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Daisy Chaining amps

 

New member
Username: T_bass

North Platte, NE United States

Post Number: 1
Registered: Aug-16
Hey,
Mycurrent preamp has a single output to connect the amps.
Two of my amps are Carver and one has outputs that allow me to connect to my other Carver. I'm short an output for my sub. Any suggestions? Thanks.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18252
Registered: May-04
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Your post isn't very helpful but it seems you simply need to split the output from your pre amp. That should work if the sub is powered with an active low pass crossover.
 

New member
Username: T_bass

North Platte, NE United States

Post Number: 2
Registered: Aug-16
Thanks.....I have a Polk Sub.I know it has a crossover that is functional if the sub is wired from speaker terminals from the amp and then continuing from the sub to left and right speaker. Will using RCA cables utilize the same crossover?
When I tried it the sub produced minimal sound. My stereo days date back beforbefore the active sub, so forgive me for my lack of knowledge. Any and all help is very much appreciated. Ted
..
 

New member
Username: T_bass

North Platte, NE United States

Post Number: 3
Registered: Aug-16
One more question for my learned audiophiles.
I've purchased a blue tooth reciever with RCA cables. I connected the cables to the CD input on the preamp. I've been bluetoothing my music to the stereo via my telephone. Is this a sound application?
Or should I go about it differently? Thanks. Ted
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18253
Registered: May-04
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" ...I have a Polk Sub.I know it has a crossover that is functional if the sub is wired from speaker terminals from the amp and then continuing from the sub to left and right speaker. Will using RCA cables utilize the same crossover?"

I'm going to assume this is a powered subwoofer (not a passive sub). There should be line level inputs on this sub. You want to run your pre amp output to the line level inputs of the sub. The signal should be split at the pre amp and prior to the cable running to your sub and amplifier. In other words, the splitter goes immediately after the pre amp out and the lines to your sub and your amp run from the splitter.

This sends a full range signal to the sub and the amps. The subwoofer's crossover should have a low pass filter with a control to select the "crossover frequency". Set this control at about 80Hz and the filter will roll off the frequencies above that point. The amplifiers will be receiving a full range signal as if the sub was not in line. The sub should also have a level control which allows you an adjustment between the output of the sub in proportion to the speakers.

If this is not how your sub is configured, then it appears you have a passive sub. Passive subs are meant to pair with the specific speakers they have been sold with as a package. If the output between such a sub and your present speakers isn't a reasonable match, you have no options. That wouldn't be a loss in terms of sound. Passive subs sold with inexpensive systems are junk. They are not true subwoofers. In that case, you need to buy a real subwoofer.




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Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18254
Registered: May-04
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"One more question for my learned audiophiles.
I've purchased a blue tooth reciever with RCA cables. I connected the cables to the CD input on the preamp. I've been bluetoothing my music to the stereo via my telephone. Is this a sound application?
Or should I go about it differently?"


I prefer hardwired cables all the way around. If you've been running this set up for awhile, you are using low bit rate MP3 files. If that's your source material, you wouldn't detect any difference between your phone based system and hardwired cabling.


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Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18256
Registered: May-04
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One last thing, don't worry about sending both left and right channels to the sub. If you can, that's fine. However, in 95% of the recordings you'll use, the bass frequencies have been mixed to a mono signal. Whatever appears in the left channel would also be in the right channel.

If you have a powered sub, most often using the right channel input will place the signal into a mono mode.


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Gold Member
Username: Magfan

USA

Post Number: 3366
Registered: Oct-07
My parasound STEREO power amps have a single ended (RCA) output per channel. So it is easy to daisy chain 'em.

For my application, I ran either left OR right channel to the amp, and looped it over to the other channel. EAsy.
And will less cable bulk / shorter runs. The I/O loop is only 18" and could be done with a somewhat shorter cable if needed.

I COULD run a sub per side off of speaker outputs, but use the preamp sub out. The disadvantage of running the sub off the amp speaker outputs is that I'd have to run the main speakers full-range which simply does not sound as good as restricting bass to the mains and sending it ALL to the sub.
 

New member
Username: T_bass

North Platte, NE United States

Post Number: 4
Registered: Aug-16
JAN & LEO,
Thanks for your valuable input! It is much appreciated!
My sub is a active Polk 100W subwoofer.
So.....If I split the signal at the preamp I assume that I will use one of my Y cables?
There wowon't be any signal degradation?
Well...I don't know if anyone has the desire or the inclination to tackle this one.
My question about the sub was due in part to the fact that I am boomer and a two channel,pre sub guy.
It's new to me.
All of my equipment ranges from about 1979 to 1995ish.
Ive been having a hell of a time getting everything hooked back up right since we moved.
Maybe someone can help.
I have a preamp...a Carver M400 amp to drive Bose 901 series 4 speakers and the bose 901 equalizer.
The other part of the mix is a Carver tfm-6cb that drives two pair of 201's...a Soundcraftsmen eq...and a phase linear 1000 autocorrelator.
I assume that the amp,bose eq and 901 part of the system will be stand alone and that I won't be able to route the signal through the oth eq and the PL1000?
Anyway thats how ive been setting it up.
ANY HELP WILL BE APPRECIATED!
 

New member
Username: T_bass

North Platte, NE United States

Post Number: 5
Registered: Aug-16
Jan Vigne,
You are apparently a very learned individual in this field.I do appreciate all the help that you give me.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18257
Registered: May-04
.

"If I split the signal at the preamp I assume that I will use one of my Y cables?
There wowon't be any signal degradation?"




Not if you use quality connectors and cables. Though the question itself raises other issues with your system.

Not to be offensive but you have what could easily be termed a fluster cluck of a system. Too many points of connection where every connection and cable has the potential for sonic degradation. Far too many components whose sole intention is to … uh, … , "manipulate" the signal in some fashion. Your system is about as far away from the ideal of a "straight wire with gain" as is practical to fit into one shelving unit.

Most of your components were … uh, … "designed" with the selling point being the source is deficient in quality. Most often, the remedy to a weak source was to return the system to something that resembled the the mixing desk used to create the source. The greater the number of components and the higher the count on buttons, knobs and sliders, the more likely you were to return the source to its full glory.

The majority of your components were introduced well in advance of the mainstream concept of room treatments. While not ignoring the multitude of potential errors introduced by a poorly mastered source, what we have come to realize in the last few decades is that most source material is actually much better than we had been told by those folks selling buttons, knobs and sliders. A well constructed system of moderate quality lacking all the bells and whistles of yesteryear can retrieve rather substantial amounts of information from even the most ancient source.

To that end, I will only say that every time you add a connector and cable and every time you run a signal through another electronic circuit, you have the potential for sonic degradation. Your description of your system leads me to think the addition of a splitter would go totally unnoticed in any case.

The splitter for the sub would go immediately following the output of the pre amp. You can run one or two cables. It is not essential to have two channels into the sub, most active subs have a provision for a "mono" input which is typically just plugging into the right channel input. Since the vast majority of recordings have been mixed to a mono bass, this is typically sufficient. The advantage to running two cables to the sub is to increase the Voltage input to the sub's amplifier. You will gain +6dB by adding the second channel which would result in a somewhat lower level set for the gain control on the sub. There is no real sonic advantage to this hook up if you can match levels with only one channel's input.

So, one leg of the splitter goes to the sub and the other to the Bose eq. Set your crossover filter on the sub to approximately 60-80 Hz. Adjust the gain accordingly. Placement of the sub within the room is a matter of acoustics. Corner placement is typically less than ideal as is center of the wall. Your sub probably has some instructions for placement and experimentation is the key.

To some extent, a problem occurs with the Bose eq and the sub. The splitter will feed a full range signal to both components. The crossover filter in the sub will roll out the upper frequencies at a fixed rate while the eq will boost the bass frequencies being fed to the 901's. Where the Bose system is prone to large amounts of comb filtering and acoustic phase issues, the addition of the sub has the potential for adding to the problems. How much or how little this issue bothers you is not something I can predict. A substantial amount of the issue will exist in the placement of the speakers relative to the sub and your listening position relative to both. The room is the most important contributor to what you hear.

Why did you add the sub? Is this system being used for music only?




"The other part of the mix is a Carver tfm-6cb that drives two pair of 201's...a Soundcraftsmen eq...and a phase linear 1000 autocorrelator.
I assume that the amp,bose eq and 901 part of the system will be stand alone and that I won't be able to route the signal through the oth eq and the PL1000?"




It's not impossible to combine the components. I would though ask, why? The addition of the Carver, Soundcraftsmen, etc complicates the system by exponential margins. How long has it been since you had these components up and running?

If the system has been in storage, there is the high potential you will run into problems with most of these components. Pulling a "vintage" component out of storage and slamming it with a full 120VAC is the best way to ensure the component will not sustain the blast of current. That assumes the component might be in OK condition to begin with. Capacitors in 1970's gear tend to dry out and leak. Since a few of those caps would have been used to filter out unwanted DC Voltages from the system, you can destroy your entire system by simply returning old gear to a system.

Switches, knobs and sliders plus all the tin plated RCA type ins and outs tend to oxidize over time and become noisey to the point they will not make contact at all positions. Oxidized connections add more noise and distortion to the signal. They also have the potential to destroy components if you begin to troubleshoot a problematic channel.

Of course, the previous issues of multiple connectors and numerous circuits also applies to this equipment menagerie. The more you add the more you increase the noise, distortion, electrical and acoustic phase shifts plus the deeper you make the comb filtering effects. I cannot predict what you will like or find acceptable as far as music quality goes but this would surely be a grab bag of effects if all of this gear were put into use.

Adding his part of the system would mean adding more splitters and cables. Eventually, the jumble of cables alone becomes nearly unmanageable when problems arise. If you are intent on having this system set up with everything included, you first need to check the gear for proper operation. Do not simply plug this gear into a wall outlet. It needs to be brought up slowly to operating Voltage while being monitored on an oscilloscope. This is best done by a tech on a bench.

If you proceed with the addition of each component, I would advise you to do so individually, one at a time after you've listened to the system in a lesser state. Add only those parts which really must be added and do so in a manner in which you can begin to tell just what you are adding to the music and what you are hearing that is less than ideal. If you simply hook up everything at once, you will, in all likelihood, have a mess that has problems you cannot troubleshoot without disassembling the entire system and starting fresh with one component added individually.

Personally, I would set up the simplest system possible and just listen for awhile. Adding more gear does not mean you have more or better music quality.




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New member
Username: T_bass

North Platte, NE United States

Post Number: 6
Registered: Aug-16
Jan....I only listen to music.
So this is what I got from your advise.
Get rid of the sub...the S.C. equalizer...and the PL 1000 correlator.
Just use the Bose eq the 901"s and the M400.....and the smaller amp with the 201's.

Nothing else ...right?

The only component that hasn't been on the bench is the SC equal.....but you're right. The cables are a fuster cluck and I'm sick of screwing with it
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18258
Registered: May-04
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The sub is optional. No matter what Bose claimed, the 901's cannot reproduce deep bass. The eq simply applies boost which, according to the Dopler Effect all humans perceive, gives the impression of bass deeper than what actually exists.

I would say listen without and then add the sub. IMO balancing the eq with the sub will be dicey. However, it's still your system and you get to pick what you prefer. If the sub proves intrusive, remove it.

Otherwise, your description is fair and a good start.


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Gold Member
Username: Magfan

USA

Post Number: 3367
Registered: Oct-07
Wrong Term?
Doesn't the Doppler Effect apply ONLY if the listener and source are in RELATIVE MOTION? Approaching or receding from one another.

Other than that, Agreed. 901s have a limited bass response. And that equalizer? Just makes them VERY power hungry. Adding 10db to the low end means you need 10X the power, at those same frequencies. I can't imagine a HT receiver being able to sink that much power, except at the VERY lowest levels.
 

New member
Username: T_bass

North Platte, NE United States

Post Number: 7
Registered: Aug-16
What I am using to power the 901s is a Carver M400. One is currently in the shop. When all is said and done each speaker will have 500 watts monoblocked to it.
Will this suffice?
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18259
Registered: May-04
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"Doesn't the Doppler Effect apply ONLY if the listener and source are in RELATIVE MOTION? Approaching or receding from one another."



No more than the simple idea of Newton seeing an apple drop defines the concept of gravity.

The Qts of a speaker system (or bass alignment in a vented system) relies on the Doppler Effect for its perception of "speed", "accuracy" or "jump factor" on the showroom floor.

One of the best examples I know to demonstrate the effectiveness of the Doppler Effect remains the original LS3/5a. The system goes against the grain of most BBC designs which aimed for higher accuracy by giving in to the fact the shoe box sized enclosure of the 3/5a simply cannot support "deep bass".

The 3/5a was not new in its use of this design element. The knowledge of a "doubling" effect was well known to speaker designers by the time the 3/5a was being designed. This is merely the knowledge of how most acoustic music operates - a stronger second harmonic than the actual fundamental frequency - and the use of a louder upper frequency component which sold the idea of bass in many early (mostly vented) speaker systems.



By providing a system Q which was somewhat higher than the typical BBC design, the small studio monitor gave the cognitive impression of deeper bass than it could actually produce. Had this same bump been designed into a larger enclosure, the bass would have appeared to be rather bloated and (slightly) less controlled.

The concept of a "West Coast" loudspeaker sound is based primarily on the Doppler Effects' cognitive perception by the listener. Lumpy and bumpy sells as "extended" in most showrooms.

While the same boost/bump might have seemed out of place in a larger BBC designed enclosure, it worked to the advantage of the 3/5a which existed initially for use in mobile production systems (think the inside of a medium sized British van in the mid-1970's). When combined with the slight dip in response just above the 120Hz range which has become somewhat famous as the "sound of" a 3/5a, the tiny monitor continues to amaze listeners in all of its variations.



Looking at the frequency response graphs of most "standmounted" loudspeakers of today you will see the same fudge applied to most popular systems. It has become an almost universally applied standard for most small enclosure systems sold to the audiophile to include this little bit of a cheat in order to deviate just far enough away from accuracy to satisfy many less well educated listeners.




Think also of the vintage sound of a loudness contour switch. Better yet, think of the variable loudness control Yamaha included in their early line of receivers/pre amps.

What was the purpose of the loudness circuit?

It was to function as a Doppler Effect stand in. At lower volume levels, the human cognitive functions begin to lose their linear response to frequencies. Switch on the loudness "contour" and the bass was magically restored though in reality it had only been boosted several dB in level.


Most loudness controls simply boosted the bass and that was enough. That was one of the rationales for despising a loudness function in 1970's high end audio. It was not useful and it had all the potential to add less than pleasant effects. Unfortunately, audiophiles did listen at reduced volume levels back in the day and the rules of the Doppler Effect meant they were not hearing an accurate portrayal of their system's output. For the most part, this same problem exists today if your pre amp lacks any sort of compensation for reduced volume listening.

I know of no conventional loudness control that was not phased out of the signal path as the volume control was advanced. Normally, by about 11 O'Clock on a volume control the loudness circuit was taken out of the signal path. Why? Because it was assumed that any loudspeaker paired with the amp would be at a sufficient volume level that "loudness compensation" would no longer be required - and could actually cause the amp to distort due to the additional demands of a boosted bass response.

A few manufacturers actually applied a more realistic compensation control which adhered more strictly to the presumed need for a Doppler Effect anti-curve. These controls boosted both the lowest and the highest frequencies.

Yamaha went one better with their variable loudness compensation control. Since the amount of compensation for cognitive losses is not fixed but rather exists on a sliding scale relative to actual SPL output of the system, the Yamaha control allowed for continuous (until it too was removed from the signal path) compensation with a variable control that adjusted the actual volume of the system. It's a bit complicated to describe the way the Yamaha control worked in a few short sentences but the essence is the "volume" control was set to the expected highest level required and the loudness compensation was then used to adjust the actual listening level along with the (more or less) appropriate amount of Doppler compensation needed.

The polar opposite of the 3/5a's effectiveness would be the classic purple Honda that cruises through your neighborhood at 2 AM. Due to the inability of a typical auto interior to support deep bass, boost is applied to the one note sound of a car's subwoofer. Relying on the idea most listeners do not have a concept of "bass", loud is sufficient to sell what is otherwise crap.



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Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18260
Registered: May-04
.

"901s have a limited bass response. And that equalizer? Just makes them VERY power hungry. Adding 10db to the low end means you need 10X the power, at those same frequencies. I can't imagine a HT receiver being able to sink that much power, except at the VERY lowest levels."



Not exactly true.

901's have the real world bass response the user selects. As with any speaker system with front and rear firing drivers, room placement greatly affects response. Also, as with any system employing front/rear firing drivers, cancellation and addition effects can strongly influence real world bass response.

The eq does, however, provide the user the ability to add some sense of deeper bass (see above) though normally at the expense of other musical values.

You could roughly think of the 901 as the equivalent to a line array - sort of. By coupling the output of multiple drivers, the total response is greater than the multiple of the individual drivers singularly.

But, in the end, nine 4" drivers have their limits. As does the total enclosure volume. T/S parameters are not ignored when it comes to the 901.




Bose doesn't publish "specs" in the traditional sense for the 901. It would be rather difficult to do so even if they wanted to - which they don't.

Due to the manner in which "sensitivity" is "correctly" measured, the 901 shares many of the same disadvantages seen in any dipole/bipole system. Depending on room placement and the deep nulls which might occur in response due to the direct vs reflected output of the system, a traditional measurement which well suits a monopole system could greatly disadvantage the 901.

Bose will never intentionally put themself in a position where they feel they have a disadvantage.

The point being, the traditional sensitivity measurement relies strictly upon on-axis SPL.

The concept of "power response", which is really a more meaningful idea of how a system responds in a typical room enclosure, is never stated in system specs.

Yet you should know from your experience with panels that the sensitivity spec for a multi-radiational patterned speaker system is not exactly what you get when you listen under real world conditions. The typical +6dB add from the reflected sound pressure waves is excluded from the traditional measurement technique.

This is sort of what Bose uses to say specs don't tell the story of the 901. Sort of.




Yet, you seem to be thinking of the earliest versions of the 901, leo. Yes, those speakers were a sealed system which was rather power hungry.

Once again, Bose is never going to allow themself to be intentionally disadvantaged.

The later variations on the 901 theme became vented systems with somewhat higher efficiency drivers.

While not a Klispchorn, or even a Cornwall, more like a Heresy, the later 901's lost the power hungry label and actually became a bit of a rock and roll speaker. Late generation 901's would not have given up much in the way of SPL to even the more aggressive vented rockers in any showroom.

Running the eq with the 901's in most showrooms was difficult due to the operation of the common switcher employed in simple demo rooms. In the end though, the later 901's were a fairly easy sell based mostly on reputation and a decent presentation in a shop.



The amount of boost is adjustable with the 901's equalizer. While not impossible to achieve, I can't remember ever running across a 901 user who had maxed out their eq. Usually, the drivers cried out long before the eq did - once again, nine 4" drivers have their dynamic range limits.

So I wouldn't assume T-Bass is using 10dB of boost. And, as he goes on the explain. He is not using a receiver.


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Gold Member
Username: Magfan

USA

Post Number: 3368
Registered: Oct-07
You might be correct about one thing. My EARLIEST listen of a Bose 901 occurred at Pacific Stereo in (IIRC?) La Habra or Whittier? Powered by a Phase Linear 400 or 700 it was LOUD. In retrospect, they must have also be of medium-low sensitivity.
I wasn't as 'impressed' as the average listener. Both my brother and I agreed that NOBODY plays that loud. You couldn't hear an airplane crash if it happened next door. I also felt that 'hanging' your speakers from the ceiling was NOT the best idea. I couldn't put my finger on it, but THAT wasn't in my game plan, either.

I never messed with the EQ of the 901s. And I was barely tuned in as they went on to Series V or wherever they ended up.

And yes, again, reading specs requires a Law Degree and an Engineering Degree. In nearly equal measure. This, while the Emotiva folks love the endless printouts from the Audio Precision test bench used to characterize Emo amps. While some of what MIGHT be useful measures are excluded. I, for example, would LOVE to see amp makers publish power into a REAL (though simulated) speaker load.

Here in SoCal, I wouldn't know where to start looking for a place to hear the current 901.
 

Gold Member
Username: Magfan

USA

Post Number: 3369
Registered: Oct-07
I always heard that 'doubling' referred to a bass drivers inability to produce the lowest fundamental. Say, a fundamental of 16hz (Lowest Organ Note) will reproduce as 32hz. I've heard speakers do this.
The speaker than produces the first harmonic or 'double' the fundamental.
It can happen at higher frequencies, too. A speaker unable to produce the lowest notes of a piano (Even an Imperial Grand) or a bass guitar can 'double up'. This might give the impression of more bass.
 

Gold Member
Username: Magfan

USA

Post Number: 3370
Registered: Oct-07
The only speaker which DOES use the Doppler Effect is the Leslie.
While the originals DO have moving parts, the newest versions are ALL ELECTRONIC but achieve the same effect.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18262
Registered: May-04
.

"I always heard that 'doubling' referred to a bass drivers inability to produce the lowest fundamental. Say, a fundamental of 16hz (Lowest Organ Note) will reproduce as 32hz. I've heard speakers do this.
The speaker than produces the first harmonic or 'double' the fundamental.
It can happen at higher frequencies, too. A speaker unable to produce the lowest notes of a piano (Even an Imperial Grand) or a bass guitar can 'double up'. This might give the impression of more bass."



16Hz???!!!

Geez, leo, you have the wrong thread.




You seem to have missed entirely what I posted, "The 3/5a was not new in its use of this design element. The knowledge of a "doubling" effect was well known to speaker designers by the time the 3/5a was being designed. This is merely the knowledge of how most acoustic music operates - a stronger second harmonic than the actual fundamental frequency - and the use of a louder upper frequency component which sold the idea of bass in many early (mostly vented) speaker systems."


Ignoring the idea there were no designs striving for usable 16Hz response in the 1970's (plus the fact there was/is no real world need for such a design since the recorded source for such a signal exists in virtually no real world form), the concept of doubling was not new at the time the 3/5a was being designed in the early to mid '70's.

Yes, loudspeakers did "respond" to a subsonic input but that actually became a problem in the days of LP dominance and warp frequency effects. Hence, the inclusion of sub-sonic filters in many pre amp designs.


However, until the T/S parameters were widely accepted as science for predictable driver/enclosure design, vented loudspeakers outside the tuned pipe variety (transmission lines, Vogt Pipes, etc) were largely the result of a cut and try methodology. (Ohm's mid '70's standmount systems are the first example I am aware of which mentioned designs according to T/S parameters.) This unpredictable process of uncertain results gave rise to the truly atrocious BIC Venturis, the worst of the late ALTEC and EV systems and numerous other cheaply built vented systems which produced deep and exaggerated group delay measurements. The deep nulls and exaggerated comb filtering meant the woofers were often simply flapping in the breeze. These were the large boxes which really couldn't reproduce an 80Hz signal with any degree of accuracy.

Now the "doubling" effect in such designs became a qaudrupaling effect and the strongest peak from such a system was likely to be well above 120Hz.

This lack of true bass response in any sized package suitable for home use gave rise to the popularity of Vilchur and Kloss' compact acoustic suspension designs of the original AR products; https://en.wikipedia.org/wiki/Edgar_Villchur

By the mid-'60's, Kloss had moved on to found Advent and 32Hz response from a relatively compact enclosure (the Original Large Advent Loudspeaker) was possible. Yet the Advents were in competition with the numerous cut and try JBL's, Klipschorns, Bozaks, etc. Bass extension vs bass quantity.

With the conventional tuning of an electric bass guitar's low E somewhere in the neighborhood of 42Hz, the result of the cut and try products which still filled the vast majority demo rooms of most "audio salons" of the day was the well known, and all too familiar, one note bass which mashed all instruments with fundamentals beneath that 120Hz or so frequency into a lumpy, bumpy mess.

But doubling does work to the advantage of a loudspeaker designer. Particularly when it is combined with the proximity effect of a conventional domestic listening room. There is no "might" about it.

The question, if there is one, is more of quality of effect than anything else.

You would be hard pressed to enumerate just how many audiophile loudspeaker designs have relied on these basic rules of physics for their sonic signature and their sales success.




"The only speaker which DOES use the Doppler Effect is the Leslie.
While the originals DO have moving parts, the newest versions are ALL ELECTRONIC but achieve the same effect."



So, you're saying I'm completely wrong? But you have no proof to back up your claim. Just what you believe to be true.

OK, I'll say you are at the least not totally correct.

A Leslie did not truly rely on the Doppler Effect for its primary sound. The woofer's in the Leslie cabinet did not move relative to the upper frequency drivers. There was a bass frequency "scoop" which was rotated relative to the fixed driver. Since the driver producing the lowest frequencies did not move and the lowest frequencies produced by the system will be largely imni-directional in nature, the effect of the scoop was more in line with amplitude modulation which doesn't actually give you the Doppler Effect when the source and the listener are in fixed positions.

The mid to upper frequency horn(s) of the Leslie (crossed at 800Hz and well above the lowest working frequencies of the Doppler effect) operated on the principle of amplitude modulation first and foremost which was the result of a very directional driver moving in and out of an on axis position relative to the listener. However, the Doppler Effect does not predict its existence in the broadest expanse of frequencies carried by that horn. So, yes, the Doppler effect is evident with a Leslie but only in the upper frequencies when fed a live music source.

http://www.theatreorgans.com/hammond/faq/mystery/mystery.html


However, I would stand by the idea the Doppler Effect is quite evident in any loudspeaker placed in an enclosed space such as a domestic room. As a design tool, it is well known and extensively utilized by loudspeaker designers. Check the frequency response graphs of several small standmount speakers for evidence.


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Gold Member
Username: Magfan

USA

Post Number: 3371
Registered: Oct-07
Phase Modulation Distortion? I think that's a better name for what you are calling the 'doppler effect'.
Pretty much the same-difference.

As for doubling? OK, 16hz is QUITE the musical bit of LF. No musical note is at a lower frequency. So what?
Pick a speaker with a lower limit of say�..50hz. Not unusual with a monitor type speaker. That's gonna double the lowest (or COULD double) the lowest octave of a Piano and Bass guitar.

My sub WILL do 16hz. Not loud, but loud enough to cause stuff on the walls to resonate and the largest pieces of glass (sliding door) to have a visible reflection problem. A 20hz test tone is higher. I don't think more than 4 or 5 pipe organs are equipped with the HUGE pipes necessary to reproduce this frequency.

Symphony #3 'with organ' by Saint-Saens is the source of this, though only a few recordings faithfully reproduce the full impact.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18263
Registered: May-04
.

"Phase Modulation Distortion? I think that's a better name for what you are calling the 'doppler effect'.
Pretty much the same-difference.




Not at all. I didn't name the Doppler Effect and I didn't decide how it should operate.

I can tell you, a loudness compensation circuit has nothing to do with "phase modulation".






As for doubling? OK, 16hz is QUITE the musical bit of LF. No musical note is at a lower frequency. So what?
Pick a speaker with a lower limit of say 50hz. Not unusual with a monitor type speaker. That's gonna double the lowest (or COULD double) the lowest octave of a Piano and Bass guitar.





Only if you can name a speaker with a brick wall filter at the low end. If you are discussing a real world speaker which behaves according to real world rules of physics, then you are completely wrong.

Real world speakers do not have brick wall filters. The low end ROLL OUT is dictated by the enclosure type. A sealed cabinet acts as second order filter while a vented cabinet gives twice that amount of ROLL OUT.

Therefore, a speaker with a lower limit of say 50hz is not a real world description of a real world speaker. Is the 50Hz "limit" deemed the limit of "usable" frequency response? Or, is 50Hz the -3dB, or -6dB, point? Or, greater still? Or, it just is?

You've described a speaker that doesn't exist but that gives you room to debate you could be right. Or you are simply being sloppy in your conception of an idea.

We are now discussing a non-existent speaker of your own mental construction that has no relationship to the real world and, therefore, cannot be a real speaker. Just what am I supposed to do with what you make up, leo?



Therefore, unless you claim this 50Hz "lower limit" is the result of some new design type, if this speaker is either a vented or a sealed enclosure type, the speaker you loosely describe as bass limited will still reproduce 42Hz.

It will reproduce the lowest note of a Bosedorfer grand piano less well, I'll grant you that. That is, however, another exception to what is expected of home audio equipment intended for actual music and not special effects. No one with any sense buys a standmounted speaker IF they want to reproduce the lowest note of a Bosendorfer piano with any degree of accuracy. Not unless they fully intend to add a subwoofer which then make a 50Hz limited speaker another non-existent system. Sorry, leo, we can't just make up rules that don't apply and then say they do.

It will reproduce 42Hz at a reduced level compared to 50Hz but it will reproduce 42Hz.

And it will reproduce a strong harmonic frequency at 84Hz.

Remember that frequency, leo, we are going to use it here in a minute.

That is, in many ways, a piece of the logic for a sealed cabinet. It has a natural roll out in the lowest octaves (second order) which makes a good many more frequencies usable than would a vented cabinet.




The question then becomes how the designer compensates for the lower octave of response. It is common for many loudspeakers with a roughly 50Hz "usable limit" to employ a slight bump in the 80-120Hz range - just where the harmonic will be emphasized.

And just where the psychology of perception suggests we "detect" the type of instrument we are hearing. Based on the harmonic structure of the total sound, not on the fundamental alone.

This slight bump will give a perceptual impression of slightly deeper bass extension than actually exists. That's what the Doppler effect tells us.

Do the homework and stop arguing what you "think" can top me. Look at the measurements.


How is the rest of your post helping T-Bass? You still have the wrong thread for any of this.



.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18264
Registered: May-04
.

"My sub WILL do 16hz. Not loud, but loud enough to cause stuff on the walls to resonate and the largest pieces of glass (sliding door) to have a visible reflection problem."


Walls and stuff on the walls resonate at more than one frequency. Your subwoofer driver reproduces more than one frequency. Put those two ideas together, leo.

Things resonate when they reach their sympathetic resonant frequency. That can be any one of the harmonics of any other frequency.

I have never quite caught on to how people can think sound is only one frequency. That sort of thinking leads to comments such what you posted. Things shake when the sub is outputting 16Hz.

As with your belief there are only a few organs with pipe length of sufficient enclosure volume to create a 16Hz signal, so too are there only a few walls that will resonate due to a single 16Hz frequency.

Your walls are very likely responding to the upper harmonics of that 16Hz, not to the 16Hz itself. Certainly, as you say, the 16Hz is down in level from the 32Hz harmonic, and it is very unlikely the walls are really responding to only the lowest frequency. 32Hz is being reproduced at a higher level. So too is 64Hz. And 128Hz.

Your sub is producing multiple long pressure waves. Glass cannot contain long pressure waves and glass tends to shake at frequencies much higher than 16Hz.

It is how things work in the real world, leo.

Can we be done with 16Hz for awhile?



.
 

New member
Username: T_bass

North Platte, NE United States

Post Number: 9
Registered: Aug-16
Jan,
Will I need a seperate preamp for the other sete of speakers....so as not to introduce them to the processed signal from the 901 equalizer?
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18265
Registered: May-04
.

Do you plan on playing all the speakers simultaneously?

Your set up is however it works for you. There's less right and wrong and more I like it that way to a system set up when it comes to cabling. You do it so it makes sense to you and achieves your desired goals.

If you plan to run all speakers together off the two monoblocks, then you'll have problems with the eq intruding on the smaller speakers.

If you intend to run the speakers separate from each other (two pairs), then you can place the Bose eq in the pre amp's tape monitor loop.

Switch the tape monitor circuit into the signal path for the 901's and out for any other speaker.

Does that work for your set up? I can't tell without actually seeing your system and discussing how you want the system to operate.


.
 

Gold Member
Username: Magfan

USA

Post Number: 3372
Registered: Oct-07
I know ONE thing that is doubtless true, Jan,
Even if your WRONG you're right!
Who mentioned Loudness compensation?
And calling something by a more correct name? Who are YOU? The Vocabulary / Dictionary police?

And speakers DO double in the manner in which I describe. I've heard it.

http://www.sweetwater.com/insync/frequency-doubling/
Speaker won't produce the fundamental but it darn sure can the first harmonic! Doubling.

I simply do NOT care what is causing my walls to vibrate. I have ONLY noted this phenom when playing (VERY infrequently) a good recording of the Saint-Saens symphony #3 'with organ'. Oh, Sure, I've rattled stuff before. My now 35 year plus AND long gone 12" 3-way JBL copies with that fat, ported mid-bass would do it too. But the subjectively low amplitude and definately low frequency of the organ tone in the Saint-Saens piece REALLY do a job.

don't forget there are 2 major divisions of 'vibration' in the manner of which I speak. Forced and Sympathetic.
Tesla once claimed he could topple the Empire State Building with (some controversy) 20 hp. His resonant frequency experiments were the terror of the neighborhood and would send even the stoutest man to the men's room on an emergency basis! Fun for all.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18266
Registered: May-04
.

"I know ONE thing that is doubtless true, Jan,
Even if your WRONG you're right!"




OK, we will talk about leo some more. Your favorite topic. Your only topic it seems.

I have never reconciled with the idea of simply allowing you to go on in life with your misconstrued ideas or to actually correct you when you want to spout BS that has no basis in reality. I do know, you react to being told anything contrary in much the same manner as Trump. Proof? Well, you did finally get tired of my corrections to your BS political emails. I assume you still have me on your ignore list so you can go on being misinformed forever.

Remember, leo, you started this one.



"Who mentioned Loudness compensation?"


I did, way back before you mentioned phase modulation.

You obviously do not bother to read what I post. The link I sent regarding the Leslie system explicitly mentions AM and FM performance of the system. It says nothing about phase modulation. You pulled that one out of your backside.

So do yourself a favor, leo, and read for a change, eh? I doubt you will accept anyone's opinion if they disagree with what you want to think, but give it a try.

As the D says, what have you got to lose - other than a bunch of poorly informed opinions that are not serving you well.


"And calling something by a more correct name? Who are YOU? The Vocabulary / Dictionary police?"


I have no idea what that means.

Other than the simple fact there is a correct name for most stuff. And making sh!t up does not make you correct. Nor does it facilitate assisting anyone in need of help or trying to explain a concept to someone who wants to think they know something when they really do not.

Sometimes, leo, I just can't help you.




"And speakers DO double in the manner in which I describe. I've heard it.

http://www.sweetwater.com/insync/frequency-doubling/
Speaker won't produce the fundamental but it darn sure can the first harmonic! Doubling."






OK, you read something in your mad search to prove me wrong. Unfortunately, it doesn't say what you think it says.

Maybe this reading thing isn't for you after all.

And I will go so far as to say, you cannot handle the truth about what you hear. Saying you have heard doubling is like saying you have seen the sun come up in the morning. Geeez!

Musical instrumnts do not produce a single fundamental frequency! Can you not get that in your head? Of course, you've heard doubling. It exists in nature. Virtually everything that resonates creates a strong second harmonic and surely anything that is set in motion produces multiple harmonic frequencies.

Why would you be the only person who has heard a second harmonic?

Good grief, leo!

Here's what your Sweetwater link says, "Generally caused by overloading a low-frequency speaker, frequency doubling makes bass instruments sound an octave higher than they really are. This is because the overdriven speaker is making the second harmonic louder than the fundamental pitch."

OK, first, take note that doubling refers to the SECOND harmonic. There is no first harmonic. The first harmonic is referred to as the fundamental and the second harmonic is the doubling of that frequency. Get that part right and we might be getting somewhere, leo.

Sweetwater is essentially correct, doubling is a more serious matter when the low frequency driver is overdriven. Doubling occurs though whether the driver is overdriven or not.

Why?

Because in the vast majority of acoustic instruments, the second harmonic - that frequency that is the result of doubling the fundamental - sounds at a higher level than does the fundamental itself.

"doubling" = twice the amount = "second", got it?

You act as though musical instruments produce only a fundamental without any harmonics. That's incorrect.

The only case where that would be partially true is in the case of, say, a drum where there is such a strong fundamental as to say there is really only one note being produced. However, just as the laws of physics say we cannot cut a length of string to the point we have its final division, the drum head and body still resonates in the familiar pattern of all things that create sound acoustically.

Therefore, we can say all acoustic instruments resonate from the fundamental out to infinity. And, we know from measurements (those things you refuse to look at) most instruments create a stronger second harmonic than they do their fundamental note.

Therefore, if the driver is down in level sufficiently at the fundamental - remember, speakers do not have brick wall filters but rather roll outs based in their enclosure type - what we will perceive is the strong second harmonic which is within the driver's bandwidth.

That is a simple rule of physics. leo.

I'd say check it out but you have probably stopped reading by now since this is already showing you have no idea what you are talking about.

Add to this fact of how instruments work and plug in the basic idea the driver itself resonates in sympathetic reaction to the electrical input of signals and you have a system with not only high harmonic distortion levels but a device which tends to act much like a musical instrument in that is creates a strong second harmonic also. So, if the second harmonic is present and the driver's harmonic distortion is high due to being overdriven, guess what's going to happen.

Put together the two concepts - I know, leo, abstract thinking is not your forte - and you have a driver which plays a second harmonic more forcefully than the fundamental for multiple reasons.
So, yes, loudspeaker drivers double. No one here ever denied that fact. You simply stopped reading before you got to anything other than what leo wants to think.




"I simply do NOT care what is causing my walls to vibrate."



Then why do you keep talking about it?

Most of these threads have absolutely nothing to do with the equipment you own, leo. Yet almost every thread you enter has you discussing your stuff and how it performs. Why?



"I have ONLY noted this phenom when playing (VERY infrequently) a good recording of the Saint-Saens symphony #3 'with organ'. Oh, Sure, I've rattled stuff before. My now 35 year plus AND long gone 12" 3-way JBL copies with that fat, ported mid-bass would do it too. But the subjectively low amplitude and definately low frequency of the organ tone in the Saint-Saens piece REALLY do a job."


So, which is it? You've only noticed this with this recording or you've done this before?

Look, leo, you are making a fool of yourself. No one expected your JBL "copies" to have extended bass response. You shook the walls with SPL at a higher frequency.

Walls and windows resonate when you slam a pressure wave into them.

Make the wave large enough and you will shake things on the walls.

Period. It doesn't take a 80' long wave to do so.


And do not forget there is more to any musical note created by an acoustic instrument than the sole fundamental frequency.

You have nothing to prove what you claim; that the 16Hz signal is causing the stuff on your walls to reach their sympathetic resonant frequency. You simply want to believe it because the idea sounds good to you.

Get a signal generator, leo, and run a single 16Hz frequency through your system. If stuff resonates, then you have a bunch of stuff that responds to 16Hz input. Though, you will still have to realize your walls are not moving at only 16Hz. They are acting like all things that resonate at a fundamental and its harmonics.

Otherwise, you only think this stuff is shaking at 16Hz because that one recording is the only piece of music you have that supposedly contains the 16Hz frequency. It is printed on the label, right, that this recording contains a 16Hz organ pedal tone? That is the only way you know what frequency is being output, right?

You have no idea what other frequencies are present in any other piece of music so you think 16Hz much be the magic joojoo for shaking stuff on your walls.



Like anyone cares. And like I am about to spend time explaining this to you again.

Remember, leo, you started this.



Give it up, leo! I do not care and I'm pretty sure someone asking about their own components doesn't really give a crap either.



"don't forget there are 2 major divisions of vibration in the manner of which I speak. Forced and Sympathetic.


Yes, and we've discussed both here, you just were not paying attention.


How long are you going to carry on with this, leo?

It has absolutely nothing to do with T-Bass' thread.

And I really am tired of you posting one completely blathering thing and I must then post lots of things that try to explain to you what you posted is blather. Blather which you don't bother to read anyway.

How about you try to stick to the topic of a thread for a change and see how that works for everyone involved.

No one cares about your stuff and no one cares about this one recording with the 16Hz frequency.

OK?



.
 

Silver Member
Username: Unbridled_id

ChicagoUsa

Post Number: 742
Registered: Mar-04
Jan baby....still kicking like a mule !
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18267
Registered: May-04
.

FISHY!!!

They're letting you have access to a computer now?
 

Silver Member
Username: Unbridled_id

ChicagoUsa

Post Number: 743
Registered: Mar-04
Ahhhhh still with the acid tongue, and as they say the more thing change. I am pleased to see have not gone senile quite yet you old crazy lady. I am still waiting for that invite down to Texas. I expect a 12pk of shiner black lager and good Texas BBQ.


 

New member
Username: Djhireauc27

Newyork, NY United States

Post Number: 1
Registered: Aug-16
dj agency
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18268
Registered: May-04
.

"Ahhhhh still with the acid tongue, and as they say the more thing change. I am pleased to see have not gone senile quite yet you old crazy lady. I am still waiting for that invite down to Texas. I expect a 12pk of shiner black lager and good Texas BBQ."


What's your prospective release date?

Or, are you still waiting for the parole hearing?

.
 

Silver Member
Username: Unbridled_id

ChicagoUsa

Post Number: 744
Registered: Mar-04
Be nice Jan.... you have it in you. I see that you missed me, and I appreciate your "affection". I assume that the

invite to your place to check out your gear and drink is still in it's infancy.

I certainly would like to continue our past dialogue, and look forward to gleaning what I can from you and all the rest

of the gang.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18269
Registered: May-04
.

"Da'gang" be gone.

I think most finally saw the futility of chasing endless audio BS.

A few moved to a new forum or another. IMO they got tired of the endless wars that occurred here.

One started his own forum though it's been slowed since he was diagnosed with cancer awhile back.

A couple stay in touch with me through emails.

One or two decided to waste their money on other hobbies. From what I hear, they're doing a really good job at it.

It's been pretty dead around here for the most part. Some people wanting to revive some old pieces of gear mostly. Or relive their rock and roll days of 30 years ago.

Questions like, "How good is my 1979 JVC reciever? It's been in storage for sixteen years and now it doesn't work."


Those are people who don't typically like my answer.



For some reason, there are a lot of people who think this is the car audio forum and we have codes to unlock their radios so they can cheat on paying someone for content.

It's a bit disturbing to find someone asking for a radio code in the "phono" forum. Not a lot of confidence they'd even get the message they don't belong here.

I don't think Brian gives much of a care whether this portion of the forum stays or goes. He's making cash no matter.


.
 

Gold Member
Username: Magfan

USA

Post Number: 3373
Registered: Oct-07
Go to Elliot Sound Products and read THAT article on 'doppler effect' for the somewhat better term.

And how did TRUMP get drug into this? One half of the Ultimate Dumpster Fire election. A lunatic and an (unindicted) fellon. I'm still waiting for the Web Hubble / Chelsea DNA test results.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18270
Registered: May-04
.

"Go to Elliot Sound Products and read THAT article on 'doppler effect' for the somewhat better term."


It would have been more helpful if you had been more specific about what Elliott has to say on the Doppler Effect and how it relates to any point you might be trying to make. I've actually lost track of, and lost interest in, what point you might have been trying to make, leo, it has nothing to do with T-Bass' thread.

Elliott "says" more than one thing about Doppler and simply referring to Elliott doesn't clarify anything. "THAT" article doesn't exist unless you say "WHICH" article and say "WHAT" he says that relates to your opinion - whatever that may be at this point.



None the less, I went to Elliott and I read a bit. As I see it, he is primarily discussing Doppler distortion. And he makes it clear his statements are his opinion, which is not always the most common opinion.

In other words, Elliott is, as far as I see, discussing the intermodulation distortion created by the driver's movement toward the listener and away from the listener.

That is not what I am discussing in regard to "doubling" in a loudspeaker.

In short, as it relates to loudspeaker doubling, the Doppler Effect suggests louder equates to lower/higher. That basic idea was/is the basis for many loudness compensation circuits, Yamaha's being the most well known.

We understand the utility of this "loudness" idea as; when there is more bass, the (average) listener will perceive more to be lower. Our perception fills in the missing points just as it does visually when we view a pointilist image It's how our human mind operates by seeking to fill out an image despite missing parts or incomplete data points.

Therefore, cognitively, when there are more high frequencies, we perceive them as being more extended in frequency also.

Nothing to do with movement or distortion.

Nothing to do with phase.

Now you're trying to combine too many of your claims to make any sense when it comes to doubling.

It seems even you have lost track of just what exactly you are trying to claim, leo, and now you are simply throwing everything on the wall to see what sticks.


Doubling is a frequency issue, not a distortion issue or a phase specific issue. Even Elliott discusses the compression effects of Doppler as it relates to the upper frequency response of a driver and the modulation of frequencies produced by one single driver when it comes to phase related issues.


None of this has anything to do with T-Bass' thread.



I will try to make this as short as possible and hopefully as easily understood as possible.

One more time; the sound of an acoustic musical instrument is not simply a fundamental frequency.

The sound we recognize as any specific musical instrument, a car horn, a jet fly over, a wooden stick beating on a metal pan, a human voice, etc. is constructed of a fundamental (lowest) frequency and its harmonics which extend upward and out into infinity.

If we attempt to reproduce, say, a 35Hz fundamental frequency generated by an acoustic instrument, we depend on the loudspeaker's ability to portray that frequency with accuracy if we are to accurately recognize the low frequency note as existing.

Human perception takes our attention to the louder of two sounds.

It is well known the average listener will gravitate to the louder of two (otherwise identical) speakers when asked for a preference.

All loudspeakers have a bass roll out which is defined by their enclosure type (with the sealed enclosure being the acoustic equivalent to an infinite baffle).

At some point in the frequency response of a loudspeaker we assign a roll out point where bass extension is sufficiently down in level to be less easily perceived in relation to the upper frequencies. We say that frequency is beneath the
"usable" extension of the system.

If the 35Hz frequency is beyond the "usable" frequency response of the loudspeaker system, it will be largely ignored despite the fact the driver will respond to the signal input. Drivers are non-selective in their ability to respond to a Voltage input.

If the signal is 6-8Hz and generated by a warped LP/tonearm resonance, the driver will respond to that input. This is a problem many pre amp designers have addressed by the addition of a sub-sonic filter which typically has its greatest effect beneath 20Hz.

If the "second harmonic" of 35Hz - 70Hz - is within the usable bandwidth of the system, we will hear the louder 70Hz signal as the predominant bass sound.

Our mind will fill in the missing fundamental.

That, quite simply, logically and provably, is "doubling".




The issue is made worse when the loudspeaker is overdriven due to the excessive harmonic distortions created by a driver, not the intermodulation distortion product caused by phase cancellations and additions.

None of this is a secret.

Loudspeaker designers have been using these simple ideas for decades to create speakers which have a bumped up bass response which sells in a demo room. They rely on the basic concepts of the Doppler Effect for this bump; louder = lower = better.

Numerous audiophile speakers designed since the advent of the LS3/5a have employed the same bump found in that speaker's mid-bass to give the sellable impression of bass extension that is less than real.

Go look at the measurements of the standmount speakers reviewed in Stereophile, leo.

You still refuse to do that simple thing.

You tell me to wade through Elliott but you simply will not look at a measurement which would show you the effect I am describing.

I'm hoping that explains the bumped up bass effect sufficiently that we are not going to spend the next week on this. Because none of this has to do with any question posed by T-Bass.

You are so desperate to find me wrong on any one thing that you drive these threads into the dirt with your constant arguments that have no point and no end.

Give it up, leo.



Leave the rest of your post in the toilet, where it belongs. Trump was brought into this thread for the very reason you just demonstrated. Any comment you perceive as a slight does not require a rude response.

Remember, leo, you started this.



.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18271
Registered: May-04
.

You seem to be hung up on the mechanics of the Doppler Effect, leo.

"Doubling" in a loudspeaker has nothing to do with the mechanics of the action. It has to do with the "effect" the loudness of a low frequency sound has on human perception.

You need to separate the two ideas, "no" to mechanics and "yes" to perceptual effect, to grasp the manner in which the Doppler EFFECT relates to perception as it is being used in loudspeaker design.



If you prefer, you can begin by looking at doubling as a result of the human cognitive function we see described by the Fletcher/Munson curves, which establish the equal loudness contours used in acoustic design.

Though that is not exactly accurate either since the reason we say the loudspeaker has "doubled" the bass fundamental is the mechanical action of the resonant system which has rolled off the lowest octaves of bass response.

One law describes the mechanics while another describes the effect. We need to look at both as a manner in which we can apply what we understand to solve a problem we have encountered.

In other words, one function gives us the problem, one law gives us a solution as defined by yet a third rule of EFFECT.

Not one constrained, linear thought process of mechanics alone.

Bass roll out is described, as expected, by the laws of physics which describe the actions of bass roll out beneath the resonant frequency of a cavity, roughly the rules of a Helmholtz resonator. Bass roll out however does not describe doubling, F/M curves or Doppler EFFECT.

We need to combine all of these bits of knowledge to have the answer for why perceptually we hear "doubling" at the second harmonic yet perceive lower fundamental frequency response.



In other words, there is no one simple answer to this and you will have to carry several independent concepts in your thought process to find the perceptual EFFECT of doubling and why it works.

The speaker cannot accurately reproduce the fundamental lowest octave (as dictated by the laws of a resonant cavity) so we perceive the next higher octave (if it is within the capacity of the resonant system and knowing most second harmonics are elevated in level relative to the fundamental and the upper harmonics) and mentally we fill in the empty space which is the expected fundamental. Perceptually this then coincides with the instrument we expect. We hear an organ and not a clarinet.

The EFFECT of Doppler mechanics describes the perceptual action of the human mind to louder = lower/higher as we see in the F/M curves.

F/M basically answers the question of why Doppler works but does not address the mechanics of Doppler.

Got it?

Fletcher/Munson simply defines the perceptual loss of frequency response relative to falling loudness levels which is the Doppler EFFECT but not the mechanics of Doppler.

The Doppler EFFECT provides the description of a perceptual action (without the mechanics) we can implement to compensate for what the F/M curves tell us.

It's rather like saying we are heating water without going into the molecular actions of water and heat.



Abstract thinking, not linear process.




Better?



.
 

Bronze Member
Username: T_bass

North Platte, NE United States

Post Number: 11
Registered: Aug-16
Gentlemen, If I may.
It wasn't my intent to begin a days long dialog. Sorry.
JAN.....I am splitting the signal from the preamp to 1 (for the time being) Carver M400 for the 901's.....and to a Carver TF-6CB for the other speakers. Since the preamp has a single output that I'm splitting the equalized signal will be going to each amp. Correct?
I want to be able to operate both sets of speakers simultaneously.
And I know that the equalized signal should not be introduced to the "non-901" speakers.
Hopefully this helps.
I'm sorry for taking so much of your time.
Maybe I can get this thread ended....Thanks...Ted
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18272
Registered: May-04
.

"I am splitting the signal from the preamp to 1 (for the time being) Carver M400 for the 901's.....and to a Carver TF-6CB for the other speakers. Since the preamp has a single output that I'm splitting the equalized signal will be going to each amp. Correct?
I want to be able to operate both sets of speakers simultaneously. "




Where have you placed the Bose eq in your previous set ups?

You have several choices for locating the eq in a pre amp/power amp system.

As discussed previously, the tape loop would be a very good choice for a system with only one speaker pair (the 901's).

However, the eq can also be placed between the pre amp and the power amp. The only issue with the eq, as far as the 901's are concerned, is that the power amp driving the 901's is receiving a signal with eq applied.

So split your signal going to the two power amps at the output of the pre amp and place the Bose eq in line - after the pre amp output - going to the power amp dedicated to the 901's.

The second amp will not see the eq in that hook up.



It's not your fault how the thread goes, T-Bass.


.
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18273
Registered: May-04
.

What's your plan for controlling the relative volume differences between the two sets of speakers when both are playing simultaneously?

The pre amp's vc will only provide an overall volume level. If you need to trim the louder system, then you'll need more gear.

How are the two sets of speakers located?

In completely different rooms? Or, where they will share a common space?



.
 

Bronze Member
Username: T_bass

North Platte, NE United States

Post Number: 13
Registered: Aug-16
They will be in the same area. 901's in NE and NY corners. Hanging from the ceiling. I've decided to use Spica TC 60'S For the other set.
Where would you recommend positionong them?
 

Platinum Member
Username: Jan_b_vigne

Dallas, TX

Post Number: 18274
Registered: May-04
.

Sorry for the delay, I've been out of pocket recently.

I would recommend the TC60's go in another room.

I have absolutely no concept of the soundfield you wish to create with your positions for the 901's. Most important in your considerations I would say is the barrier effect of such placement.

By placing a loudspeaker close to a barrier/boundary - a wall, floor or the ceiling - you will create several changes in its "in room response".

For each barrier present, the pressure wave created by the speaker's output striking that barrier will be bounced back toward the direct firing front wave placing it in addition to, and in opposition to, the direct frontal wave. Due to the longest wave lengths existing in the lowest octaves, for the most part this reflected energy will provide a bass boost to the mid and upper bass response perceived by the listener.

However, unless the speaker has been specifically engineered for corner placement (seating tightly against a wall to form a perceptual "infinite baffle" or deeply into the corner so the distances are predetermined by the design itself) this will typically provide a preponderance of out of phase contributions to the perceived in room frequency response.

Now, with both in phase and out of phase wave fronts arriving at the listening position together, you have a severe comb filtering effect; https://www.google.com/search?q=comb+filter&rlz=1CAACAJ_enUS705US705&oq=comb+fil ter&aqs=chrome..69i57.7981j0j1&sourceid=chrome&ie=UTF-8

Bass quality suffers from the filtering effect which makes for a rather lumpy bumpy sound quality overall. Often, vocal qualities are diminished due to the location of the bump provided by corner placement.

If there is one "most significant" criticism of the 901 sound, it is this essential comb filtered perception with time/phase arrivals being at its base.



Quickly, when Amar Bose developed the original 901 he was basing his design concept on test results which suggested the perceived sound for a listener in a known acoustic space (a theoretical old school concert hall) was approximately 8/9ths indirect (reflected) sound. The 1/9th remaining was the direct sound projected by the typical orchestral instrument with its near omni-directional soundfield. This gave Bose the idea to replicate this 8 plus 1 formula with his eight rear firing drivers which bounced reflected sounds toward the front of the enclosure and towards the listener.

Unfortunately, what Bose totally ignored was the fact any domestic loudspeaker must operate inside another acoustic barrier which is the listening room itself. Now his theory quickly begins to fall apart when you realize his majority of reflected pressure waves will also be distributed into the room only to also be bounced off the room's barriers where they too will be perceived by the listener as containing a disproportionately high level of reflected waves arriving at the listening position and a very low level of direct output from the single front firing driver.

This realization of what actually occurs within a domestic listening room led many 901 owners to turn their speakers back to front so the eight drivers then became the direct sound source while the (now) rear facing single driver provided minimal contributions to the perceptual soundfield.

The best placement for the 901's then would be for their enclosures to be sited at ear level where the front firing drivers have the greatest chance of being detected. Too high in the air and the driver's output is sent over the listener's head only to be added to the (already too high in level) reflected sounds created by the rear firing drivers and the room boundaries.

The single cone drivers used in the 901's have very limited dispersion and the front firing drivers are best placed on axis with the listening position. Therefore, height is somewhat critical and toe in is almost always best for the 901's rather than firing directly down the length of a wall.



Comb filtering and in phase response has now been sacrificed in the 901's design on the altar of marketing of the unique Bose design concept.



While the original measurements upon which Bose developed his premise had the luxury of establishing a single location for an imaginary listener in relation to the hall and the orchestra, there are no such hard and fast rules for a home audio system.

It is certain however the Spicas are the direct refutation to the Bose concept.


By taking pains to create a time and phase coherent system, Spica's John Bau created a loudspeaker which took into account the numerous errors found in many prior designs and more or less threw their opposition to the non-coherent systems such as Bose (and many, many others) directly in their face.

In the development of the "modern" loudspeaker's design, there was a period when such attention to electrical and acoustic phase (and time arrival at the listening seat) was a prime driver for further development to follow and a critical selling point for the accuracy of such systems.




Without getting too far into the weeds of how a loudspeaker operates inside an enclosed room, the Spicas are intended to sit on a stand, well out from the barriers (boundaries) of the room itself. The listener is preferably placed on axis to the front baffle of the enclosure putting their ears in the most direct position to perceive the primary pressure waves emitted by the system and to minimize the reflected arrivals. Thus, if such placement is employed (and the room is treated as having a minimizing effect to the output of the speaker), will the source material be revealed as most closely corresponding to the original performance of the music. In other words, what comes out of the Spica will be heard as the analog to the original event as it was recorded by the microphones.



Lots of maybes still exist in that reality but that is the design concept of the Spicas.

The 901's are the antithetical paradigm to the Spicas.

I can think of no two speakers which are more diametrically opposed to each other in how their designers thought of the original event.

The 901's sounds are intentionally diffuse and "big" in the sense few instruments can be placed in the perceptual soundfield in accordance with their actual position within the performance venue. Sound is everywhere and not just emanating from two boxes. This gives the impression the Bose enclosure disappears into the perceptual soundfield. This an effect which was somewhat unique for its day.

The Spica is, in contrasting technique, capable of creating a replication of the original event with the equal, if not superior, ability to disappear from the listening experience while retaining the specificity of a realistic "image" of the performance as it was recorded on the source material. Details of the performance are enhanced as they are not lost in the phase and time errors of "lesser" designs.

If you were to take the Spicas alone as your main stereo speaker system, you could easily detect the differences in presentational style between their design and that of the 901's.



The Spicas should be placed on relatively short stands (get the tweeter about ear height for a seated position, about 32-38" off the floor) and away from the side and back wall by about 3-4' each direction. Try to place the Spicas about six feet, and no more than nine feet, apart from each other. Less than six will slightly diminish the soundfield size while much greater than nine will begin to have too many indirect sounds arriving at your ears which will dilute the benefits of the Spica design.

With the Spicas properly located within the room, then locate your listening position as the tip of an equilateral triangle with the two Spicas forming the base of the triangle. Toe the Spica cabinets inward to point at your listening position - try a toe in that places the cabinet sides of each enclosure invisible to you from your listening chair. This will be the most on "axis position" for you and should provide the best results for this experiment. Avoid, if possible, placing your listening chair too close to the wall behind you or you will once again create the reflected field of pressure waves which results in slightly bloated bass lines and rather indistinct images of performers.

Concrete blocks or a hard chair/stool where both speakers can be placed at equal height can serve as your stands for this experiment. Listen to your best recordings of your favorite performers and pay attention to the soundfield created by the Spicas.

You should be able to perceive the placement of individual performers within that soundfield. The soundfield itself should have dimension slightly in front of and extending well behind the plane of the speaker baffles. If there is a single primary vocalist in the recording, their voice should occur as a very tightly focused source generally centered between the two speakers. If you sit in your listening chair while someone speaks in the room, this is then same quality you will likely perceive from their voice. It will not be large and diffuse as the Bose would present it.


If your room doesn't intrude too much on this set up, the speaker enclosures should be perceptually difficult to place as direct points of the triangle from which instrument sounds emerge. It is actually probable the soundfield will extend well beyond the outer sides of the Spica enclosures. Given some attention to set up, what the Spicas are capable of producing is a soundfield which has a three dimensional depth and width which extends in front of, to the sides of and behind the plane of the speakers.

If you decide to try this experiment, first, make certain your amplifier and your Spicas are wired in phase. (Always make connections with your amplifier powered down and allow a few seconds during which time the amp will drain its power supply.) In other words, an in phase connection is red terminal to red terminal on both amp and speakers and black to black in both locations. There should be an identifier on your speaker cabling which allows you to know which leg of each cable is which at each end, usually a stripe or some printing. In phase will produce the best (desired) result from the Spicas while out of phase (one cable reversed red to black at ONE SPEAKER only) should give you a sound more reminiscent of the 901's - diffuse and big with no specific central image of individual instruments or vocalists.


The Spicas are the direct opposite to your (mostly out of phase) 901's in style of presentation. Where the 901's scratched at a few of the defects apparent in loudspeaker design in the late 1960's, Spica overcame many of those obstructions by the mid 1980's.




Whether you decide to experiment with the TC60 placement or not, they were a bit full in the lowest octaves and need to be placed off the floor and away from walls, especially the barriers of a corner or a tri-corner (floor or ceiling placement close to a corner) for their smoothest response.

I cannot tell you which sort of presentational style you might prefer, only that the loudspeaker world as moved beyond the, excuse the word, "mistaken" applications of direct and reflected sound waves within a room enclosure s they exist in the design of the 901's.

What you like is what you like but there is no doubt the Bose and the Spicas are not at all alike in how each treats a real world music event.

Does that help?


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