Tech Blog

                            

SPL dosage with headphones. 

The snap of the snare makes your hair blow back.  The kick of the kick makes your pant legs tremble.  The cut of the axe as the sound waves slice through the air.  We all love to CRANK THE MUSIC, but how much is too much for our hearing safety?  This question is easier to ask than to answer because there is a lot to consider, but safety of our ears is a key issue for music lovers.

It is a new time and a new way for us all to listen to music.  Live bands are king, but listening to your favorite band play through your iPod with Klipsch headphones is like the concerts in your head.  We all can enjoy this high SPL pleasure in moderation, but how do we know that we are listening at safe levels?  American Speech and Language Association claims that nearly 50% of Americans have some form of hearing damage.  Various sources of sound are to be blamed for this damage to our hearing, but the dosage of decibels are the key.  Understanding the right dosage can save your hearing if you heed this warning.

Our speakers can easily last 50 years.  We would like your hearing too also.

Headphones are an interesting new case for exposure to sound and the lack thereof.  They can actually help save your hearing in noisy environments by blocking out many sounds, which is why we have measured specific music player setups to give you an understanding of where your ears are in relationship to exposure of sound. 

In the chart above you can see the decibel readings for certain sound sources.  We typically talk at 60-70 dB, but when we mow our lawns the level shoots up to 107 dB.  Most headphones can well exceed 110 dB with most sources of media players.  Pain begins at 125 dB.  Many Klipsch speakers can exceed 140 dB depending on where you measure the speaker, but all of these sound sources have a place in our lives making it more convenient ... or just plain Rockin’ Our World!

In the chart above we have measured our Klipsch IEM’s with several Apple media sources.  The target level of exposure to sound is 90 decibels, (dB), according to Occupational Safety and Health Administration, (OSHA).   OSHA has determined that 8 hours maximum time at 90 dB is acceptable dosage for the human ear before damage occurred.  I am not sure how OSHA determined this, but it is the standard.    The sound intensity is easily measured if the power level of exposure is constant, such as a sine wave, but music is anything but constant.  Benchmarking music applications are next to impossible due to the different levels of peak to rms ratios that occur.  I am not even going to start to explain the latest “LOUD” tactics of peak to rms compression effects that the recording industry have puked on us as of late.  My test specification is starting off simple to give you an idea of where you might be, for exposure to decibels or Sound Pressure Levels, ( SPL), at the inner ear.

At Klipsch we use the Bruel and Kjaer HATS pictured below.  This is a long name for a $20,000 piece of gear that gives us 'standard" measurements.  The device simulates the head and torso of an average human from the 1960's.  The dummy is actually a little small by today's standards, but we try to not bring that up so we don't hurt our little dummy's feelings.   

Bruel & Kjaer Head And Torso Simulator, (HATS)

Bruel & Kjaer HATS Ear Simulator

The significant thing about HATS is we can simulate the impedance of air as in a human inner ear canal.  In doing so, we can predict with certain reason what the user will experience while using our in ear monitors or headphones.  We call the location of this test condition the DRP or ear Drum Reference Position.   The response in this location is not what you might expect.  A flat free-field response is depicted below at the DRP position.  As you can see the 3 kHz region is elevated about 17 dB due to what we call the "ear canal gain".   This is a difficult concept for most people to understand, but all you need to understand for this excercise is the response below is considered flat at least to about 7 kHz.  It is my personal opinion (and experience) that the HATS does not simulate accurately above this frequency.

 

 

Okay... Stand up and take a break because were about to dig into the next set up condition that derived the data above.  In order to understand the application of a headphone with an iPod for the sound source, we must repeat the entire signal path, thus an iPod is used for the generation of sound.  We are using a 0 dB sine wave sample to emulate the music signal.  When sweeping this signal through the 20 - 20 kHz range with headphones connected to the iPod we can derive the total SYSTEM RESPONSE, which is representative of a user with Klipsch headphones and an iPod. 

 

 

Below is an example of random headphones measured.  Once the microphones are calibrated for absolute reference decibels, we can take readings of our headphones at 1 kHz.  By servoing the iPod volume level, we can adjust the reading to yield 90 dB.  These levels were noted in the form of inches and percentages of full scale readings on the iPod.  So for instance if you own Image S4 headphones, you would want to set the level of an iPod Touch to 53% to give you 90 dB at 1 kHz in your ear canal. 

Are you still with me?

 

Let's take a break from this tech bullshit and look at what you are really hearing when connected to your favorite iPods.  If you have Classic or Nano 3rd Gen iPod, you probably have a very flat response out of the headphone jack, but what happens if you have a Touch or Nano 1st Gen?  In the graph below you can see your results.  If you think that most products don't have enough bass, you might not want to blame your headphones.  The coupling capacitor in both of these devices appears to be too small and the output is reduced below 100 Hz.   Ouch... You would be better to use the 30 pin connection off the bottom of the iPod with a headphone amp if you want a flat linear response on these devices. 

 

In reviewing all the facts, even a simple iPod test is not so straight forward at deriving how loud your music is in your head.  But common sense goes a long way.  If you're in a noisy environment, like a factory or bus, and can wear headphones that isolate noise, by ALL MEANS wear them.  They may help to protect your hearing.  This will allow you to listen to music for long durations of time at lower level since the ambient noise is not an issue.  If your ears are ringing after a live concert, you are probably causing damage to your hearing.  You can use your in-ear headphones to attenuate the live sound to a manageable level, because Klipsch wants you to enjoy a lifetime of musical entertainment.

For more information, download this Hearing Survival Guide.

Special Thanks to Trey Cannon for making the measurements in our iPod chart and iPod response.

If you have been following my tutorials in speaker design, you have no doubt read the previous blog on “Geek Speak on Boxes” which dealt with sealed box designs. Please review the variables listed in that blog as I will be referring to them here...  Now let’s step a little closer into the design by looking at what a vented box can do for your speaker performance.

We call a speaker “vented” because it is just that.  There is a hole in the box to support low frequency output, but not any hole will do.  You need a specific length to width ratio to align the box and speaker driver for best performance.  The ratio of air flow to length of port gives us a Helmholtz resonance at a particular frequency.  In layman’s terms, the speaker box is taking advantage of an “air spring” to reinforce the loading of a speaker cone at certain frequencies. 

I always find that analogies are the best way to start thinking about something technical.  So let’s deal with some of the terminology.

Imagine that you have a normal 12 oz hammer.  Pick this hammer up and act like you are hammering at a nail in the air, back and forth.   Now pick up a 20 oz hammer.  Act like you are hammering nails with this hammer.  Notice how the heavier hammer is more difficult to move at the same speed.  This is an effect of M_ms or Mechanical Speaker Mass.  As the M_ms gets larger or heavier, the speaker’s natural resonance, F_s or F_o gets lower.

Have you ever been or at least had your kids play at the local playground where they have the fake animals to sit on?  The animals are connected to a large spring.  As your weight (or your kids…right) moves forward the animal rocks forward and then springs back the other direction.  If the spring was longer or stiffer the oscillation time would change.  This is similar to C_ms, Mechanical Speaker Compliance.  C_ms is the opposite of “stiffness.”  As the C_ms gets lower, the spring gets stiffer and the resonance, F_s or F_o , gets higher.  In other words as the speaker gets stiffer, it tends to vibrate at a higher  frequency -- or in your case the animal is swinging faster… I mean your “kid” sitting on the animal is faster. 

Are you still with me?  Let’s try another...

Sd is the area of a driver cone or diaphragm.  Atmospheric pressure gives a specific force upon a given area.  In an 18” woofer, there is more static air pressure on the cone than a 12” woofer.  Think of holding a typical broom in your hands.  Now start to move the broom back in forth through the air in a cyclical motion.  Keep moving it faster.  Notice as it gets faster it becomes more difficult to accelerate the broom to a higher speed. 

Now imagine that the broom is twice as wide with a handle the same length.  Notice how it is even more difficult to move through the air quickly.  Why?  This is due primarily to the surface area of the broom and the air that resists the motion due to aerodynamics.  This particular variable is Cas, or Air Spring Compliance.   You are moving more air in the larger woofer, thus your sound output goes up.

All these terms reflect the resonance of the speaker in a box with a specific air volume or “air spring”.   When you put a speaker with a “free air” resonance, (F_s or F_o), of 25 Hz in a sealed box the Fs or Fo goes up.  This is because the air spring of the box is now reinforcing the compliance of the speaker.  As the cone moves in the box the air spring pushes back making the cone want to travel back the other direction quicker. 

How many poles in your box?

We use terms like word “order” or "pole" to describe certain characteristics of response.  It is important to note that each order or pole on the design creates a 6 dB per Octave slope to the design.  So in other words, a sealed box uses the mechanical stiffness, C_ms of the speaker driver as one pole.  The second pole comes from the mass of the speaker, Mms.  Thus the bass roll off is calculated as:

6 + 6 = 12 dB per Octave

This is noted in the diagram below.

Now if we add a vent to the box, we add two more poles due to the characteristic mass of the port, M_AP and the compliance of air in the box, C_AB.  Notice in the dotted curve that the rate of change with frequency is doubled.  Now it is 24 dB per Octave.  The bass extend further to a certain frequency then it rolls off quicker and has less output much below the tuning frequency.  Nothing is for free in the laws of physics.

The port creates a filter that only lets air out at a certain rate depending on how hard the air inside is pressing.  This is like a balloon letting air out at a certain rate.  The difference is that the minute the speaker cone changes direction and travels outwards from the box the air changes direction and rushes into the box through the port.   So the air in the port is creating a repetitive motion known as a sinusoidal motion.

 

 Now let’s take this particular speaker driver in the example and vary the box volume. (I knew you couldn’t wait for this.)  The original pair of curves utilized a ten cubic foot, (10 ft3).   In the graph above you can see the effects of decreasing the volume of air, or the size of the box.  We are going from 10 ft3 to 1 ft3.   See the variation in response as the air volume shifts?

Now things are more complicated. Not only is the cutoff frequency changing, but the shape of the curve is morphing also. That is why we call this box / speaker alignment.  The box needs to be optimized to match the particular Thiele Small Parameters of the speaker. 

Notice as the box gets smaller the speaker becomes less flat in the response.  It is not optimized at 1 ft3 because the response is 5 dB higher than nominal SPL.  So our speaker will get louder in this region.  If I play a piano scale, when I get to E2, (82.41 Hz), the note will sound much louder than E3, (164.81 Hz).  Thus the recording will never sound quite like the engineer or performer wanted.  In order for us to use a 1 ft3 box we have to start over with the speaker design.  The magnet will need to be larger and the vent will need to be adjusted to give us an optimized speaker design.

 Flux This!

So you may be asleep by now with Professor Thumps speaker 101 theory, but there is more to think about, so turn those brains back on!

Let’s think about the motor structure for a moment.  If this is a typical well designed woofer the voice coil is “over-hung.”  This is different from a hangover and much more pleasant.  You could debate this point but I will keep it simple as I can for the sake of understanding the fundamentals.

 

The voice coil sits in the magnetic gap waiting for signal to pass through its coil, so it can react to the magnetic flux in the gap.  As the color of the gap becomes red in our model depicted, the magnetic force of “flux” becomes stronger; therefore the coil and cone will move farther due to the level of flux.  If the coil continues up the gap the color turns yellow then green quickly.  This is called flux non-linearity.  The magnetic force has become inconsistent as the coil travels through the gap.  If we have a short coil and it is out of the gap the 3rd harmonic distortion increases.  We don’t want this so we make our woofer coils longer.  This allows a similar amount of coil to be in the gap at all times no matter where the superposition of the coil is within a reasonable amount of excursion.  Thus the THD is lower and we are higher or at least happier.   We call this term Xmax.   Which is the dimension hanging out of the gap on one side.  It is usually 1 -6 mm for most 2 way systems or subwoofers.

 

Reviewing the response of the 1 and 10 cubic foot boxes we get the results below.

 

The performance curves above are transposed to show the excursion response curves below, for the vented box. 

So I was mentioning that the laws of physics aren’t free. (Neither is your freedom, but I won’t rant about that here).  The small box is more controlled in the excursion region down to 30 Hz, but the tradeoff is that the output is much lower.  SPL means compressing air.  There is only one way to do that.  The cone must move to displace the air.  This model is using 1 watt power, so we consider this a small signal calculation.  The coil is only moving less than 2 mm.  Imagine what it is like at 100 watts?  It is safe to say that there is lots of excursion.  That is why high efficiency speaker are so important. 

 

 So now you know a little more about what acoustic engineers do for a living at Klipsch.  Careful detail to the box and speaker alignment with a port is crutial to optimizing the performance of a Klipsch loudspeaker system.  There is no substitute.  

Klipsch…Get Yours TODAY!

Recently I discussed Bandwidth VS Sensitivity.  This blog I will delve more into the fundamentals of driver and box alignment, or what we generally call "loudspeaker tuning."  This is where the speaker engineers at Klipsch dissect all the electrical, mechanical and acoustical elements of a driver and how they relate to the air volume of the speaker enclosure. 

Richard Small and A.N. Thiele developed math algorithms for loudspeaker elements which allow engineers to model the response of a design before they even start to build the physical products.  We refer to these parameters as “Thiele – Small Parameters” (TSP).  Here are the fundamental terms:

From wiki…
http://en.wikipedia.org/wiki/Thiele/Small

“Fundamental small signal mechanical parameters"

These are the physical parameters of a loudspeaker driver, as measured at small signal levels, used in the equivalent electrical circuit models. Some of these values are neither easy nor convenient to measure in a finished loudspeaker driver, so when designing speakers using existing drive units (which is almost always the case), the more easily measured parameters listed under Small Signal Parameters are more practical.

• Sd - Projected area of the driver diaphragm, in square metres.
• Mms - Mass of the diaphragm, including acoustic load, in kilograms.
• Cms - Compliance of the driver's suspension, in metres per newton (the reciprocal of its 'stiffness').
• Rms - The mechanical resistance of a driver's suspension (ie, 'lossiness') in N•s/m
• Le - Voice coil inductance measured in millihenries (mH) (Frequency dependent, usually measured at 1 kHz).
• Re - DC resistance of the voice coil, measured in ohms.
• Bl - The product of magnet field strength in the voice coil gap and the length of wire in the magnetic field, in tesla-metres (T•m).
Small signal parameters
These values can be determined by measuring the input impedance of the driver, near the resonance frequency, at small input levels for which the mechanical behavior of the driver is effectively linear (ie, proportional to its input). These values are more easily measured than the fundamental ones above.
• Fs – Resonance frequency of the driver

 

 
• Qes – Electrical Q of the driver at Fs
 
• Qms – Mechanical Q of the driver at Fs
 
• Qts – Total Q of the driver at Fs
 
• Vas – Equivalent Compliance Volume, i.e. the volume of air which, when acted upon by a piston of area Sd, has the same compliance as the driver's suspension:
 
where ρ is the density of air (1.184 kg/m3 at 25 °C), and c is the speed of sound (346.1 m/s at 25 °C). Using SI units, the result will be in cubic meters. To get Vas in litres, multiply by 1000.”

 

At Klipsch, we tend to model speaker analysis prior to producing a single driver, but once we have some general targets for the TSP we will build drivers and then measure them using Klippel or our proprietary measurement system.  Many programs are used for TSP analysis but the program that is generally accepted by Klipsch is LEAP (Loudspeaker Enclosure Analysis Program) produced by Linear X. LEAP is quite diverse in its abilities to model in different conditions which we eventually calibrate against our lab measurements to make sure all tools are properly aligned.

 

 

In the first graph you can see LEAP in action with a generic 8 “ woofer driver.   In this simulation I have manipulated the box volume Vab from 1 cubic foot to 10 cubic feet.  What you can see is this result, which affects both the mid bass level and primarily the low frequency extension.  What you may not be used to seeing is that the response above 200 Hz is elevated.  This is what we call a "free-field response" with baffle effect from the box included.  Essentially the baffle effect is a result of the width and height dimension that the driver is place upon.  This is similar to a horn effect increasing the on axis sensitivity. 

In the picture above, you can observe how the simulation of the response is taken.  The cross hairs are where the microphone is pointed and the anechoic wedges are depicted around the walls of the measurement room with the speaker box under test in the middle.  The term "free field" means that the sound can travel spherically in all directions, as in outer space with no boundaries to reflect off of, excluding the speaker enclosure.  This is a pure condition unlike any application, but this empirical method allows for uncomplicated data to be manipulated easily.

 

 

In this next graph, the response is a bit closer to what you might see in real life conditions.   What we call this response is an “Infinite Baffle” response. The speaker could be considered mounted flush with the floor.  In this response the Low Frequency is elevated similar to what the baffle effect would be and there is no restriction on the floor length.  This condition is still somewhat artificial but similar to a measurement in a parking lot with no cars.  What you can see from this is the relationship of how the room starts to play an important role in our listening experience.

 

 

This is the pictorial equivalent for an infinite baffle with the speaker box flush with the plane of the baffle.  You can think of digging a hole in the parking lot and burying the box flush to the baffle as the empirical response condition.  There are no sound waves reflected in the back axis, so there is reflective energy at all frequencies from the plane.  This means that the “baffle effect” becomes reflective all the way down to 20 Hz.  It actually goes all the way to 0 Hz empirically but for the sake of practicality we are only modeling to 20 Hz.  Good luck finding music or movies below 20 Hz. 

Excursion Effect with a Box

Now here are some other things to think about…  Mechanical and Electrical Linearity, Distortion, Air Spring Mass.  These are more critical elements of speaker design and performance.  You can think of them as the side effects of a particular design. 

 

In the photo above you can see a cross-section view of a typical loudspeaker from Wiki.  This speaker isn’t designed correctly but I won’t get into that.  If you looked at my recent blog it shows a moving coil speaker animated.

Notice the voice coil moving up in down in the magnetic gap. 

Bl Model 

In the photo above you can observe the colors to the flux field.  This is a magnetic model showing the levels of magnetic power in the various areas of the motor structure.  As the color gets redder the magnetic flux is more dense or “hot.”  This means that more mechanical force will be driven to the cone assembly which translates to more acoustic energy.  This would be the definition of "peak efficiency."  As you move the coil up and down the placement in the flux field changes.  You can observe this by seeing that the color changes to yellow, green then blue as the density of flux lowers.  This is an effect that happens with all loudspeakers in general.  When the flux field is lower this translates to less force driving the moving assembly (cone) so the speaker becomes less efficient and the SPL goes down.  This entire effect is considered "Electro-Mechanical nonlinearity."  The primary effect of the magnetic change to the field is increase third harmonic distortion.  What you can think of it is that the cone has become detached from the translation of sound and the effect is typically harshness to the sound or coloration. 

 

 

Excursion Graph

Let’s look at the excursion differences from two sealed boxes from the previous model.  The first one is green and indicates the 1 cubic foot box volume.  As the frequency goes lower the excursion elevates slightly.  Excursion is the distance of voice coil travel in the magnetic gap.  Now let’s look at the 10 cubic foot box volume.  This blue curve rises to 3.2 mm as the frequency approaches 20 Hz.  The representative top plat thickness is 6 mm thus the coil is 50% out of the gap.  You will experience extreme distortion at 20 Hz with the 10 cubic foot box but the output will be nearly 10 dB louder.  What is different?  Air Volume... Air Volume acts as a mechanical spring.   When the cone move the air reacts in the sealed box.  If the box is small the spring coefficient to the air is also small, so it cannot displace much energy before it starts to brake the cone from travel. A vacuum starts to form in the box eliminating much chance of travel.  If the volume of air is very large, than the cone does not see the air mass very much.  Displacing a small ratio of air affects the braking effect much less, so the coil tends to jump out of the magnetic gap.   This is why the excursion is much higher for the 10 cubic foot box compared to the 1 cubic foot.  These are some of the trade offs of loudspeaker designs.  Nothing is free in the laws of physics. 

There are no perfect speakers, but Klipsch gets you a little closer to perfection.  Rock On!  

P. Thump

As most of you know, Klipsch works very hard at insuring our speakers will be “LOUDspeaker.”  Some ignore this philosophy because it is difficult to achieve, but it is the first rule that PWK founded and the first rule for our designs engineers at Klipsch. 

High Efficiency…

What some of you may not know is that high efficiency speakers do not necessarily dictate wide bandwidth or full range.  In particular, bass extension is one of the biggest challenges for high efficiency loudspeaker.  Here is why…

You can generally treat the characteristics of a woofer to the gain curve of a transistor.  This may not help you if you are not an electronic engineer but the same characteristics apply.  To make drivers more efficient you can fool the laws of physics.  Typically the motor has to be much more powerful.  This means a bigger and badder magnet and thicker magnet steel to avoid saturation of the flux.  This also means the cone has to be structurally rigid yet be a very light mass.  All of our systems are horn loaded but this effect only goes down to a certain frequency unless it is a pro theater speaker.  As I had mentioned earlier a 30 Hz horn must be 11.5 meters in length to control properly to that frequency.  We do this buy folding horns in the cabinet on our pro models and some of our home models.  So if you have a normal speaker size in a living room and you want it louder you should get a more efficient speaker first but remember that the bass may suffer if it has not been thoroughly optimized by a Klipsch engineer.

Attached is a representative graph of four theoretical driver responses.

 

In the graph you can see that the bass cutoff frequency is higher as the SPL goes higher.  This can tend to be a general characteristic for a high efficiency loudspeaker.  The 100 dB reference SPL driver has a -10 dB response at 35 Hz; whereas an 88 dB driver shows 23 Hz.  This example is highly simplified but for the sake of understanding I thought I would start out with a simple set of trends.

 

 

To make up for this difference there is several things you can do in a 2 channel or home theater system.  The most common thing to do is add a powered subwoofer.  This is a very practical approach because it takes a lot of power stress off of the receiver for the main channel speakers and reduces the chance of IM Distortion by elimination of the cone modulations in the subwoofer band.  In other words the 30Hz is no longer going to the main speaker (HP) but to the subwoofer so the IMD can’t occur.  Another option would be to use a larger speaker enclosure.  Typically the bigger the better to some extent when it comes to increased box size.  This makes some incorrect assumptions that the Thiele Small parameters are still optimized in this larger box but if the engineer has done his homework he will have modified the driver to allow for the larger box.  This is generally referred to as the Cas or Compliance of Air Mass for the speaker.  

If anyone is interested I can get into the more intricate parts of a driver design where we look at Q of systems and drivers and box volume, etc.

The next time you think about how to improve your sound system you might want to start with the speakers.  Amplifiers can only do so much, but if you have an efficient speaker such as a Klipsch you will gain more in decibels.

Keep Thumpin!

 

 

Have you ever wondered why Klipsch is so horny?

From Wikipedia, the free encyclopedia…

“Horny is an adjective that can describe any one of the following conditions:

• An animal that possesses a horn
• A slang term for sexual arousal and/or desiring sexual gratification
• Having a rough, knobbly surface — e.g., a horny skin, found in some lizards“

Hmm… Well some of that may be true but it didn’t really help describe Klipsch as a whole.

Looking under the term acoustic horn…

“A horn is a tapered sound guide designed to provide an acoustic impedance match between a sound source and free air. This has the effect of maximizing the efficiency with which sound waves from the particular source are transferred to the air.”

Getting closer….Looking at Horn Loudspeaker…

“Acoustic horns convert large pressure variations with a small displacement into a low pressure variation with a large displacement and vice versa. It does this through the gradual, often exponential increase of the cross sectional area of the horn. The small cross-sectional area of the throat restricts the passage of air thus presenting a high impedance to the driver. This allows the driver to develop a high pressure for a given displacement. Therefore the sound waves at the throat are of high pressure and low displacement. The tapered shape of the horn allows the sound waves to gradually decompress and increase in displacement until they reach the mouth where they are of a low pressure but large displacement.

A modern electrically driven horn loudspeaker works the same way, replacing the mechanically excited diaphragm with a dynamic or piezoelectric loudspeaker.
Modern horn designs typically feature some form of conical, exponential or tractrix taper… “

Now we are getting some where.  At least the term tractrix is used.

 

 

“…Horn technology history
 
 Photograph of the original painting of Nipper looking into an Edison Bell cylinder phonograph with a horn loudspeaker.
The physics (and mathematics) of horn operation were developed for many years, reaching considerable sophistication before WWII. The most well known early horn loudspeakers were those on mechanical phonographs, where the record moved a heavy metal needle that excited vibrations in a small metal diaphragm that acted as the driver for a horn. A famous example was the horn through which Nipper the RCA dog heard "His Master's Voice". The horn improves the loading and thus gets a better "coupling" of energy from the diaphragm into the air, and the pressure variations therefore get smaller as the volume expands and the sound travels up the horn. This kind of mechanical amplification was absolutely necessary in the days of pre-electrical sound reproduction in order to achieve a usable sound level.[3]”

Ironically there is no mention of Klipsch in this entire article whatsoever.  That will have to be remedied later.  Klipsch didn’t invent the horned loudspeaker, but it has stuck true to its principles. What we can learn from this is that PWK was correct in pursuing high efficiency designs for his 10 watt amplifier. The laws of physics haven’t changed.

So lets understand a little more about why Klipsch is so horny and why you should be horny too.

 

Acoustics 101

To start off, we need to define general concepts about acoustics, starting with what an acoustic wave or waveform is. 
Sounds are waves of pressure variations moving through the air. These waves can be compared to waves moving in the ocean, although they move much more quickly and in three dimensions, unlike water waves, which are confined to a two-dimensional surface. When we hear sounds, what we are actually experiencing are minute vibrations of air. As waves of these small air currents pass through our inner ears, they stimulate the nerves in tiny hair-like projections. Our brain then translates this stimulation into the audible sound that we hear.

 

Since a sound wave consists of a repeating pattern of high pressure and low pressure regions moving through a medium, it is sometimes referred to as a pressure wave. If a detector, whether it is the human ear or a man-made instrument, is used to detect a sound wave, it would detect fluctuations in pressure as the sound wave impinges upon the detecting device. At one instant in time, the detector would detect a high pressure; this would correspond to the arrival of a compression at the detector site. At the next instant in time, the detector might detect normal pressure. And then finally a low pressure would be detected, corresponding to the arrival of a rarefaction at the detector site. Since the fluctuations in pressure as detected by the detector occur at periodic and regular time intervals, a plot of pressure vs. time would appear as a sine curve. The crests of the sine curve correspond to compressions; the troughs correspond to rarefactions; and the "zero point" corresponds to the pressure which the air would have if there were no disturbance moving through it. The diagram on the next page depicts the correspondence between the longitudinal nature of a sound wave and the pressure-time fluctuations which it creates.

 

 

 

                

A simple two-dimensional plot will partially describe the entire three-dimensional pattern. Sound travels in a spherical 3D pattern, but this is difficult to document on a piece of paper.  The waves depicted in this line of the plane wave tube are just that.  (Plane as in one geometric plane)  Imagine if you will that these are travelling to all areas of space as the sound moves away from the vibrating cone that originally created this wave.  This is where directivity is of concern for an design engineer at Klipsch.

 

 

 

Polar Directivity Introduction

Wavelength of Sound as a Function of Sound Speed and Frequency...
 
Now that we know a little more about sound waves lets discuss the why Klipsch uses horns to project sound wave patterns. 
When a sound wave is generated, it travels away from its source in a beam, in similar fashion to a beam of light emanating from a flashlight.  The angle of the wave propagation is determined by the ratio of its wavelength to the size of the opening through which the beam is projected. 
The acoustic radiation pattern is a function of the frequency of operation and the size, shape and acoustic phase characteristics of the vibrating surface of the diaphragm.

 

 

 

Transducer Radiation Patterns

The beam width is usually defined as the measurement of the total angle where the sound pressure level of the main beam has been reduced by 6 dB on both sides of the on-axis peak.
 
When describing the beam patterns of transducers, two-dimensional plots are most commonly used. They show the relative sensitivity of the transducer vs. angle in a single plane cut through the three-dimensional beam pattern. For a symmetrical conical pattern, such as that shown below.

 

 

At low frequencies (when ka is small) a loudspeaker radiates sound equally well in all directions (a boxed loudspeaker will even radiate low frequency sound into the region behind the box). As shown in the animation below, sound waves radiating from the speaker spread out evenly in all directions. This behavior is primarily why the location of a subwoofer doesn't really matter - you can place it pretty much anywhere and it will still fill the room with sound.

 

As the frequency gets higher, but assuming the speaker diameter does not change, the value of ka increases and the speaker becomes directional. That is, the sound energy produced by the speaker becomes channeled into a preferred direction and very little energy is radiated at other directions. In the animation below the radiated sound is pretty much contained within a cone of 55 degrees from the center axis.

As the frequency becomes even higher (and ka becomes much bigger than 1) the sound field radiated by the speaker becomes even narrower and side lobes appear. Now the main lobe of radiated sound is limited to about 20 degrees on either side of the central axis, and the pressure amplitude falls off rapidly as you move away from the central axis. Notice that the side lobes are much lower in amplitude than the main lobe. Also note that the sound waves in the side lobes have the opposite phase as the sound wave in the main lobe.If you were using the same speaker (a large woofer) to produce both low and high frequencies, you would definitely notice a severe drop-off in the loudness of the higher frequencies as you step away from in front of the speaker. Fortunately, well designed loudspeaker systems from Klipsch don't attempt to send all frequencies through the same speaker but instead use a waveguide or horn for high frequencies.  

3D Directivity Maps are Another way to view directivity in a 2D or 3D plot Amplitude (color), Angle vs. frequency.   

 

So the next time you are thinking about loudspeakers, and LOUD is one of those terms, think about Klipsch.  We are still sticking to horny!

 

A key factor in acoustics is the effect of the room environment on the loudspeaker.  This is actually half of what you are hearing.  The loudspeaker distributes the sound around the room, drastically, influencing what we hear.  Loudspeaker systems do not radiate uniformly at all frequencies due to the enclosure shape, diffraction, driver directing effects, and driver interference near the crossover points.  A uniform frequency response off-axis results in more uniform room reflections, which directly contribute to a stable virtual source that is not frequency dependent. 

If that is too complicated, just remember you are hearing the room as much as the speaker.  We call this effect Room Gain and Imaging.

When someone speaks to you, they usually face you.  It would be odd if the person holding a conversation had his back to you, but if he or she did you would expect their voice to sound different.  Why?  It has to do with where the sound is traveling and how high the pitch is for that tone.

In order to properly evaluate a loudspeaker on and off-axis, Klipsch developed a custom program for dispersion acquisition.  This program allows the design engineer to quickly analyze the level of sound in relationship to the vector angle of that sound wave.  The results show how the Klipsch horn technology helps to control the energy radiated by the loudspeaker. 

A typical speaker distributes energy unevenly throughout the room in an uncontrolled manner.  The energy reflecting around the room influences how we perceive the recorded sound.  Loudspeaker systems do not radiate uniformly at all frequencies, due to the enclosure shape, diffraction, driver directing effects, and poor summation between radiating sources at the crossover regions.  A uniform frequency response off-axis results in more uniform room reflections, thus directly contributing to a stable virtual source which is not frequency dependent. This effect, in turn, directly contributes to a stable depth of imaging or perceived reverb as intended by the recording engineer.

 

The graphs shown are directivity data in relationship to frequency.  We call this a directivity response or map.   The colors define 3 dB increments in Sound Pressure Level, (SPL).  5 straight bars would be considered a perfect system but merely impossible to design.  If it existed, it would surely be a single source in one VERY large horn.  The mouth on the horn would need to be about nearly 57 feet for a 20 Hz horn.  The angle of dispersion would define the mouth size.  57 foot horns are very acceptable to wives, so we typically make the horn shorter and sometime fold it, like a Klipschorn.

The graph below is a competitive loudspeaker whose name will be undisclosed, but safe to say it is a direct competitor of the Palladium P-39F and NOT horn loaded.  So now you can see that the back axis dispersion is much more excessive at some frequency than others.  This data alone supports the fact that the competitive speaker will be more dependant on the room for a reasonable frequency response.  There is no choice.  The room becomes a bigger part of the equation.  If you are listening to this speaker in a linear reflection room, the sound will still be nonlinear due to the erratic dispersion off axis.  Please don’t overlook this fact.  Not only is a Palladium speaker much lower distortion it is also a purer response off axis. 

 

Nothing sounds like a Klipsch… and nothing sounds like a Palladium.

 

 

 

 

The Art of Palladium
Get yours here...

 

 

 

50.8?  

Thomas Holman (THX) claims that anything more than 20.2 is diminishing returns.  Most people are happy with 5.1, but with the new formats of 7.1 via Dolby True HD and DTS HD Master Audio superior formats can be enjoyed. 

What the heck is this? Is it really worth it? How do you get there?

 

SO WHAT DOES HD AUDIO Mean?

From the Dolby and DTS websites we have the following definitions:

“Dolby TrueHD Benefits
Produces 100 percent lossless audio that is identical to the studio master
Takes full advantage of Blu-ray Disc capabilities, with 7.1-channel playback
Offers the ability to support more than 16 channels of audio
Makes connecting your home theater easy, with a single-cable HDMI™ audio and video digital connection“

 

“DTS-HD Master Audio provides lossless audio that matches, bit-to-bit, the original movie's studio

master soundtrack and is fully backward-compatible with all DTS decoders.”

Standard 7.1 speaker layout

Standard 5.1 speaker layout

 Standard 7.1 Rear Surround speaker layout

It is already a little confusing with the discrepancy of the layouts from the pictures but what DTS is

trying to describe is different scenarios for speaker placement.  As if consumers aren’t already

confused enough, I say we stick to one standard.

As you may know from my previous P Thump Blogs, we have a new Palladium Theater room.  We are currently

running it off a simple receiver to prove that you can get stellar audio performance at beer barrel

prices…well the speakers are a kajillion dollars but the electronics are approachable for the average

consumer.  You can get compelling 7.1 audio from any Klipsch system so don’t fret.  But converting your

current multichannel system to from 5 to 7 channel can be a challenge with out spending some coin on new

electronics. 

I currently have a 9.2 setup in my home theater, but the electronics are piecemeal separates from

Klipsch employee sales and 5.1 channel receivers.  I want to keep things discrete to have the ultimate

control of the signal chain but decoding in new formats, such as the 7.1 formats, requires me to upgrade

my electronics. 

 

Cost effective decoding…

I am a bit of a tight wad when it comes to certain things and an overachiever when it comes to others. 

I can understand spending a lot of money on loudspeakers.  Steel and Magnets cost money and you can’t

easily cheat this equation.  Thus I think big motors and lots of them is the only way to get that

maximum horsepower.  Spending more money on Klipsch speakers can save you 90% in amp power requirements

because the speakers are so efficient.  So I go big when it comes to loudspeakers.

Electronics are another issue. 

 

 

 

 

 

DTS has recommendation as follows for hooking up your DTS HD Master Audio

"1. Your high-def TV, high-def DVD player, and AV receiver all have HDMI connections.

This is the best possible scenario. Do this: Using an HDMI cable, connect your Blu-ray Disc or HD DVD player’s HDMI output to the AV receiver’s HDMI input. Then, connect the receiver’s HDMI video output to your high-def TV. The receiver processes the HD audio signal and passes it to your speakers; it also processes the HD video signal and passes it to your high-def TV. Simple, all digital, no loss in quality, and you get true high-definition audio and video. Plus, because HDMI allows communication between components, the audio and video “move together” when you change sources with your receiver’s remote. Note: your AV receiver must be able to process very high-quality video signals.

2. Your high-def TV and high-def DVD player have HDMI; your AV receiver doesn’t.

Use a HDMI connection between TV and player, for full high-definition video. Then, use a coaxial digital audio connection between player and receiver. Remember: because movies and music with DTS-HD encoded content contain a DTS Digital Surround “core”, your older receiver will play back DTS-HD material with DTS surround audio at twice the data rate of other DVD video surround formats. So, you’re still going to get higher quality sound than you’re used to hearing.

3. Your high-def DVD player and your AV receiver have HDMI; your hi-def TV doesn’t.

For high-definition audio, connect your player’s HDMI output to your AV receiver’s HDMI input. For the video hook-up, use a Component Video connection between your player and your hi-def TV. You will get regular high-definition video (1080i/720p) with this connection, which is excellent but not quite as good as full high-definition (1080p)."

 

Considering the options that I have on a limited budget this is what I am considering...

1.DVD Decode

This is a cheap solution because the practically give the decoder away with the Blue Ray player.  I have

a Sony player that has a decoder with 8 analog channels out. But now I won’t have an easy solution for a

common volume control if you are using more than one preamp or receiver.

 

2.Cheap Receiver to Decode

This may be a good solution but for me I already have several good receivers that will yield 100 watts

per channel.  Without spending bigger money this may be a compromise.

 

3.Aragon and updates

There was a rumor that I can use a DB 25 to RCA cable to send the Decoded Audio to the Stage One.  This

system give the impression of 7 channels because it will decode Dolby EX.  Ignore that fact that this

format has a mono set of rears that are matricides of RS/LS.   In talking with Rick Santiago from Indy

Audio Labs, he indicated that there would not be an upgrade for the Stage One to do HDMI and TrueHD

Audio but there most likely were limitations to decoding from the Blue Ray.  This would depend on the

particular player.  He also has a model number from Whirlwind for a DB-25 to RCA breakout cable.  Here

are his comments...

"The DB-25 to RCA that allows a Stage One to take full advantage of a 7.1 decoder-equipped player is

custom-made for Indy Audio Labs by Whirlwind Cable and is available exclusively through Full Compass. 

The part number to reference when ordering is: 061609-004-DG.  We’re directing our customers to go that

route to experience the latest HD audio formats with a Stage One (or ACT-3 or SoundStage with the Stage

One-level internal upgrades).  Customers can contact Full Compass directly or call Indy Audio Labs with

any additional questions."

Indy Audio is planning on some new announcements in the fall for products.  We at Klipsch wish them ALL

the best!

 

I can't even imagine how consumers are doing with considerations of Blue Ray formats.  Most are probably

still struggling to understand what Prologic is and why their rear speakers don't make sound all the

time.  It is hard enough for an engineer to understand the logical steps for upgrading their theater

systems.  But in the end,(0r at least in the next 5 years), people will be plugging their newly bought

integrated systems with HDMI cables and letting the system do the rest.  Now if only they had robots for

the placement of speakers...Forget about WAF!!!

 

 

Thanks to Falcon20X for the cool picture at the front of this blog!

In this final installment of the Palladium crossover design I will get into the thick of it, the final touches and delicate nuances of equalization to give you that satiny smooth shimmer from a speaker system with ultra flat response. 

In the picture of the crossover you can see many large components on this board.  This is only one of two boards on the P-39 crossover.  No expense was spared in the deliberate design of this crossover.  If you look closely you can see that all the components are very high quality components, with thick wire, fat caps, and premium ceramic resisters.  The coils are all air core, which means that saturation issues are minimized.  The caps are very high power at 250 Volts.  The resisters are elevated off the PCB to improve the thermal transfer to air.  All of these characteristics are determined to improve the linearity of the acoustic response at high power.

 

 

In the schematic for the circuit you can observe the simple topology.  It is simple because the tweeter and horn mate together eloquently and require very little massaging to line up with the rest of the system.  In a perfect world you would not even have a crossover other than for power protection at the frequencies it was not intended.  That is basically what you have here.  The super-tweeter was designed specifically for this system thus there is nothing to fix.  While most Hi-Fi speaker companies are forced to buy off the shelf components, Klipsch customizes every component for their need.  Not only is this a compression driver with high efficiency it is also a “super” tweeter with output to 30 kHz.  In horn loaded systems it is difficult to get flat responses.  The trade off is high SPL and lower distortion, but in this tweeter there are no trade offs since the response is very smooth and requires not equalization.  In this schematic there is a third order high pass filter with a pad to attenuate the level.  Simple…Pure…Magic!

 

In the low frequency section I have left out the cascade network because it is redundant and I am running out of paper to show you.  But what you have is two different low pass 2nd order networks set at different frequencies which we call a cascade network.  Why two LP systems?  The strategy is to match the area of the radiating surfaces to blend the propagating wave fronts in a balanced way.  In other words, if you have two areas such as a mid bass horn and a woofer cone with equal area, their influence on the summation at the crossover region will be equal thus the energy will be equal in the room as the spectrum of sound move up in frequency.  This will give you equal Q values so that equal energy is loaded into the room.  If this rule was not applied you would have additional sound shooting up or down but would not be measured on axis.  You could hear this additional energy in the room from the reflections on the floor or ceiling but it would not show up in the frequency response on axis.  This is something most people ignore if they are following the polar directivity of a system.  This measurement requires an anechoic chamber large enough to have the microphone.

 

The filter depicted is a second order with an additional EQ to add a shelf to the upper woofer so that the phase summation is complete.  This network sums with the midrange. 

 

 

 

There is a general rule that I use in designing acoustic loudspeakers…“Get the vocal region right!”  It is crucial to get the midrange right because the vocals fall into this region.  Everyone was birthed by their mother.  Well unless you were a test tube baby  but the point of the matter is that we all know what a female voice sounds like.  It was the first thing we ever hear and hopefully it was the thing we paid the most attention to growing up.  This inherent trait defines how we react to the midrange.  Our ear is tuned to the upper tones of the female voice, so one must never ignore the importance of this region of tones.

 

The heart of the P-39F system is the midrange.  This defines the uniqueness of this product more than any other area.  Few if any of our competitors have a higher sensitivity in the midrange than our palladium floor standers and center channel and that is what sets us apart from the rest.  When you listen to the Palladiums it will be the first thing you notice after the great looks, transparent vocals that are effortless, because the distortion is so much lower than anything else you have ever heard.  On top of this we have adjusted the response to be ultra smooth so nothing honks or barks at you.  This is why there are so many components in the midrange.

The eloquent design of the midrange can be broken down into regions which I have denoted in the picture of the schematic.  This may not be straight forward for the novice crossover designer so let me break it down further and show you the response with and without that particular node. 

 

In part II we revealed the basic function of a band pass filter above.  There is the resistor pad “R”, the HP filter “C” and the LP filter “Z” (which is a wire wound inductor).  The symbols are superimposed on the graph to denote the filter effect.

 

 

 

 

 

 

 

 

 

 

 

I will now go backwards in development of the finished midrange design showing the output response. The graph above shows the effects of the Z2 LCR node (resistor R47, inductor L 25 and the capacitor C19). This is a tuned notch filter.  The frequency is centered for the L and C and the resistor resist a total short to ground.  If indeed there was no resistance in this node the impedance of the system would drop to zero, because the circuit is shorted to ground at that tuned frequency, thus the resistor inserted to increase the system impedance.  Notice how Z2 pulls out the spike at 5 kHz.   This allows the response to be well behaved when converging with the tweeter circuit.

 

 

Z2 and Z1 are now deleted from the circuit to show changes from these nodes.  Z1 is a similar tuned circuit at 400 Hz which allows the same smooth transition into the woofer circuit.

 

 

C18 is now considered in this parallel trap.  R44 is the resistor that is attenuating the midrange but when we include the C18 cap we create a short or bypass around the R44 attenuator.  At this point we see a high pass filter centered at around 1300 Hz.  We can consider this a HP EQ.

 

 

Inductor L24 is now considered in the circuit above.  This device is the sister to C18 in the fact that it is a low frequency bypass instead of a high frequency bypass.  The response again is elevated in this LP region creating the effect of the final haystack shown.

 

 

 

The response above demonstrates how each section of the Palladium P-39F work together.  We call it a

“crossover” or Xover because the separate filtered section create an “X” when the slopes crossover.  

The summation of all sources is plotted in red, the green is LF, Blue is midrange, yellow is HF and the

black curve is the system impedance curve.  Since each of the sections are in phase with one another we

show total acoustic summation.  If one were to reverse the polarity of one section you would see a deep

valley in between sources due to the lack of summation equal to the slopes of the filters.

 

The Palladium midrange is a major achievement in the book of audio.  Check it out at our listed dealers.

 

 

 

 

Today we are going to dig deeper into the design theory and process for the Palladium P-39F crossover. 
 

If you were to study Electronics 101 which would go over simple filter design you would find that there are three primary components for passive crossovers; Resistors, Inductors and Capacitors. 
Resisters perform similar to their term; the resist  electrical current. 

Inductors according to Wikipedia… “An inductor or a reactor is a passive electrical component that can store energy in a magnetic field created by the electric current passing through it. An inductor's ability to store magnetic energy is measured by its inductance, in units of henries. Typically an inductor is a conducting wire shaped as a coil, the loops helping to create a strong magnetic field inside the coil due to Faraday's law of induction. Inductors are one of the basic electronic components used in electronics where current and voltage change with time, due to the ability of inductors to delay and reshape alternating currents.”

Capacitor according to Wiki…“A capacitor or condenser is a passive electronic component consisting of a pair of conductors separated by a dielectric. When a voltage potential difference exists between the conductors, an electric field is present in the dielectric. This field stores energy and produces a mechanical force between the plates…

An ideal capacitor is characterized by a single constant value, capacitance, which is measured in farads. This is the ratio of the electric charge on each conductor to the potential difference between them…

The properties of capacitors in a circuit may determine the resonant frequency…”

In the first schematic to the right you can see a high pass filter.  This filter allows higher frequencies to pass through the crossover without attenuation due to the use of a capacitor (C).  The resistor (R) is included to attenuate this section for purposes of equalization.  You can see in the response curve that the light blue “raw” curve is transposed to the dark blue curve after the crossover filter is inserted in the circuit.  This design helps protect the drive component by limiting the exposure to low frequency power.  In this case it minimizes the excursion of the diaphragm due to the low frequency.

In the left schematic you can see a low pass filter.  This filter allows lower frequencies to pass through the crossover with little attenuation due to the inductor (Z or L).  You can see the effects of the crossover in the graph.  The higher frequencies are reduced, and the sharp resonant peak is greatly reduced to an insignificant level. 
Setting the value of both the C’s and Z will tune the crossover to the specific needs of the driver to yield a flat frequency response.

 

The schematic above shows all three types of components…R, C and Z.  This circuit is considered a band pass filter because it allows a midrange bandwidth of frequencies to be passed to the midrange driver.  By setting the Z value to the top of the filter range and the C to the bottom we can create the optimum band pass filter.
 

This next graph shows the schematic for a P-39F with all values listed.  Multiple components or poles are used to give more extreme filtration to their intended drivers.  This allows for minimal interference or cancellation between drivers in the “crossover” regions. 
 
You cans see the prescribed poles or orders to each section of the crossover.   There is also a fourth filter set for cascading LF woofers which is not depicted. 

We will break this down further in the next installment of the "Art of Palladium" blog. 

Crossover Features
  
 • Optimized 3 ½ Way Crossover (low, mid, high)
 • 4th Order Crossover for all sections
 • High linearity air-core inductors
 • Polypropylene capacitors (No electrolytic capacitors)
 • Ultra low inductance power resistors
 • Inductor layout optimized for minimal electromagnetic interference (EMI)  
 • Crossover physically isolated within enclosure to reduce sound coloration
 • Crossover matched at factory within 1% tolerance

Through very accurate driver engineering, the system offers excellent performance without the need for corrections in the crossover network.  The crossover should provide smooth transition between each drivers allowing for operation within a given bandwidth of the overall loudspeaker response. Thus, the network can be of straightforward design with extremely high quality parts.  This also means the speaker will have a more direct and coherent character as compared with contemporary products that must employ more complex network circuits. High quality polypropylene capacitors are used along with air-core inductors in all circuit topologies to ensure the best performance and reliability.  The network system produces a 4th order electro-acoustic filter when combined with the smooth, natural roll off of the drivers.  The 4th order filters provide a rapid transition between drivers at a rate of 24dB/octave successfully meeting the criteria initially set forth in this design.

The crossover was designed for the woofers to operate up to 500 Hz, the horn loaded midrange from 500 Hz – 3.2 kHz, and the horn loaded tweeter 3.2 kHz to 30 kHz. The graph on page 16 shows the transition of each filter and the combined result of all of the filters. 

The network is located within the speaker’s base plate which allows for the tri-wire input connectors to be conveniently located at floor level.  This minimizes the length of speaker wire and produces a very clean finished look.  The tri-wire input allows the end user to have a choice of amplifier/loudspeaker combinations, such as connecting a separate amplifier to each section of the loudspeaker providing individual amplification of the low, mid, and high frequency drive sections.

All network components are integrated into two separate high current printed circuit boards (PCB).  Each PCB is a dual layer board made with a 2oz. copper on each side allowing for very high currents to be passed.  All components are soldered into place and securely attached to the PCB to minimize component resonance.  Each board is securely fastened to the bottom of the P-93F with special plastic isolation grommet mounted between the board and the enclosure.
A picture of each assembled crossover is shown below.

Let's take a look at the un-equalized or raw response of each transducer now.  In the graph above you will notice three discrete frequency responses for the LF, MF and HF sections of Palladium P-39F.  These measurements are 3 meter mic distance with 2.83 Volts drive level to each section.  To convert you would add almost 10 dB to give you a 1 watt / 1 meter reference level.  So the design is quite sensitive with ranges in the pistonic region from 95 to 108 dB.  

With considerations of the shape of the raw driver response and the roll off of each section, the crossover points and design can be considered. To define the best crossover frequency is not trivial.  This will predict the power response of the speaker, the power capacity, the level of distortion at high drive level and the final sonic characteristic of the system.   

Acoustic Power – Acoustic power out is a combination of the driver efficiency and dispersion of the components.  Compression Phase Plugs will improve driver gain in the higher band of the driver and the angles of the walls on the horn will define the Di of the component. 

Power Capacity and Fc – Defining the low cutoff frequency (Fc) of the component will define the amount of excursion a driver will exert.  This will ultimately define the mechanical limitation of the component.  Equalization will further determine the thermal capacity of the driver by steering the spectral density of power into the. 

THD and IMD – Total Harmonic Distortion and Intermodulation Distortion will be affected greatly by the Fc of the xover. 

SPL – Efficiency will be determined by the mix of the crossover and passive EQ. 

So it is clear that the crossover pays a most important roll in the design of any good speaker system, and Palladium is no exception to this rule.

 

 

High-Frequency Driver (Tweeter) Design

 Horn Loaded Tweeter Driver Features
   
 • 90° x  60° Modified Tractrix Horn
 • 10 to 1 high compression phase plug
 • ¾” lightweight titanium diaphragm
 • Ferro-fluid cooled voice coil
 • High efficiency magnet structure containing 2 Neodymium magnets
 • High Temperature N35SH Neodymium magnets

In order to deliver the dynamic range targeted in the design brief for this product it was necessary to develop a new tweeter from the ground up.  Starting with a clean sheet of paper, the engineering team created a driver that is both sensitive and accurate.  Low distortion, high sensitivity, wide bandwidth and neutral tonality were pushed beyond the standards for any previous Klipsch driver.  A detailed exploded view of the tweeter is shown below.

 

 

The chamber behind the tweeter dome is resistively damped via a filled tube to reduce reflection of the back wave and thereby the distortion that would otherwise be produced.  The phase plug that serves to put the dome into a compression mode also has a unique chamber within it that is designed to extend the upper frequency limit of the driver by more than 10 KHz.  This proprietary technique offers numerous benefits including the elimination of undesirable standing waves in the high pressure layer between the phase plug and diaphragm. 

This driver is inherently flat from 3 KHz to 30 KHz. In this application, this tweeter will operate from 3.5 kHz to 30 kHz.  It has an inherent sensitivity some 10 dB greater than direct radiating designs, and distortion is reduced by a factor of 10 dB vs. conventional tweeters.  A detailed cut-away of the horn loaded tweeter is shown below.

 

 

Built around a ¾” titanium dome, the tweeter makes use of two very large high temperature N35SH neodymium magnets.
As with all of the drivers used in the P-39F, the motor structure for the tweeter has been developed in house and optimized using FEA analysis.  The below picture shows the tweeter motor structure magnetic FEA.  This shows the high flux saturation in the steel around the voice coil which helps to reduce flux modulation thus reducing distortion.

 

 Mid-Frequency Driver (Midrange)

 Horn Loaded Midrange Driver Features
   
 • 90° x  60° Modified Tractrix Horn
 • 4 to 1 high compression phase plug
 • 1” Diameter – High temp copper voice coil 
 • Aluminum cone w/ Aluminum dust cap
 • High linearity magnet structure containing 3 different Neodymium magnets
 • High Temperature N35SH Neodymium magnets
 • One copper pole shorting cap
 • Half-roll foamed rubber surround
 • Low reflection die-cast aluminum basket
 • Optimized rear midrange chamber

The inverted dome midrange driver used in the P-39F is a unique design not found on any previous Klipsch product.  PWK is famous for saying, “The midrange is where we live”.  A true high fidelity loudspeaker reproduces a flat midrange response, where the human ear is most sensitive.  This particular midrange driver has been designed to operate from 500 Hz to 3.2 KHz.  This preserves polar uniformity; avoiding assignment of information to drivers too large to deliver uniform dispersion.  A detailed cut-away of the horn loaded midrange is shown below.

Special attention was given to heat management; ensuring full dynamics regardless of drive level unlike many dynamic designs where increased power leads to voice coil heating and a rising voice coil resistance.  That change causes driver output to fall just when it should be at its peak.  The heat generated by the voice coil is dealt with in two ways.  The first, being through the low thermal resistance of the aluminum cone.  The second, through the purpose-cast aluminum basket which acts as a heat sink for the motor parts attached to it. This basket is designed to have very narrow spokes avoiding reflection of rear wave driver output that might otherwise smear the sound.   A detailed exploded view of the midrange is shown below.

 

The horn loaded midrange driver raw sensitivity is almost 110 dB allowing for minimal excursion at normal playback volume.  When cone excursion is reduced, distortion is also reduced.  The midrange is direct, detailed and clear without being forward or strident.  High efficiency and a low moving mass are the best ingredients for maximum transient attack.  It’s quite an achievement and one that will provide a great window into the subtle details of music.  

Similar to the woofer the 4 ½” midrange follows the same design philosophy and is driven by a group of 3 high temperature N35SH neodymium magnets.  As with the woofer this motor design maximizes magnetic field strength while optimizing the symmetry of the magnetic field surrounding the voice coil, ensuring linearity during high dynamic transients.  The high strength magnets also allow for a minimum profile behind cone, reducing any unwanted reflections. 

The midrange driver’s pole piece is topped by a copper cap whose purpose is essentially the same as the faraday rings used in the woofers.  The end results also the same, reduced distortion and greater dynamic capabilities.

The midrange unit is housed in its own, acoustically tuned, sealed enclosure within the larger cabinet; this enclosure also serves to magnetically shield the unit.  Furthermore, this sub-enclosure isolates the midrange driver from acoustic energy generated by the woofers.  Every effort has been made to ensure great accuracy from this unique driver.  From the choice of voice coil formers and coil wire to the mechanical design of the speaker frames; every part is evaluated not only for accuracy but also for long-term durability and the ability to produce all the information in the source. The Klipsch P-39F will accurately deliver all micro and macro dynamics found on today’s most advanced high definition media.

Transducer Design

     Low Frequency Driver (Woofer)

 LF Driver Features
   
 • 1 ½” Diameter – Flat-wire “low mass” aluminum voice coil 
 • 0.72” (18mm) linear peak-to-peak high excursion design
 • High linearity magnet structure containing 3 different Neodymium magnets
 • High Temperature N35SH Neodymium magnets
 • Two Faraday aluminum shorting rings – flux stabilizing rings on each side of the gap
 • Reverse half-roll low density foamed rubber surrounds
 • High air-flow – low restriction die-cast aluminum basket
 • Hybrid Aluminum/ROHACELL® cone

The P-39F uses three 9” (225mm) diameter low frequency drivers.  This custom woofer utilizes a low mass hybrid aluminum/ROHACELL®/Kevlar  cone with an over hung voice coil design, ensuring there is always an equal amount of coil winding in the magnetic gap regardless of the excursion of the driver.  Even during high excursion, the driver remains linear and low in distortion.  A cut-away of the woofer is shown below.

 

The cone is constructed of an outer aluminum skin and inner Kevlar skin bonded to a lightweight rigid core of ROHACELL®. This material is commonly used in high performance vehicles and aircrafts.  This unique closed cell polymethacrylimide (PMI) foamed plastic core adds increased rigidity to the cone while minimizing ringing and maintaining low mass.  The sandwich constructions results in an extremely stiff but light cone allowing the woofer to operate as an ideal piston throughout the woofers pass band.  This translates into much less coloration giving an incredibly natural and realistic sound.  A three part high intensity neodymium N35SH magnet design is used in the 9” woofer with a main magnet plus two supplementary magnets, placed above and below the main magnet, to ensure linearity, reduce stray magnetic energy, and provide intense field strength in the voice coil gap.  The field strength of the neodymium magnet structure has an equivalent magnetic field strength of an 80 ounce ceramic magnet, with only a combined weight of 13 ounces.  A magnetic Finite Element Analysis (FEA) of the motor structure is show to the right.   Additionally, dual aluminum Faraday rings are also located inside the motor structure and on the pole piece.  Faraday rings, also known as shorting rings, are used for four reasons:

1. Minimize unwanted inductance. 
2. Minimize inductance change with voice coil position. 
3. Reduce flux modulation caused by the magnetic field generated by the voice coil. 
4. Increase heat dissipation. 

These all result in reduced distortion and compression and lead to improved dynamic capabilities.  The effect of the aluminum rings can be seen in the graph below sowing the variation in electrical inductance with voice coil position.A superior motor structure is complemented by the design of an efficient voice coil.  The 1 ½” voice coil is wound with a flat-profile low-mass copper clad aluminum wire.  Flat-wire allows for efficient packing of the wire when layered on the voice coil resulting in an increase in the Bl.  The Bl is the product of the magnetic flux density (B) and the length of the coil in the magnetic gap (l). This increased Bl directly correlates to an increase in the sensitivity of the woofer.  The flat wire voice coil/motor structure provides 0.72” (18 mm) peak to peak linear excursion.  Below is a graph showing the difference in BL(X) with and without the additional bucking magnets in the motor
 

Another unique feature about this woofer is the surround.  The surround is made of half-roll low density foamed rubber and inverted to allow for minimum diffraction.  The low density foamed rubber helps to reduce the moving mass while maintaining surround integrity under high excursion and box pressures.  The flat-sided shape was created to maximize the linear excursion while increasing the radiating area.  Special care has also been taken in the design of the driver suspension which, along with the surround supplies the restoring force to keep the driver moving linearly about the rest position.  It is important that the restoring force is symmetrical in the forward and backward motion of the woofer cone. The graph shows the measured stiffness vs displacement for the P39-F 9”woofer compared to an inferior design.

The cast aluminum woofer frame made specifically for this model employs narrow spokes which are grooved to provide maximum surface area in their minimal profile. This assures maximum heat transfer without any reflection of the back wave of the driver that would otherwise color the sound.  A Finite Element Analysis (FEA) of the heat transfer of the aluminum basket is shown below.  The aluminum frame is designed to acct as a heat sink for the woofer motor, conducting heat generated by the voice coil away and thus reducing the effects of thermal compression, translating into greater dynamic capabilities.

Enclosure Design    

Enclosure Features           

• Non-parallel wall construction
• 1.06” (27mm) thick side wall – 1.2” (30mm) front baffle
• Triple Layer Laminated medium density fiberboard (MDF) and particle board
• Mid/High Horns combined into single bulk molding compound (BMC) structure
• Three flared rear facing port tubes
• Internal volume of the cabinet 4.2 ft3 (120L)
• Interlocking vertical and horizontal support braces
• Ultra high quality custom machined binding posts
• Optimized separate midrange enclosure
• All transducers are flush mounted for minimal baffle deflection
• Separate high mass machined base coupled to enclosure
• Four adjustable floor anchors for improved cabinet decoupling

When you first see the P-39F, the beauty of the cabinetry is apparent. The industrial design of the enclosure was guided by Design Works, a division of BMW Automotive.  The P-39F is an artistic blend of beauty and functional performance.  Functionally, the loudspeaker enclosure provides a mechanical placement of each transducer, a method for suppressing acoustical radiations, and a resonant chamber to extend low frequency performance. 

Enclosures can also degrade the sound quality if they are not properly designed.  Some of the most common problems are unwanted cabinet resonances. Enclosure resonances can be suppressed through the geometry of the enclosure and mass of the panels. This is why the P-39F enclosure contains nonparallel wall structures and thick, high mass walls.   Nonparallel wall structures significantly decrease standing waves, which naturally occur because of constructive interference in the waveform. At the node of a standing wave, large sound pressure levels are generated potentially creating the enclosure walls to resonant uncontrollably. Cabinet wall thicknesses are a minimum of 1” thick with an even thicker front panel for a resonance-free platform to mount the drivers.  Inner and outer laminated panels sandwich a middle layer of dissimilar material designed with a very different mechanical impedance to ensure the cabinet does not color the sound produced by the drivers. 
Each of the curved cabinet side panels are comprised of multiple laminations of MDF and particle board that are formed in a proprietary process.  A sample of the cross section of wall is shown below.

To further increase enclosure stiffness four “H” braces are strategically positioned in the enclosure to further reinforce the structure and control resonance. The inner surface of the panels therefore minimizes modal standing waves.  Another feature is the high mass front panel key for the transducer mounting.  The front baffle has also been designed to allow all woofers and the mid/high horn structure to mount flush the exterior edge.  The width of the front baffle has also been minimized to decrease the total area.  The overall enclosure was designed to extend the low frequency response of the three woofers to 39Hz (-3dB).  This fourth-order vented enclosure contains three custom flared ports each of which is cleverly blended into the rear of the curved enclosure.  The angle of the port flare helps to minimize the affect of standing waves developed inside the port tube. 

Just below the ports, the high mass machined base is integrated into the rest of the enclosure.  The base contains four adjustable floor spikes which anchor the speaker to all surfaces and also allow the speaker to be angled for fine tuning of the stereo sound stage.  The base of the speaker also contains three sets of custom machined binding posts allowing the end user to have a superior connection.  The front of the enclosure also contains integrated high intensity magnets allowing the two piece grill to seamlessly attach to the front of the enclosure.  A detailed cutaway of the enclosure is depicted below.

Notes from the past...  600 Acoustic Watts!

 

Gentlemen:

 

This could be a photo of one of a number of installations dating from the heyday of California Aerospace in the sixties and seventies. Fancher Murray may be able to shed some light on this, inasmuch as he described something similar to me that had been built in El Segundo, CA, for Wyle Laboratories when he was working there.

 

It looks like about 80 375s and 40 LFs. Assuming an input signal of, way, 25 average watts per HF driver with an average efficiency of 30 percent, we would have 25 x 0.30 x 80, or about 600 acoustic watts. The actual level would depend on the size and treatment of the receiving space. (Do you all recall that the acoustical impedance of a highly reverberant space is quite different from that of a free field?)

 

John Eargle

On Aug 9, 2005, at 3:23 PM, Don McRitchie wrote:

 

I can add that I know there was such a testing setup at the White Sands Missile Range in New Mexico that was recently decommissioned. Around four years ago, the drivers from that setup began showing up on Ebay. The HF drivers were labeled 375H and “Manufactured by JBL for Transducers Inc.” The LF drivers were labeled 150H with the same Transducers Inc. brand. Transducers Inc. was a private company that was wholly owned by Bill Thomas (the then owner of JBL) as separate company that built sonic and vibration simulation equipment for the military. Someone from JBL of that era (it may have been Harvey Gerst) recently told me that the 150H was nothing more than a 130A chassis with an LE15A cone kit (for the reason’s Mark cited). The drivers were all painted the same blue colour as the original LE series drivers which would date them from the early 60’s. It was comical to see them advertised on Ebay as rare hi-fi drivers. Some auctions even alluded to the 150H as being the driver from the Hartsfield, confusing it with the 150-4C. I’m not sure if there were any physical differences in the 375H compared to the standard 375. The diaphragm was still aluminum and not the later high power phrenolic version. However, I can’t imagine an LE15A cone kit in a 130A chassis having a “hi-fi” response. BTW Mark, if you ever come across photos like this in the archive again, let me know. I don’t remember seeing anything like this when we rummaged through it.

 

 

Don

 

 

 

From: Gander, Mark

Sent: Tuesday, August 09, 2005 3:06 PM

Subject: RE: Any idea what this is?

 

 

 

This was an acoustic and vibration testing set-up at one of the aerospace testing companies in the 1960s.  I remember this and other similar photos from the JBL archives.  JBL 375 compression drivers on the rectangular mouth horn that was used with the HL89 serpentine lens, and probably 130A woofers, judging by the pot and frame (same as a D130, but the light cone and coil of a D130 couldn’t take this level of abuse), though the driver may have been a special hybrid – notice the white Lansaloy(Rubatex) surrounds.  All to generate 150+ dB sound fields to simulate rocket launch levels of sound and vibration.

... Mark Gander as you may know is the VP of Marketing at JBL.