and each have there good and bad points.

Mesa/Boogie Manual

Yes, they do both have their uses. It's too bad that some people cant't see that the various types of signal devices are all in the audio toolbox, to be used as the application warrants.