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.