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).]