Doc modding Marantz imperial 7

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karatestu
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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

More from Roy. Hope you are enjoying and understanding it all so far :epopc:


gma952b65 posts
02-13-2003 1:49am
Cdc, it is time to dispel the myth hidden inside their statement, as many other firms say exactly the same thing.

You say Revel states, "The crossover networks . . . maintain a 24db per octave, 4th order acoustic response..."

FYI: Acoustic response- this is the actual frequency vs. amplitude response a mic would measure, freefield at one meter on swept sine wave tones (no floor bounce). This is the right way to describe any crossover- by its final acoustic response (assuming you even listen at one meter... which is a whole other problem).

". . . the steep filter slopes ensure good acoustical behavior in the crossover regions, with a minimum of acoustical interference,.."

This is true -no question- when measured by swept sine waves or pink noise, w/o the floor bounce. FYI: "good acoustical behaviour" means there are major no peaks or dips in the frequency vs. amplitude response on swept tones or pink noise.

"along with low distortion and wide dynamic range."

Yes, because 4th-order rolloffs keep the drivers protected really well from low frequencies- the upper-range cones/domes don't even wiggle on a bass drum.

"The somewhat steep 24dB per octave slopes also provide the benefits of keeping ALL DRIVERS IN PHASE AT THE CROSSOVER POINTS.."

Yes they are in phase, which is a benefit, but THEY DID NOT START AT THE SAME MOMENT, NOR WILL THEY STOP AT THE SAME MOMENT. THEY ARE NOT TIME COHERENT.

With a tone generator, feed that woofer and mid a steady sine wave at the frequency where the crossover occurs. Put a mic out front- look at the combined single sine wave from the woofer and mid on a `scope. You cannot see the beginning or end of this steady tone- it's just a sine wave going up and down.

Then unhook the mid- look at just the woofer. Then look at just the mid. THEIR WAVE'S PEAKS AND VALLEYS LINE UP- they are in phase. But if you could see the beginning or the end, one starts A FULL CYCLE LATER THAN THE OTHER. And when they stop, ONE STOPS A FULL CYCLE LATER THAN THE OTHER. You could even say their combined output rings at that crossover frequency.

ALL of that is in any second or third-year electrical engineering "Filter Theory" book in plain English- that is the behaviour of the 4th-order filter. In phase, yes, but 360 degrees out of step.

Can you hear this? Yes. Just listen to a speaker (or headphone) without that time delay. How much time delay was imposed? Exactly one full period of the crossover frequency. If that was 400Hz, then the time delay between the two drivers is 1/400th second, or 2.5 milliseconds, which is ~32 inches for time of travel, acoustically.

You just smeared the guitar spatially by ~32 inches, front-to-rear, and transiently by 2.5ms, across its 400Hz range (just above middle C). It sounds like the upper strings of the guitar are "leading" the lower strings, or that there is more pick on the string. It also changes the wave envelope, which means a loss of clarity. And all that means it changes the musical message.

But how would you know unless you've heard the real thing? All you really know is that there are many "poor recordings" you can't play, can't enjoy. Many performances "you just don't get". Why? Because that phase distortion is distorting everything that comes into it- including any recorded distortions- so you hear distorted distortion- which is multiplicative, not additive.

You are being presented a speaker that warps not only the image of the guitar, and its dynamic attack, but mis-aligns the individual tones that go into shaping its harmonic envelope, which is its timbre- the very nature of why it sounds like a guitar.

"...The steeper fourth order slopes, however, avoid the power handling problems associated with first order crossover networks."

Yes, if you use less than the best drivers.

Cdc, you remark, "and comments made above about bad in-room frequency response (some Revel is very good at) with slow roll-off x-over design 4th order sounds very convincing."

I am not sure what you mean by "(some Revel is very good at)", but all you have to do is listen to end the debate. Music waveforms have little to do with the math Revel and others use, math based on sine-wave arguments, "proven" by sine wave measurements.

The standard response to all that I've stated above about the drivers not being in sync is, "Well, you can't hear the phase errors, anyway. There are tests that prove that."

The tests don't prove that- they are ambiguous at best, because they were based on clicks and other unfamliar sounds.

If a designer has never built a speaker that is minimum phase (the standard term for any time-coherent system), he has not even tried to hear the difference.

A 4th-order acoustic rolloff is used because:
--the designer believes the sine-wave arguments,
--it allows high-tech-looking metal cones to be used, as most all of those have a severe (>10dB) peak at the cone breakup frequency, along with 5-10% distortion caused by that peak. Look at the SEAS and others' metal-cone driver frequency responses- readily available. The 4th-order rolloff chops them off.
--a 4th-order rolloff lets the designer lower costs by using less rugged drivers.
--a 4th-order rolloff is often computer-aided in its design. Sounds high-tech, and good for advertising.
--the designer doesn't have to learn time-domain math, which is quite difficult.

All the math of acoustics and physics and music supports my claims, fully. What's known about how the ear uses the time domain for image formation and timbre retrieval supports my assertations. The actual times and timing of music's transients support them. The method behind how any timbre-containing wave envelope forms as time evolves, supports them.

As far as off-axis interference? 1st-order drivers overlap a lot. So those drivers must be really good- and it's hard to find "the best". However, they are overlapping TIME COHERENTLY, `most everywhere in the horizontal plane, so there is no interference or lobing. Period.

If you stand up- yes, they start to go out of phase by many degrees. But the 4th-order circuits are already HUNDREDS of degrees out of phase, no matter where you stand or sit.

Some would claim this time-coherence stuff is not a question of right or wrong, but a matter of taste. I would remind you that keeping the speaker a minimum-phase design (not possible w/4th-order) is the ONLY way to preserve the actual shape of the musical waveform received by the mic. For inside that "shape", that irregular, non-sine wave envelope, lay all the tones of the music, all the dynamic changes, all the musicality, and all of the musical message. Why disturb it?

I welcome any attempt to refute my points on any grounds.

Most of all- just listen. Non-time coherent speakers sound like a wall of sound -no depth- as the time domain is scrambled. Time is distance. Time delay is depth. Time is transients coming and going, and tones building then decaying. Scramble the timing between bass and treble- you lose the depth. You lose the timbre. You lose the musical intent. You lose access to many, many recordings.

Time coherent behaviour can be heard in any decent headphone- even a $30 Walkman headphone, let alone the Grado or Stax. Just listen, especially to music that has energy near the crossover frequency.

Thanks. I don't want to be the constant "Answerer" here, but I really hope this helps.

Roy
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

More........ :roll:


gma952b65 posts
02-23-2003 8:47am
The simplest electrical crossover on a speaker is an inductor placed in series to the woofer, and a capacitor to the tweeter. The amplifier drives into both simultaneously. If they are perfectly equal and opposite in "reactance", then they cancel out, as far as the amplifier is concerned. This cancellation is what makes this the only dividing network without time-delay distortion.

This is a first-order network. Its two components can be used only when the drivers and cabinet designs are "perfect".

Not bloody likely.

More complex circuits are usually required, whether using two or ten more parts. The result can still be a "measured" first-order acoustic rolloff. The extra parts "modify" driver non-linearities and "make up" for cabinet reflections. Which they cannot- but they can fool the microphone. Of course, extra circuit-parts cannot be perfect either. Reductions in transparency and dynamics are givens.

To keep the number of crossover parts to the barest minimum, one has to use the most linear drivers- which are relatively few. However, not that few: specific examples include tweeters from Morel, Dynaudio, Foster, Stage Accompany, Pioneer TAD, and Scanspeak. Certain mids from Audax, Eton, Davis Acoustics, Bandor, Jordan, Foster, Peerless and Aurasound. Specific woofers from Scanspeak, Davis, PHL Audio, Volt, Audax, Peerless, Pioneer and Aurasound. There are others.

And every one of them is far more expensive than the drivers used in most designs.

For a commercial designer who wants to use the simplest crossover, it's hard to find the best drivers under deadline conditions. But by using the most linear drivers, within proper cabinetwork and correct bandwidths, the crossover circuit can be reduced to just a handful of parts, for clarity and for time coherence. The converse is entirely true.

Best,
Roy
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Roy certainly wasn't scared of sharing some of his knowledge. Here's some more......


"So, to answer your original questions, with respect to first-order crossovers-

Dispersion characteristics:
No problem with us. This does become a complicated issue when cabinet reflections are considered, which we avoid. Do note, however, you have never read of dispersion problems with most any 1st-order speaker design. The math is `way too involved to show why here, but there will be info on this on our website.

Smoothness of power response:
About the same, although this is usually botched by choice of crossover point (any style crossover) and by using spaced, double drivers in one frequency range. Biggest deviation we see when "power response" is poor, is a hollow-sounding voice range past about 30 degrees off axis to the sides. For those not familiar with this term, it was coined to describe how it might be good in some circumstances for a speaker to put out a "smooth amount" of acoustic power per frequency into the room- pretty vague, considering the "results" were an integration of the output at various angles over a complete hemisphere, which could be skewed by having a tweeter very bright on-axis and dull elsewhere- just to mention one of the flaws in "integrating". This method was championed first by the AR LST, Design Acoustics, and the Walsh driver, and now the current omni designs.
It is better to say that we want a speaker to have a smooth dispersion w/frequency off to the sides (no holes), tilted downwards in the highs so we don't send too many highest-highs to the sidewalls or wall behind the speakers.

Distortion:
Harmonic- depends on the design of the drivers. The best drivers have NO problem in any type of living-room use, with an appropriate crossover point that respects the dispersion pattern and the radiation impedance seen by each driver.
IM distortion- depends on the drivers again AND also the crossover points AND the woofer excursion allowed below 50Hz.

Wave interference:
Here's a concept- THERE ARE NO WAVES.
We heard sound only AT our ears; an air pressure fluctuation- rising and falling minutely, UNPREDICTABLY. Unless you listen to pure, single tones, sustained, like from a tuning fork- then the wave concept is useful. But only as a solution to that very simple, eighth-grade wave math. It does not describe very much about how we will hear music.
- Speakers designed only via sine wave analysis sound radically different from each other, and do not measure like they sound, because of their designers' preferences and interpretations about what sine-wave measurements mean.
- Speakers designed via the time domain approach (uses extraordinarily difficult math which can keep track of the music signal's demands), include sinewave analysis automatically (the converse is never true). And guess what- these designs sound `way more similar than different, and their measurements- no matter how performed, consistently more correlate with what we hear on music.

Off-axis lobing:
Audible only on selected sine waves. You must realize, of course, that cancellation arguments depend on "relative distance to the drivers is now different when you are standing up". And thus to get a cancellation of a particular sinewave, you must be exactly a half-wavelength farther away/closer to the tweeter compared to the mid. Which is 180 degrees. Which is 4.5" closer/farther at 1.5kHz, and 2.25" at 3kHz, and 1.5" at 4.5kHz.
So pick your frequency for cancellation. If you are standing up, remaining motionless at one spot, there is only one distance difference, say 1.5", which would then put a null on sine waves at 4.5kHz, 2.25kHz, and 9kHz. And also create partial nulls beginning within +/- 20% of that primary 4.5kHz frequency (as the distance difference reaches less/more than that 180 degrees). Which means a general dip from 3600Hz to 5400Hz. Which is less than a half octave- a few notes on the piano- only its harmonics go that high. A dip which could be "covered up" (usually is) by tweeter "splash" off a flat cabinet face.

So then move around the room a little (why else are you standing?), and the null frequencies move to different tone ranges- usually higher as you move away.

So you hear a different tonality/tone balance/timbre in the treble. Is it unpleasant because of the transient distortion? (see below) Yeah, if you play it at >95dB on music with a lot of treble information.

You did want complete honesty, right??

The sine wave math for these nulls is not inaccurate- it is just useful for sine waves and on pink noise.
What we hear from first-order speaker designs on music with its varied tones and timbres (i.e., no sustained single, pure no-harmonic-content tones) is a reduction in the treble, a compression of the depth of the image, an accentuation of the leading-edge of the low-treble sibilants (`cause tweeter is closer), a blurring of the dynamic contrasts, and a reduction in the clarity of separate performers singing the same line (think chorale and massed strings). All because of the time delay imposed by being at that "1.5-inch" distance offset. Which is a constant 1/2250 second of time DELAY (= 4.5kHz half-period = .11 milliseconds ).

Contrast that with the constantly-changing HUNDREDS of degrees of time DELAY that the higher-order designs impose, no matter where you sit or stand. Ten to several hundred times longer time delays than the .11msec above!!! Delay times that also VARY with EACH frequency no matter where you sit or stand, unlike first-order designs which give you only that ONE, constant, time delay at every frequency when you stand up. And this gross amount of varying time delay creates far worse distortions of the same kind mentioned in the paragraph above.

Not to mention that some of those designs (many) also invert the polarity of the mid vs. the tweeter and woofer, so the initial transients are also warped by one driver sucking in, while the others push out. This is a POLARITY INVERSION, which many try to tell you "well that is just 180 degrees".

Yeah- on sine waves. Tell the drummer pounding outwards on the kick-drum skin that you are going to make his snare drum whack SUCK IN, and also that his kick drum will get there THREE FEET late because the speaker has the woofers around the side of the cabinet, time delayed even more by the crossover. And then try to explain that NONE of that kick drum's harmonics will be arriving three feet late- only the lowest fundamental. The higher tones will arrive sooner, so his pulsing rhythm will sound lagging, and less powerful. And that his "sucking snare" will likely sound hollow. And the crack of his stick on the snare head will be of positive polarity, AND arriving a few inches sooner than the sound from the sucking-in skin...

Thus, I do not see the point in warping the time-domain for critical home or studio listening, especially since, for the last 15 years, we have had drivers that will handle the power and excursions required.

So, it is (not only) my humble opinion, that high-order crossovers screw up the music's timbre, dynamics, rhythm, transients and imaging, because they warp the time domain so grossly, and differently, at each and every frequency. And so on them, certainly it does not matter much where you sit or stand, or measure- it is always "out of phase", far more than standing up on a first-order speaker.

To demonstrate this, play a particularly poor recording on high-order speakers vs. 1st-order speakers. And then hear it over decent headphones- which also have little time-domain distortion (better to call it that than "phase shift").

And finally compression:
Not a problem for home or studio nearfield with the best drivers out there. Most drivers are not very linear in terms of power compression (from voice-coil temperature increases and from magnetic-field non-linearites vs. stroke). And a high-order crossover protects those drivers, and sounds high tech, and needs to be "computer designed" for the "best" results. Which is also good for advertising. And for which is easy to present the "benefits" via sine-waves.



Seandtaylor99-
Sorry Spicas, although easy to listen to, are not time coherent. They are instead smoothly time-delayed as the music moves into the treble- but this does compress the image from front-to-rear, and make for laid-back dynamics. It was indeed a higher-order circuit, necessary to protect the drivers he had available back then. The circuit he used warps the time-domain to much less degree and more gradually from frequency-to-frequency than the highest-order crossovers. He was among the first to do that. Celestion did it some in their old Ditton 33 10" 3-way bookshelf model from the mid `70's. Kef is trying that again, I believe in their new series.

Hope this helps. I cannot seem to explain the benefits of a time-coherent approach to design any more easily. Tried many times.

Best regards,
Roy Johnson
Green Mountain Audio
greenmountainaudio.com
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Had enough yet ? Here is Roy talking about what makes a speaker good at playing low volumes

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royj151 posts
07-25-2003 5:29am
Lack of phase shift is one criteria I know helps a lot- for clarity, texture/timbre, subtle dynamic contrasts and sharp imaging. But none of those comes through unless three things are present:

- the drivers have suspensions designed for high compliance at micro-amounts of stroke.
- the crossover parts, primarily capacitors, can pass very faint signals (most caps cannot).
- a very quiet cabinet on the inside.

We can rule out (as first-order causes) the linearity of the magnetic fields around the voice coils- there is no voice-coil stroke occurring.
We can rule out voice-coil venting and high-temperature voice-coil construction- as there's no stroke to create any air pressure to be vented, and little power input to have thermal changes in the voice coil.
We can rule out "extreme" cabinet rigidity, because of the low levels of energy input.
We can rule out cone rigidity, for the same reason.
We can rule out the way the enclosure is tuned (ported/sealed/T-line) as those become non-linear with INCREASES in SPL, if they are going to mis-behave.

Of course, I'm sure you know a lot of gear isn't that great at soft levels (especially interconnects- which is why I recommend the Audio Magic Sorcerer cables before any component upgrade). In fact, I know of some amplifiers which have a decidedly "off-on" type of sound that actually gives speakers with poor low-level response more "jump". Of course, an amplifier which does have excellent low-level response is termed "laid back" when auditioned/reviewed with those speakers- too suave and graceful and subtle for those speakers.

To the others- thank you for your kind compliments.
Best,
Roy
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by SteveTheShadow »

Excellent stuff, that was Stu. :guiness;

I love my single-driver, triple-cone Fanes and early on, he puts into engineering terms, what I’ve always loved about the drivers I’m using at the moment.

Only problem for most people is the very lengthy run-in time. It takes a couple of years at hi-fi rather than pro levels to get the larger whizzer to tone itself down. A lot of listeners would abandon them long before that; a mistake IMO, but apart from nutters like me, the run in time, measured in years, is a deal breaker for 99% of people.

Loosening them up by using high power 25Hz sinewaves, would not necessarily work, as it’s the upper mids that need the work. Loud warble and sweep tones would probably be the best method of accelerating run in but I don’t think your ears or the neighbours would be very happy. :lol:
Amplification - DIY solid state amp with twin power supplies and NVA boards, or single-ended triode amp.
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Speakers - Large floor standing sealed cabs, with Fane 12-250-TC 12" full range driver.

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Hi Steve, I am very glad you found that lot interesting and informative.

I certainly did and it ticked a lot of boxes and a few light bulbs have illuminated. He very accurately describes in as simple a way as possible what is going on with time lag and the way speakers without crossover can make instruments sound so realistic.

If it was me I would have put them Fane's in my band PA system and given them some abuse for a while. :grin: Well done for persevering so long with the run in. Having no crossover at all is certainly very alluring but I have chosen my poison now and there's no going back.

As far as semi omni's are concerned there is never going to be complete time coherence between the woofer and tweeter. The tweeter is always a few inches closer and firing at you whereas the mid bass is set back (because it has to be) and the sound has to travel to the room's reflection points and then to your ears. A much longer distance which obviously takes more time.

With my speakers it is even worse as I have multiple bass & mid bass as well as four tweeters at different distances from my ears and the tweeters pointing in different directions. At least I get browny points for doping and no inductor. But then there is still a cap on the tweeter with it's phase shift. The thing is there is no inductor phase shift to cancel it out unless them firing at 90 deg to each other makes a difference. Out of my depth with that a bit at the moment and need to learn more.

What puzzles me is instruments sound more like real instruments now than they ever have. I have less imaging now but that was a trade off I was prepared to make. Distortion (can't remember which kind) is much lower now I have multiple tweeters and mid bass sharing the work even at low volumes when the cones seem to be hardly moving at all. The added detail and harmonic information was a revelation.

More understanding is required
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Somewhere amongst all that Roy gave his opinion on why some people have issues with sibilance. His view is it is because the tweeter output is in front of that of the mid bass. Sounds feasible to me as that particular sound is smeared in time.

There is no hope for us semi omni users in this regard unless you position the tweeter forward firing on top of the cabinet with the dome at the acoustic centre of the mid bass. But even then the mid bass output is taking longer to get to you as it has to bounce off the room boundaries. Also there will be a reflection off the top of the enclosure. :roll:

Can't win it seems unless going to point & squirt (did I really just say that) or using an up firing fane or similar.

The more I think about this time coherence thing the more I realise what a compromised design my speakers are. Sure, the 12" and the two 5" drivers are roughly the same distance from me along with the side firing tweeters. The other two tweeters are closer to me though and this lot are reflecting off the boundaries all over the place.

It is a wonder that I have anything resembling music at all :oops: BUT it sounds glorious (although not perfect). Instruments sound more real than I have ever heard, I get regular spine tingling moments. Musicians are playing together and making some beautiful music which is rich in time, dynamic and harmonic information

I have lost count of the number of albums and songs I have revisited. Often there were tracks (or even whole albums) that never made sense on my old speakers so the tracks were sometimes skipped or the album wasn't played. Playing them now I realise I can understand them now and they have the ability to move me emotionally. Listening to Foo Fighters New Way Home from the album The colour and the shape (my favourite track from that album) and the emotional reaction is off the scale.

Job done then ? Err no. Mr Roy Johnson(RIP) has made me think of ways to improve them. Oh no, another two years of agricultural style Heath Robinson experiments were recycled materials, chainsaws and not even a frequency response, impedance or power response plot in sight. Well I have got this far without any measurements. Maybe they could have helped me who knows but the rest is going to be done in the same way I started - with real music and ears not sine waves and microphones.

Annoying spikes or dips can be addressed if we know what is causing them. My 120 -150 Hz suck out is room related I think and I don't think there is much I can do about it (apart from repositioning the speakers) at least dips are not as audible as peaks. The peak I have at 1.2 KHz - It's not a major one but noticeable with a test tone CD and my ears. There are several possible causes that I need to research more.

Reflections off baffles can be a problem it seems and I have been worried about the slot I have between the lower and upper enclosures. Am I getting slot resonances ? Well the bass and midrange will be bouncing around in there and it is bound to having some effect.

My speakers have large baffles - the upper enclosure is 40 cm wide. Plenty of scope for the lower output of the tweeters to bounce off the cabinet and smear the sound. So what do I do about that ? Reduce the baffle size somehow without affecting the volume of the upper enclosure. That would also lessen the ability for sound to bounce around between the lower and upper cabs.

Only one way to do that in my mind. I don't want to go back to only one mid bass. The only way I can think is to go isobaric with both up and down firing 5" mid bass :shock: But isobaric loading is said to muck up the midrange because of resonances in the volume between the two cones (to do with the wavelengths at higher frequencies). Well nobody complained about the cubix and that has a huge volume of air between the 8" drivers. Doc said it wasn't a problem.

But, with the clamshell configuration we can reduce the volume between cones to a minimum especially as the five inchers are doped almost up the surrounds. Slip in a reduced panel thickness between the two drivers and that will reduce it even more. Hope fully that could put any resonance above the pass band of the drivers. It's not like I play loud and these drivers don't have a huge xmax so the panel could be reduced to just enough to keep it stable in just the rebate.

Doing that I could reduce the Vb (box volume) for the two drivers to approx 6 Litres instead of 12. That would make an enclosure with internal dimensions of 16x20x20 cm. That is a lot smaller than what I have now. Just have to check that the bass response will be what I want before proceeding.

Why not a cube ? Well I need to keep the height down because of the height needed for the lower bass enclosure and the space between the two enclosures. Also at approx 16 cm internal height I can join the two internal magnets together that are facing each other and get some force cancellation that is said to dramatically reduce cabinet resonances. That could even allow the wall thickness to be reduced.

Diffraction from cabinet edges is not addressed in many designs (including Doc's cubes) but if you can do it then why not. I need to think about how rounding of all the corners can be achieved without increasing the size of the panels too much. Corner rounding with a chainsaw maybe :whistle: :lol:

But what about diffraction from the chassis and magnet of the drivers outside the box ? That can't be helped with this design I don't think. At least it attenuates the higher frequencies because there is no dust cap facing outwards , it's behind the magnet. Why not shove the up firing tweeter on top of the magnet of the up firing mid bass ? :shock:

That's enough to think about before my brain explodes.
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Another quote from Roy. I love this bloke....

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royj151 posts
06-12-2015 9:42am
Hi Bombaywalla,

It is good to be back. Working on many new things that have just come out. Hope you are well.

I thought a few folks might have posted their thoughts on my statements above, about how and why large audio manufacturers 'schedule' their new technology introductions. But at least these will be there for others who search out threads bearing this sort of title.

I wish to add that far too many audiophiles are distracted by technology or even totally fascinated by it. It completely occupies their minds, which contents them.

These same people always describe how much better their music 'sounds' by its increase in "clarity, impact, detail, soundstage, imaging, depth, dynamics, ..."

They never remark about how their favorite music changed for them, changed in what it meant to them, what it did to them, for them, where it took them, ...

Either they are insensitive to the subtleties of music (and not natural dancers) -OR- their systems/physical setups do not reveal HOW BEAUTIFULLY the world's best artists are playing just for them.

Ever have any thoughts along these lines?

Best,
Roy
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by SteveTheShadow »

karatestu wrote:
Sat Oct 10, 2020 12:43 pm
Another quote from Roy. I love this bloke....

...I wish to add that far too many audiophiles are distracted by technology or even totally fascinated by it. It completely occupies their minds, which contents them.

These same people always describe how much better their music 'sounds' by its increase in "clarity, impact, detail, soundstage, imaging, depth, dynamics, ..."

They never remark about how their favorite music changed for them, changed in what it meant to them, what it did to them, for them, where it took them, ...

Either they are insensitive to the subtleties of music (and not natural dancers) -OR- their systems/physical setups do not reveal HOW BEAUTIFULLY the world's best artists are playing just for them.

Ever have any thoughts along these lines?

Best,
Roy
Audiophilia in a nutshell:
Does your system sound as if the musicians are in the room playing for you?
No?
Keep going.
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karatestu (Sat Oct 10, 2020 2:03 pm)
Amplification - DIY solid state amp with twin power supplies and NVA boards, or single-ended triode amp.
Analogue - BTE Designs Lenco L75 turntable, SME 3009 Series II Improved pickup arm, Goldring E2 cartridge, NVA Phono 1 phono stage with ridiculously huge outboard power supply.
Digital - Musical Fidelity M1 DAC, AppleTV4 with Apple Music subscription, Tangent CDII CD player.
Speakers - Large floor standing sealed cabs, with Fane 12-250-TC 12" full range driver.

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

More from Roy - this might be better in the Houdoonu thread :lol:

royj's avatar
royj151 posts
06-05-2015 6:50pm
New technology is most often introduced to aid in marketing (even by the raw-driver factories). The technology typically does not make the sound noticeably better, perhaps just 'different'.

For large speaker manufacturers such as B&W, new technology is used to create a new model lineup. Any franchised retailer must order the 'usual amount' each quarter or lose that franchise to a competitor.

Also, that 'new lineup' feeds the review machine. Reviews and 'product of the year' pronouncements rarely create immediate demand, based on my experience and that of very many other designers I've known for more than 25 years.

However, a retailer will show reviews to a prospective buyer, likely not an experienced audiophile and whose friends, like most of ours, are definitely not.

One important yet unspoken issue for any customer is that his friends do not make fun of his purchase.

I think the following is important to understand:
When that new model lineup is announced in October, in magazines and on websites, those press releases were submitted in early August, thus photographed in early July or before.

This means the warehouse was stocked in July, ready to ship. Which means part of that big batch of 10,000, sufficient for one year of worldwide sales, was shipped to the USA warehouse from China during June.

So that entire batch was made in March, April and May.

Which means the new parts, the new technology, was established in January. So it was 'proven to' and approved by marketing before Christmas.

Therefore, it was invented a year before its introduction to the public. Which means the pressure is on the R&D team for 'making improvements' every year.

This results in 'improvements' that make only small differences. Also, that new lineup in a store is already being redesigned.

I vote for using one's ears, including listening to live music, and for looking at how often a firm announces yet another breakthrough.
DIY inspired by Richard "The Doc" Dunn RIP

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