DIY Axially symmetric oblate spheroid CD waveguides, in solid Oak

[jack_bouska]

I posted some images of my recent waveguide construction on the "Horn system pictures" thread, <http://www.audioheritage.org/vbulletin/showpost.php?p=122818&postcount=49 (http://www.audioheritage.org/vbulletin/showpost.php?p=122818&postcount=49)>, and in post #51 "Titanium Dome" followed my post with a request for the & measurements I made on my waveguides.
Rather than clutter up the "Horn system pictures" thread, I have chosen to start a new thread under the DIY section, with more details of the design and images.
The two new waveguides are mounted on a pair of 1" exit TAD 2002 compression driver, and a 2" (49mm actually) exit a pair of JBL 2441 compression driver.
I purchased the TAD new, however the JBL 2441 originally saw duty as part of the sound reinforcement system of the Calgary Saddle dome, home of the 1988 Olympic games <http://www.pengrowthsaddledome.com/foundmain.html (http://www.pengrowthsaddledome.com/foundmain.html)>. It may be unusual to use a device which once shared about 1/15 of the coverage duty for a 20,000 seat stadium in my modest sitting lounge, but it should be no surprise to this membership that achieving realistic dynamic range with low distortion does require some extreme measures to achieve our acoustic goals
The full list transducers in the speaker system consist of (refer to picture 4 in this post)
2 x TAD 2002 (4khz - 20khz)
2 x TAD 2441 (1khz-4khz)
4 x JBL 2123 10" cone midrange (250hz - 1khz)
4 x JBL 1401nd 14" cone upper bass (60hz - 250hz)
2 x Altec 3182 18" cone sub bass (7hz-60hz)
The waveguides were designed using information from published papers by Earl Geddes. (reprints can be found at: <http://baseportal.de/cgi-bin/baseportal.pl?htx=/Data/exdreamaudio/download (http://baseportal.de/cgi-bin/baseportal.pl?htx=/Data/exdreamaudio/download)> )

Images and measurments graphs to follow:

Jack Bouska

[jack_bouska]

The first four pictures, in this post illustrate the Oak waveguides during construction, mounting, and in their final location in my listening room. The first images shows the "green" untreated French Oak on the faceplate lathe partway through turning. Note the blue plastic template used to guide the turning operation. I nicknamed the horn flare "the rams head horn" for obvious reasons.
The second image shows how the horns are mounted to a retaining flange which forms part of the brackets which retain the compression drivers. The horns are not bolted directly to the compression drivers. BluTak is used as a flexible gasket to ensure a smooth transition from waveguide throat to driver exit.
The third image shows a front perspective close up of the mid-high section of the loudspeaker.
The fourth image shows a full view of the components, left to right: TX columns contain an Altec 18" in 4 meter folded transmission line cabinet, centre: "stack of loudspeakers" and on the right, the equipment rack with CD, 2x Behringer DCX2496 and the 5kW Tripath power amplifier "plate"
The full system can play peaks well in excess of 130dB, over a power range of 10hz to >20khz
I nicknamed the loudspeaker system "IMPACT STACK"
Jack Bouska

[jack_bouska]

The first two pictures in this post show the waveguide design profile for the 2" throat, and the 1" throat compression drivers respectively. These images can be downloaded, scaled (to match the dimensions listed on the diagrams), and printed to produce templates should anyone wish to construct a copy of these specific waveguides.
Image 1: The 1" exit waveguide is stock Oblate spheroid shape, with a 1/2 round radius at the mouth The 2" exit waveguide is a compound shape, starting with oblate spheroid from mouth to 3/4 horn length, merging (seamlessly) into a tractricx taper for the last 1/4 of the waveguide, to the mouth, finishing in a 1/4 round radius curve to the outside edge.
When I worked up a waveguide design which followed E. Geddes recommendation, to terminate the mouth with a radius edge equal to 1/4 wavelength at lower frequency. limit, this resulted in a radius of over 3" for 1khz, which meant that for a full 1/2 round mouth, I would need an annular ring 6" wide all the way around the mouth.
Image 2: My oblate spheroid mouth is 5" in diameter, which would have made the whole horn 17" in diameter, much too big. Instead, the Tractrics horn mouth termination (which goes from 40deg curving around to 90deg) is only about 1 1/2 inches wide, and I add a 1/4 round radius just to smooth the edge. This makes the whole diameter much more tractable at less than 10" diameter
The last three images show the directivity response of a 2" throat tractrix horn, followed by the 2" waveguide, and finally the 1" waveguide (mounted on the JBL 2441, and TAD 2002 respectively). The measurements were made with a Panasonic mic element, Altec microphone preamp, SoundBlaster Audigy ZS 24/96 pcmcia sound card, IBM laptop and ETF5 software. The measurements were made at low SPL (~88dB) at a distance of 50cm from the mouth.
Image 3: The spl response vs. angle for the Tractrix horn is seen to "fan out" fairly rapidly from about 2.5khz upward, with the curves widening as frequency increases. Only the 90 deg (on axis) curve stays close to flat. (These curves were measured without using any horn compensation) The curves are annotated as angle away from horn axis. This angular coverage is about the same as I would expect from a cone of approximately the same diameter. (not CD)
Images 4-5: Both the 2441 and TAD curves illustrate good examples of constant directivity using the axially symmetric oblate spheroid waveguides invented by Earl Geddes. The spl response curves cluster closely (within a few dB) over the full +/- 40 degree design coverage. The curves are annotated as angle away from the plane of the waveguide mouth (so 90deg is directly on axis). Both these devices have response equalization applied using a Behringer DCX2496 digital crossover.
The spl response curves which are outside this angular coverage (yellow, black and red curves) are seen to drop in amplitude very rapidly with rising frequency, which indicates that the CD waveguide "does what it says on the tin"
Image 4: The 2" throat device on the 2441 is good from about 1khz up to 10khz (where the aluminium diaphragm break-up starts), and the phase plug no longer produces nice planar waves above that frequency.
Image 5: The 1" throat device on the TAD appears to be good from 1khz up to beyond 20khz. No evidence of diaphragm break-up is visible. (Beryllium, the right material for this job)
Jack Bouska

[jack_bouska]

The first four pictures in this post show the uncorrected, and EQ'd-corrected frequency response of the two waveguides, using white noise, averaged over a 4ft x 4ft x 2ft volume at/around the listening position, followed by Fourier transform spectral analysis averaged over about 45 seconds of recorded noise waveform. Use of spatial averaging smoothes through the short period room modes (peaks and nulls), but highlights the response anomalies which are inherent to the device under test. This form of measurement is a balance between the direct arrival energy, and the reverberant sound field in the listening room. (as opposed to the previous ETF5 measurements, which are intentionally restricted to direct arrival only, via close mike placement, and short window on the FFT transform).
The volume of averaging is around the defined listening position, and stays well within the +/- 40deg coverage angles. Both left and right channels are run simultaneously, with uncorrelated white noise between channels. Amplitude is less than 90dB, distance approximately 7-11ft from each channel.
Note that the compression driver/waveguide combinations are not useable without response equalization applied using a Behringer DCX2496 digital crossover) shown annotated on the graphs. The waveguide CD devices do not have the characteristic on axis high frequency boost noted with exponential or tractrix horns.
Image 1: shows the response of the JBL 2441 2" throat drivers on the waveguide, using spatially averaged white noise.
Image 2: shows the response with final EQ applied. I chose to use a parametric EQ (Band pass) centred on 2.62khz with a Q of 0.4, level -8dB cut.
Note the very low level of ripple on both these graphs (around +/- 1db worst case). This is a very good indication of the amount of internal (axial) reflections within the waveguide and coupling throat of the JBL. Poor acoustic impedance matching at the mouth can cause reflections which send energy backwards in the waveguide to be reflected at the phase plug, and generate resonances. These axial modes generally exhibit themselves as the classic "horn honk" and sound similar to the distortion you can hear when speaking through the cardboard tube from kitchen food wrap (or toilet paper tubes). The resonance modes (reflections) are generally visible as ripple in the response of most horns, and a smooth curve is evidence of low levels of reflected energy, and high quality tone. My choice of tractrix mouth instead of 1/2 round radius was apparently a wise one, as borne out by these graphs. The sound is indeed very natural and clear. To quote my audiophile friend on his first listen: " I have never heard horns sound like real music before this!"
Image 3: shows the response of the TAD 2002 1" throat drivers on the waveguide, using spatially averaged white noise.
Image 4: shows the response with final EQ applied. I chose to use a parametric EQ (Band pass) centred on 4.238khz with a Q of 0.5, level -7dB cut, cascaded with a HighPass shelving filter set at 20khz, +15dB, single pole 6dB/oct.
Note the modest increase in the level of ripple on the TAD response (now +/- 2dB). I attribute this to two possibilities: 1) the use of a small 1/2 round radius on the horn mouth which is intended only to operate above 4khz (the crossover frequency for this device) and 2) reflections in the phase plug and throat of the device itself. I have noted these same ripples, at the same frequency locations, on a number of horns, of different taper and length, when used with this same compression driver.
When the crossover is applied, the same graphs show marked reduction of ripple below 4khz (in the crossover band), which indicates that offending reflections (poor mouth termination) can be suppressed by restricting the bandwidth to the range appropriate for the round over at the mouth (1/4 lambda = r)
Image 5: shows the pair of impulse response graphs, using the 90deg measurement from ETF. In the interest of time (I have to go now) I will describe these two graphs in the next post, later this week.
For now the above graphs give a fair overview of the performance of these devices, and I promise more information to follow shortly
Jack Bouska

[jack_bouska]

The last image in the preceding post (#4 above) showed the pair of impulse response graphs for the compression drivers on waveguides (JBL top, TAD bottom). The amplitude scale is linear (+/- 32k, 16 bit wave file), and the time scale is in samples @ 48khz sample rate. The display window is 120 samples long, or 2.5ms. The energy on both devices is seen to decay by 99% in the first 1/2 millisecond, and further attenuate to approximately 99.9% within the first full millisecond. Both wave shapes exhibit a classic high-pass differentiation filter shape (prominent peak-trough), followed by an exponentially decaying tail of band limited lower frequencies. The impulse response of both devices also display a 2nd negative pulse following the first trough. I suspect this is a function of the digital EQ applied via the Behringer DCX2496 crossover, and I intend to investigate by running some "raw" measurements, driving the compression drivers directly from the soundcard/amp with no equalization applied.

The first image in this post shows the ETF5 cumulative spectral display, in linear scale format. The spectral plot serves two useful diagnostic purposes: first it illustrates the decay in energy with time, (similar to a waterfall), by using four overlapped FFT graphs (each 2.6ms long), with start times of 0,1,2,3ms respectively. Second, the linear scale spectral graph can be used to identify/analyze internal mouth-throat-phase plug resonances (axial modes), which cause response ripples due to comb filtering of the reverberation pattern.

The 2441 graph (top) shows that between 1.5khz and 10khz, the energy from 1-3ms is on average about 20db lower than the first arrival, which implies very little energy storage within the device or diaphragm. Much of the measured energy beyond 1ms is likely reflected from surfaces near the microphone (floor, walls, furniture), while some of it is related to the unavoidable effect of abrupt horn mouth termination, and the acoustic impedance contrast associated with the rapid change in coverage angle from +/- 40deg to full space (+/- 180deg). At the rim of the waveguide (or any ordinary horn), the acoustic impedance contrast generates diffractions, sending some energy backwards within the waveguide from the mouth towards the throat. This is reflected again from the position of the phase plug (another cause of acoustic impedance contrast within the compression driver itself), and the affair sets up a series of decaying "normal mode" reverberation, with reverberation time (period) roughly equal to the length of the horn (plus compression driver coupling throat), divided by speed of sound. (~ 35cm/ms)

In their article: "Round The Horn" Philip Newell and Keith Holland, (Speaker Builder, 8/94) suggest that much of the classic "horn honk" is attributable to these mouth-throat resonances. For a practical demonstration of how important good mouth termination is for neutral tonality, simply take your favourite magazine, roll it up into a conical shape "megaphone" hold it up to your mouth and clearly utter the phrase: "this is the sound of horn honk". (and you will be speaking the truth). Horn honk can be easily detected by your ears as acoustic resonance and emphasis of some frequencies, along with recognition of the delayed energy caused by the trapped waves. Bad horns have a great deal, and good horns have almost no detectable evidence. The axially symmetric horn tested by Keith Holland was documented as sounding like a quad electrostatic by the listening panel.

These resonances take the form of delayed signals, which are added back into the acoustic output (with alternating polarity). The delay+addition of signal creates a comb filtering effect, which on a log scale amplitude plot appears as a sinusoidal ripple in the (otherwise smooth) response. The period of this sinusoidal pattern is related to the time constant of the reverberation according to the equations: time period = 1 / frequency (as measured peak to peak on a linear frequency scale spectral graph)
Inspecting the first image of this post shows the most pronounced ripple to have a peak to peak span of about 400-500hz, which relates to 1/400 = .0025s = 2.5ms or around 87cm, which corresponds to the distance from the horn to the floor, and back to the microphone. Intra-waveguide reflections would need to be less than the 20cm (x2) length along the waveguide axis. (40cm / 35cm/ms = 1.14ms, and 1/(.00114) = 877Hz) This corresponds to ripples with duration of 900hz or longer. Evidence of these periods are difficult to discern on the blue spectral graph, but are noticeable on the later spectral lines, with some periodic notches noticed at 1-2khz. This would imply some reverberation between the throat and mouth of the wooden waveguide section (10cm) or more likely, between the phase plug and the transition to the wooden waveguide section (~6-8cm). The exit point from the compression drivers to the waveguide acts as a circular diffraction aperture, such that the near planar wave front emerging from the compression driver effectively diffracts into a spherical wave front, to ultimately fill the conical expansion of the oblate spheroid contour.

This ripple (900-2khz period) is very low amplitude, perhaps 1-2dB out of the total 40dB on the vertical axis. This effect is not audible by me, as I rate the waveguides as having near-zero horn honk characteristic.
Incidentally, the intra-phase plug resonances would generate ripple with a 10kHz period on these graphs, and indeed there does appear to be some visible on these graphs, however I would need to do more work to determine if this is truly a phase plug problem, or if it has been induced by my heavy handed equalization scheme.

The second image in this post shows the measured phase response of the waveguides, (blue line) compared with the calculated minimum phase for a filter with the same frequency response (using Hilbert transform). Ignore the vertical lines on the top plot, where the measured phase slightly exceeds the +/- 180deg vertical scale (phase wrapping).

The third image shows the 1/3 octave smoothed, on-axis, frequency response of the waveguides, with a gating of ~10ms. The close mike position, coupled with the short window, mean that these graphs correspond to the direct arrival "anechoic" frequency response (although some pesky reflections do affect the response accuracy). Full equalization has been applied, with the lower frequency of the crossover set to 350hz

The fourth image of this post contains the waterfall plots for the waveguides, and are included mainly as "eye candy". Apart from the obvious quality of the rapid energy decay with time, also note that the devices are relatively resonance free (apart from the common bump and ridge at 1khz). The ridges down at the noise floor are largely influenced by early reflections from floor and surrounding objects. However one important feature of note is visible on the JBL 2441 graphs at approximately 10khz. A dip in the frequency response (T=0) rapidly turns into a non-decaying ridge along the time axis. This is a clear indication of a diaphragm break-up mode, and is audible when the 2441+waveguide is run full band (1-20kHz). For frequencies below 10kHz the performance is exemplary, and rivals (or perhaps even betters) the performance of the TAD with it's modern phase plug and exotic Beryllium diaphragm. (Note, the JBL 2441 has 1dB more sensitivity, than the TAD 2002, and can handle 9dB more power input-watts).

The fifth image in this post (and the last in this series) shows the ETF 1/3 octave smoothed (direct arrival weighted) frequency response, compared to the spatially averaged, white noise excitation, spectral estimate (reverberant field weighted). The frequency response is wide and smooth, but more importantly, the direct arrival response closely matches the reverberant field response, which is a direct consequence of employing the constant directivity oblate spheroid waveguides. This allows me to choose a single set of response EQ curves which are appropriate for both the on-axis direct arrival wave front, and also appropriate for the reverberant portion of the wave field in the room. The CD waveguides permit the generation of a smooth 360deg (solid angle) power response, when the speakers are placed in the corners of my listening room.
Jack Bouska

[Titanium Dome]

Jack

That's a plethora of information. Thank you.

You've clearly been working and thinking hard on these units. :applaud:

I see some room treatments there. Comments?

[jack_bouska]

Jack

That's a plethora of information. Thank you.

You've clearly been working and thinking hard on these units. :applaud:

I see some room treatments there. Comments?

Mr Dome (can I call you Titanium?)

Actually, the room has not had any specialized room treatment at all. It is furnished normally, with both soft chairs, and large leather sofa's. The floor is carpeted, and the walls and ceiling are untreated other than normal hanging pictures. The room contains no curtains. Also rather interestingly, the entire construction is cement block, with poured cement floor and ceiling. The room modes are well distributed, but very pronounced. The typical RT60 in the room measures less than 0.4 seconds for frequencies above 120hz. Below 120hz the RT60 increases to a time of 1.2 seconds, owing to the massive cement container construction (dimensions: 18.5 x 14 x 8.1 ft)

I use high Q notch filters to attenuate the worst of the axial modes.
I do need to add damping in the low frequencies, and intend to used large diameter rigid fibreglass pipe insulation as an aperodic absorber, placed upright in the corners of the room, just have not gotten around to this yet. In the mean time, the only low frequency absorption comes in the form of the leather sofa.

When you mention room treatment, I suspect you are referring to the faux-marble columns visible behind the main speaker stack. (in the corners)

These are actually 4 meter long, folded transmission line loudspeakers, loaded with one 18" Altec each.
Pictures of the driver are available in my image gallery on this Forum:
http://www.audioheritage.org/photopost/showgallery.php?cat=500&ppuser=844 (http://www.audioheritage.org/photopost/showgallery.php?cat=500&ppuser=844)
Details of the construction are shown below in the following three images.
Jack Bouska

[Robh3606]

Nice job Jack!!

So does it sound as nice as it looks??

Rob:)

[yggdrasil]

Fantastic. You must certainly have spent a lot of time/fun planning and finishing off with very skilled craftsmanship.

Do you have original DCX2496? There was an article in audioXpress this summer about replacing the analog section completely.

[morbo!]

Thats totally friggin amazing

I always wanted cyndricral spreaker`s

now i got a new project
gonna take me awhile to save up for 18" drivers
but i really wanna feel that bass


thanks again to this forum
and it`s crazy but well balanced members

some people get a rush jumping out planes
personally i cant im imagine the rush of doing that from scratch


lets hope the cash gods smile on me soon

[jack_bouska]

Nice job Jack!!

So does it sound as nice as it looks??

Rob:)

Rob: (ok you asked for it, part I)

My simplest answer could be: "they sound much better than they look!", but that statement does not tell you very much.

In a previous post, I elaborated a little more by saying: "The sound is very natural, and well recorded live instruments sound convincingly very real."

But to be honest, these are only subjective comments, which need to be taken with a grain of salt, because my own enthusiasm could easily be explainable as a consequence of Beranek's law:

"It has been remarked that if one selects his own components, builds his own enclosure, and is convinced he has made a wise choice of design, then his own loudspeaker sounds better to him than does anyone else's loudspeaker. In this case, the frequency response of the loudspeaker seems to play only a minor part in forming a person's opinion." - L.L. Beranek, Acoustics (McGraw-Hill, New York, 1954), p.208.

However in my own defence, at least I have been overly generous with providing a comprehensive set of frequency response graphs & measurements to help verify my claims. No doubt you have reviewed all the images above, but you still need to ask if the system sounds as good as it looks, because frequency graphs don't tell the whole story. On the other hand, I have never heard a good system that I measured to have a bad frequency response, so flat, wide, bandwidth, with good phase and rapid energy decay is the mandatory entry level for any system which claims to be world class.

It's probably best if I describe the system design, and explain the engineering behind why various aspects sound as they do, while I proceed through the list of sonic qualities. As it turns out, I tend to rebuild my speaker system about once every decade. Two years ago, after obtaining four used JBL 1401nd drivers, I spent just over a year (in my spare time) designing this system, and another year (again, in my spare time) constructing it. The system is mostly complete now, so I will likely spend another year writing about it. (here on this forum, then on my own web page, and eventually a full design/construction article for AudioXpress).

The original system goal was to achieve a wide, flat, frequency response, with low distortion, and wide dynamic range, including high SPL capability, with low acoustical, mechanical, and electrical noise floor. These design criteria match or exceed those required in a good studio monitor (adapted from my website):

1) Wide bandwidth (starting below 10hz extending above 20khz)
2) Realistically wide dynamic range (at least 130db SPL peaks, undistorted)
3) Flat bandwidth with smooth anechoic on axis response (max rate of change < 2db between 1/2 octave bands)
4) Flat bandwidth with even distribution of full field power response at listening location (Smooth white / pink noise spectrum, power response uniform in horizontal plane below 800Hz, narrowing to horizontal&vertical +/- 40 deg above 1kHz, all +/- 3dB)
5) Low distortion (Harmonic and IM < 0.1% midband)
6) Less than 1 db power compression, all amplitudes up to 125db SPL
7) Stable stereo imaging (via controlled acoustic dispersion and absence of early reflections)
8) Natural sounding speech and music (freedom from any identifiable timber or tonality.)
9) A system which is compact enough to fit in an average living room.

The system has met most of these design goals, with 11 octave bandwidth, +/- 3dB over more than 9 of those octaves, utilizing drivers known to have low distortion, and a peak output SPL of more than 130dB per channel (courtesy the 5kW amplifier rack driving them).

Before describing the sound as a whole, I will expand a little by discussing each of the constituent components in turn and describe some of the technology and design choices underlying the transducers and cabinets in each frequency band.

Starting at the bottom end, I can say sincerely that this system can produce a level of deep bass extension and fidelity that 99.9% audiophiles simply have never experienced in any home setting.

With proper equalization, the -3dB point of the 18" Altec transmission line starts at 7 Hz, and ignoring room modes, is flat up to over 1khz. In practice, the Altec's are crossed over at 60hz low pass with a, 6dB/octave slope. This is well beyond scope of normal commercial HiFi, and in all my visits to high-end shops, and Hi-Fi shows (in London), I have never heard anything to compare. (Some systems go deep, but not loud enough, some visa versa, and most are hopelessly distorted at anything approaching 110dB). The deep bass from my transmission line devices is prodigious (each Altec woofer can produce continuous spl levels of 113dB over the 10hz to 60hz band), with very low distortion. This combination works to lay the very foundation for rock and pop music.

Suffice to say that this level of power can easily cope with anyone's desire for chest thumping, furniture shaking and window rattling from any collection of Rock, Pop or techno dance disks. But there is much more to it than that. The system can produce uncanny ambience effects which impart a true sense of the venue-hall size and volume.

The quality of the deep bass (sub 30Hz) makes it possible to deduce if instruments, such as bass and drums, were recorded in a small basement studio, or in a large 60 foot x 40 foot commercial recording hall. The ambience and reverb from other instruments (organ, tuba, double bass, etc) also can impart a sense of the space that the recording took place in. For most systems (that are limited to upwards of 30-40Hz) the spatial cues come from wall reflections higher in the range, but when the same music is played on a system, such as this, that has true deep bass fidelity, the low frequency information adds an extra dimension to the experience.

This extra bandwidth can be a double edged sword, as these speakers are readily adept at revealing a host of mastering and recording problems (thumps, HVAC rumbles, traffic noise, and wind pops from close mic'd strings) which made it onto the final CD, without the recording engineer ever being aware of the problem. Of course with the Behringer DCX2496 I can add a low cut filter with the push of a button, however these anomalies are so rare on the music I listen to, I just leave the bandwidth as wide as possible. Besides, I get a kick out of being able to reproduce even more low bass on modern electronically generated dance music than the dj/recording engineers who composed the pieces could hear themselves when they laid down the tracks.

Everyone who auditions my system reports being duly impressed by the quality, fidelity, and sheer impact of the transmission line bass cabinets. One commercial manufacturer (in the U.K.) has actually started prototyping their own version (using 15" drivers) following a listening session at my home. A fellow club member of the LondonLiveDIY HiFi Circle made the following comments after a home meeting:

"No trip to Jacks can pass without mention of his system's ability to reproduce low frequencies. This was my first experience of his system and I was of course impressed with the lf capabilities. Where there are low frequencies the artist/musician intended this is a real treat. However, many tracks seem to have a good number of thumps and bumps that were clearly not intended. Jack says this is all detail that made it through mastering without being noted because the studio monitors were not going low enough.

Whatever the explanation, I found it a distraction on some tracks. However, as one who enjoys the sensation of low bass it is certainly good to experience a system that can provide it and I am very glad I went on Saturday.

Trouble is it leaves one thinking about building speakers!"

End of part I
Jack Bouska


Jack Bouska

[jack_bouska]

Nice job Jack!!

So does it sound as nice as it looks??

Rob:)

( ok, you asked for it, Part II)
Moving up the frequency scale to the next band, are the four JBL 1401nd 14" transducers each in its own separate 22 litre cylindrical cabinet (see my avatar), crossed over at 60Hz and 243hz, 6dB/oct. It turns out that about 40-50% of the total radiated acoustic power is contained within this two octave band, so I can't possibly over-emphasize the importance that these drivers play in the total system performance. (These drivers do most of the proverbial "heavy lifting")

The JBL 1401nd driver was first used in JBL K2 high-end system and later in the formidable JBL DMS-1 large format studio monitors. The exceptional qualities of the sound of these two speaker systems has been described numerous times elsewhere on these forums, so I probably don't need to elaborate too much on the superb qualities of this particular driver. It is one of Greg Timbers favourite woofers, and he has several in his own home system. (If they are good enough for Greg T ……)

Given the exemplary pedigree of this driver, the remaining questions are related to specific implementation in a given system. Both the K2-S9500 (and M9500), as well as the DMS-1 systems employed the 1401's in large ported enclosures. These ported enclosures helped stretch the frequency response to about 30Hz on the low end, however in my experience the sound of bass from a port or vent is never quite as good as that obtainable directly from the cone, when the driver is in a sealed enclosure.

Part of this is because the area of the port is generally smaller than that of the cone, and part of the problem is because of the inherent energy storage in the Helmholtz resonator arrangement of a ported enclosure.

Sealed enclosures are also less sensitive to box tuning, so driver to driver variations are minimized. The difficulty in using the 1401's in a sealed box is that the Vas, and Qt are both relatively low, which means that the volume of the enclosure must be made quite small to prevent the system Q from being over-damped. With a volume of 22 litres, I get a -3dB point of around 80Hz, and when stuffed with long hair wool, the cabinets have a system Q of between 0.5 and 0.6 (near Bessel alignment). I use a modest amount of equalization to make the drivers flat down to about 30Hz (in room) prior to application of the 6dB crossover at 60Hz. Although this design means the drivers are slightly less efficient than when used in a ported enclosure, I find that this setup results in the best transient response possible, and the best overall tone and scale.

The conscious decision to utilize small, sealed enclosures in a five way system is appropriate given the incredulous foundation of bottom end afforded by the use of the 18" Altec transmission lines. This freedom from the requirement to produce the lowest frequencies opens up a host of other design benefits from the use of small volume enclosures, namely:

A) A smaller overall system footprint which fits better in residential sized listening rooms.

B) Each individual cabinet (and total system) weight is lower than for an equivalent rectangular box (in MDF or plywood).

C) The small front baffle size aids close spacing and better integration among various drivers in the system.

D) The total surface area of each small cylindrical cabinet is much less than a large rectangular box, resulting in a lower ratio of enclosure-to-driver surface area, which reduces the radiation of unwanted cabinet vibration.

E) Cylindrical cabinet construction is very strong and stiff in the axial direction, so the whole cabinet works as true reaction mass acting against cone acceleration forces, resulting in overall less stored energy in system.

F) Each cabinet is compliantly mounted (using sorbothane), which reduces the level of mechanical coupling, and vibration transmission to the other drivers in the system, as well as assisting with reaction mass functionality (in E).

G) Any residual vibration in the cylindrical walls will be in phase for antipodal points (180deg), but will be out of phase at the points +/- 90deg around the circumference (typical of bell modes). For lower frequencies, in the far field the interference among these out of phase waveforms will cancel, reducing the already low level of radiated vibration.

H) The dimensions of the cabinet are adjusted to be less than 1/2 wavelength for all frequencies in the pass band, so each cabinet generates a true omni directional sound radiation pattern.

I) The front baffle is adjusted to be only slightly larger than the diameter of the driver, so that the response anomalies due to diffraction from the abrupt cabinet edge are moved up in frequency, and out of the pass band of the device. When the crossover is applied, the individual drivers are both omni directional, and diffraction free.

J) The small internal volume behind each driver does not have any dimensions larger than 1/2 wavelength for all frequencies in the pass band, so the volume will be entirely free of internal resonant modes (which only form at higher frequencies), eliminating any hint of overtones due to internal cavity resonance impacting the overall temporally delayed frequency response of the driver and cabinet. Transient response and energy decay are kept as short as possible.

K) The cabinets use a novel double wall construction, with a sealed air gap between inner and outer walls, so that shear wave vibration modes are not transmitted from interior to outer wall. This cavity wall construction greatly reduces transmission of vibration from inner to outer cabinet, and into the room.

L) The system design employs two drivers per channel, in a modified MTM configuration, for better time alignment at listening position, and better stereo imaging in the horizontal plane. The dual position also helps to excite a broader range of room modes, for better balance of mid-bass .

Implementation of any one, or two, advantages from the above list would be sufficient to claim an advance over the traditional ported enclosure design, however utilizing all dozen of the listed benefits (A-L) provides a substantial integration of incremental improvements, which collectively result in a true step change in acoustic performance compared to ordinary enclosures. Perhaps the best that can be achieved with these drive units.

As I mentioned before, the 1401 drivers, in almost any enclosure, have enviable pedigree, and as with most JBL woofers, they are truly adept at recreating the chest compression impact that I experience at live concerts, as well as the bite to cello strings, the pluck to bass guitar, and a real sensation that the drum kit is in the room. On well recorded rock and jazz CD's, these drivers re-create the impression that the band is in my house.

But the illusion does not stop there. The implementation of all dozen benefits (A-L) listed above are not in common use in the industry, so it is unlikely that the acoustic phenomena that I am about to describe will have been experienced by many readers of this forum.

The collection of design elements in A-L combine to create small, inert, omnidirectional loudspeakers, which effectively suppress; internal resonances, cabinet wall resonances, diffraction response anomalies, and temporal smear due to stored energy. (And yes, you don't need to ask, I have verified this claim by conducting a series of acoustic and accelerometer based measurements at various times during the construction phase of the project) Technically, this contributes to creating excellent impulse/transient response in the time domain, and acoustically, acts to eliminate most of the normal set of cues which we unconsciously use to localize sound sources. The overtones, resonances, and delayed energy which impact tone, and also usually betrays the position of loudspeakers in our room are mercifully absent from this design.

The best verbal description would be to say that the speakers "disappear" during reproduction of music (poorly mastered R-L *pan-pot* recordings excluded). With eyes closed, or room lights off, it is nearly impossible to point directly at the loudspeakers which are creating the illusion of a broad and accurate 3D soundstage across the front 1/3 of the room, not just beyond the speakers, but beyond the walls as well.

This is an effect which needs to be experienced to be understood, and so that limits what I can effectively write about it, except to say that the quality of imaging from this system is at a level that I have heard only very rarely, if at all, in other systems, and never before from any traditional "square corner, flat panel" box loudspeaker.

End of part II
Jack Bouska

[jack_bouska]

Nice job Jack!!

So does it sound as nice as it looks??

Rob:)


The next band up the frequency scale uses the venerable 10" JBL 2123 midrange driver, also housed in a cylindrical cabinet of 4.2 litres internal volume, crossed over at 243Hz 6dB/oct and 1kHz 12dB/oct LR).

My choice of this 10" driver was inspired by the Drew Daniels articles:
THE ANCIENT AUDIOPHILE'S QUEST FOR THE ULTIMATE HOME SYSTEM (D. Daniels) http://www.audioheritage.org/html/perspectives/drews-clues/audiophile.htm (http://www.audioheritage.org/html/perspectives/drews-clues/audiophile.htm)

and confirmed by continued use in his second system design:
CONSTRUCTION AND SETUP DETAIL FOR DREW DANIELS' HIGH-EFFICIENCY LOUDSPEAKER SYSTEM http://www.audioheritage.org/html/perspectives/drews-clues/system.htm (http://www.audioheritage.org/html/perspectives/drews-clues/system.htm)

Drew chose the 2123 for both these systems, and he aptly describes the sound of those drivers in the previously referenced articles, which I quote just below:

"As it turns out a mid cone supplying 300 Hz to 1200 Hz gives the proper effortlessness with very little power, and thus has extremely small cone excursions and low distortion.... The power absorbed by the mid cone driver amounts to milliwatts most of the time, which helps to hold harmonic distortion to very low levels, typically well below 1% THD up to dangerously loud volume. I experimented with a dozen midrange drivers before I was confident that the 2123H with its high efficiency and limited excursion linearity would produce sufficiently low distortion. It is a wonderfully transparent driver and a large part of the reason this speaker system sounds like listening to live music rather than loudspeakers." D. Daniels 1990 (If they are good enough for Drew D……)

I obviously don't need to "preach to the choir" of JBL fans on this forum about how good the 2123 driver sounds, but I will emphasize again that I believe I have "raised the bar" on the level of performance achievable from these units (or any other cone loudspeaker), by careful design of the enclosure, using small cylindrical cabinets employing all 12 of the design innovations listed in A-L above. The cabinet uses identical construction as the 1401 cabinet described above, and benefits from the same technical advancements in minimizing; diffraction effects, cavity, and panel resonance, and time smearing.

The only difference is in the conscious choice to use a 10" diameter driver over a frequency range extending high enough so that the diameter of the cone becomes a significant fraction of the wavelength of sound at it's upper crossover point. (as explained in D.Daniels tower system writeup, above). This results in a gradual change from omni directional radiation (below 600Hz) to approximately +/- 45deg at 1.5khz. This directivity narrowing, combined with the MTM configuration is intended to provide a smooth transition from the omni directional response below 600Hz, to the constant directivity response afforded by the waveguides crossed over at 1kHz.

All of the previously described, "invisible-inaudible enclosure" attributes, derived from the innovations in A-L above, also apply to the smaller enclosures as well. In fact this is mandatory, in order to match the unparalleled 3D imaging accuracy of the devices in the bands above and below this one.

In summary, the small cylindrical enclosures achieve all the design aims I intended. I suggest that the days of assembling a collection of high quality drivers, then mounting them all on the front panel of a rectangular wooden box, with passive crossovers, are over. Simply put: such an outdated method cannot approach the state of the art performance available today.

The frequencies above 1khz are covered by the waveguide & compression driver combinations which have been described (at length) in the previous posts of this thread, so I will restrict my comments to my subjective impression of their acoustic performance, as the technical aspects are well described previously.

In uncharacteristic brevity, let me just say: the waveguides sound better than I expected, and perform exactly to my ultimate stretch target, namely accurate and realistic 3D imaging with neutral tonality.

So each of the constituent parts fulfils it's duty as a stunning individual performer, yet this pales in comparison to the overall performance of the system as a whole. At the heart of this integration are a pair of Behringer DCX2496 digital crossovers, which provide the needed flexibility in crossover frequency and slope, as well as digital accuracy in time alignment and probably most important, comprehensive equalization capability for each individual band.

The overall choice of a five way design adds a few extra degrees of complication, but in return for battling the frustration of working through ten sets of wires, connections, amps and EQ settings, the reward comes in the form of an expansion of design choices not available with 2-way or 3-way systems.

The off-loading of the lowest octaves to the Altec allows for employment of small cabinets, and the ability to eliminate in-band diffraction effects is a direct consequence of restricting the diminutive cabinets to operate only within the two octave band in which they are both efficient, and omni directional.

Also, as each of the cone drivers is capable of operating over a much wider (5+ octave) bandwidth, this implies the frequency response is well behaved beyond outside the selected crossover points, allowing the use of audibly superior, phase perfect, single pole crossover slopes, for each of the cone transducers. This fully compliments the chosen Bessel system alignment, yielding the best possible phase and transient response attainable. Most readers will agree that 1st order xovers are very difficult to implement in a 3-way system, and probably impossible (at anything above whisper volumes) in a two way system.

I would remind everyone that this system has exceptional dynamic range capability, which also means that it can play VERY loud. (I use an amplifier rack capable of delivering around 5kW total into this load.) No doubt most people have heard commercial HiFi played loudly, at best it is uneventful, and uninspiring, at worst, just horribly distorted, compressed and clipped.

Again, attempting to avoid "preaching to the JBL choir" on this forum, I assume most members who use pro-sound drivers are familiar with the particularly desirable "non-HiFi" dynamic characteristic of low distortion at high SPL. When I turn the volume up on my system, the normal commercial-hifi cues of increasing distortion with increasing volume are absent, so the listener is immediately imparted with a sense of anticipation: how loud can this thing go? The effect can be downright frightening to owners of electrostatic systems (which have limited dynamic range, by nature), leading one fellow audiophile, after hearing my system, to describe me with the nick-name: "impact Jack". I resemble that remark enough to have adopted the same phrase as the name for my new system, which I call: The Impact-Stack.

End of part III

Jack Bouska

[jack_bouska]

Nice job Jack!!

So does it sound as nice as it looks??

Rob:)

If you have jumped to the last post in this thread, you may wish to scroll up and read the previous three posts (parts I, II and III) to get the answer in logical sequence.
(hey, only four posts required to describe a five way system, would have been less, but who knew that this forum limits each post to be 10,000 words or less!)

So, in searching for an apt description of the overall performance of this Impact-Stack, I decided to leave you with a single anecdotal description derived from a pair of listeners reactions during audition of the following track:

Chesky Jazz & Tests Volume 2, Catalog No: JD068 Track 47.General Image and Resolution

In this track, four musicians are recorded in a large, reverberant church venue, with voice and simple percussive instruments, parading up the isle, towards the crossed-array Blumlien stereo microphones, then three times around the mic-pair, and exit stage right. I personally find the presentation of this recording on my system uncannily realistic, and on two separate occasions, I recently played the same track to my 13 year old son, and later to a fellow audiophile from our London club.

The effect that all three of us experienced, is that of the musicians advancing toward the plane of the loudspeakers, then moving to the right of the room, beyond, and then out in front of the right loudspeaker, and, uncannily, *behind* the listeners head, then beyond the left speaker and back to the front (three times). The effect of the musicians moving in three dimensional space, in an oval pattern around in front, side, and behind the head is one that is rarely experienced with two channel stereo, and relies on wide band, flat frequency response, with adequate suppression of early reflections (which otherwise destroy imaging and interfere with low level resolution retrieval).

I should mention that both listeners were asked to read the Chesky liner notes prior to auditioning the track, so there is an obvious element of pre-conditioning, however on both occasions (1 week apart), I intentionally sat well behind the listeners, so they could not get any prompts or cues from me during the playback.

I was surprised, and gratified on both occasions to hear each listener spontaneously emit an excited chuckle at the precise point when the sound of the lead musician seemed to step out of the plane of the loudspeakers, and walk from right to left just *behind* the listeners head!

This amazing demonstration of the Blumlien 3D technique is one which clearly illustrates what level of realism is achievable with good stereo microphone techniques, as well as showcasing the (verifiable) quality of reproduction, and three dimensional imaging I have managed to achieve with the present system design. The underlying reason that both listeners emitted audible surprise at the same point in the track owes much to the nature of realism in the recording, and the listeners palatable, physically based, expectation and acoustic experience of four people, with percussive instruments, walking around behind them in the same room. All of this is accomplished without resorting to Q-sound DSP-tricks, and so naturally it is very surprising (and pleasing) to hear such realism coming from a two channel playback system.

Have I done my level best with the current design and implementation, is it now time to hang up my hat? Not by a long shot, I need to complete the new amplifiers which I am building for the compression drivers, and I have a backlog of ideas to try in terms of modified EQ for each driver, as well as switching the crossover slopes to be fully 1st order, top to bottom, as well as general level adjustment, and system tuning (and my to-do, wish list keeps growing).

So in closing, all hyperbole aside, although the above graphs can convey a great deal of objective information related to the performance of the system, and I can easily be accused of excessive verbosity in my subjective comments, the original question remains :

"how does it sound?",

the ultimate answer is: Great! but not nearly as good as it is going to sound after I tinker with it a little bit more!

End of part III
- end -

Jack Bouska

[jack_bouska]

Fantastic. You must certainly have spent a lot of time/fun planning and finishing off with very skilled craftsmanship.

Do you have original DCX2496? There was an article in audioXpress this summer about replacing the analog section completely.

Johnny - I do subscribe to AudioXpress, and did have a good look at the DCX mod article. I should mention that I'm not all that keen on modifying (Poogeing) commercial equipment - I much prefer to build from scratch where practical, or from kit when convenient - over the last 35 years of my active audio hobby, I have followed many a "cold trail" of replacing this capacitor, or that op-amp or power supply, only to find the resultant change would generally fail to meet my expectations for uplift, or the originators claims of sonic improvement. My experience is that overall system design and implementation (in electronics) is usually the dominant factor compared to individual component influence.

I don't remember the details of the article you mention, but I do remember being underwhelmed by the authors suggestion, deciding that in my case the result would be a significant degradation of the unit's capability (loss of single ended output, and much lower gain).

As an aside, considering scratch built crossover systems, I am slowly being convinced by Ed Wildgoose

< http://www.duffroomcorrection.com/wiki/Main_Page (http://www.duffroomcorrection.com/wiki/Main_Page) >

that it would be beneficial to replace the DCX's with a dedicated rack mount PC (with dvd transport, high quality sound cards, etc), which would also open the possibility of designing my own -ultra high spec - time domain crossover filters, including full room correction facility *intra-band*!

Wish us luck
Jack Bouska

[jack_bouska]

Thats totally friggin amazing
now i got a new project
gonna take me awhile to save up for 18" drivers
but i really wanna feel that bass


Morbo
Friggin amazing is just the start! After one listening session round my place, an audiophile friend remarked that the deep bass actually cured his constipation!

If you are saving up for drivers, I recommend trying the JBL 2245. The Fs of 20Hz would make a good drop in replacement for the Altec 18" I am using.

But be forewarned, these enclosures do require significant bass boost below 60-80Hz to work properly. Consider using 2nd order low pass, centred on 15-20Hz, as in the circuit attached below.
Good luck, and post the results of your project when you get it going

Jack

[yggdrasil]

Hi Jack.

I believe the mod's was the removing the opamps from the analog section, leaving no gain.


As an aside, considering scratch built crossover systems, I am slowly being convinced by Ed Wildgoose

< http://www.duffroomcorrection.com/wiki/Main_Page (http://www.duffroomcorrection.com/wiki/Main_Page) >

that it would be beneficial to replace the DCX's with a dedicated rack mount PC

Have not bumped into this site before, but: A PC with everything digital inside has to be interesting. And no need for A/D - D/A of the digital source.

[yggdrasil]

BTW: This could be interesting http://audioheritage.org/vbulletin/showthread.php?t=4850&highlight=crossover+theory

[jack_bouska]

BTW: This could be interesting http://audioheritage.org/vbulletin/showthread.php?t=4850&highlight=crossover+theory

Johnny - again, thanks for the link to the thread with the LeCleach spreadsheet and PowerPoint, very interesting! - I have briefly reviewed the information, and my interpretation is that J.M. LeCleach is proposing a specific set of time-shift, polarity-flip, and Fs adjustments, to compensate for the inevitable discontinuities, or slow phase rotations inherent in high order crossover filters (3rd or higher order). If a designer is forced to use high order filters because of two or three way topology, or employment of low power handling, or narrow band transducers, then I could see value in testing the LeCleach approach.

My system, and design approach, however, is a very different animal.

I use Dirac-Delta single sample pulse excitation, with a soundcard running at 96kHz A/D to estimate time-of-flight between each driver in my system, and the microphone at the listening position. I dub this "the gold standard" in time delay measurement, and can then adjust the intra-driver mis-alignment to sub-sample accuracy (limited by the time steps on the DCX).

I maintain this phase coherent accuracy by employing only 1st and 2nd order crossovers between frequency bands, which reduces (or eliminates) any of the objectionable phase rotations, and cross band polarity discontinuities which are described by LeCleach. I predict his method would do little for my system, because I have chosen a topology which does not suffer from the same high order xover problems. (no disease, therefore no need for a cure).

Apart from prodigious SPL and ultra-low IM distortion, this is one of the major benefits in five way design. Each driver is only required to cover a 2 octave power bandwidth, which is generally much narrower than the devices broader capability for bandwidth, power handling, and low-distortion frequency range. This means that 1st order crossovers, which generally place higher demands on out of band fidelity and power handling, are entirely permissible in a high-power five way design, employing ruggedpro-driver transducers.

I plan to take this concept to the technical limit (later this month) and test an xover implementation using all 1st order xovers, top to bottom. As extra LF power handling protection for the compression drivers, I will be using the method proposed by T. Sandrik which combines 1st order phase response, with 2nd order (ultimate) roll off. Sandriks method consists of using 1st order crossover slopes at the xover point, and adding an additional 1st order pole at 1 octave spacing on either side of the xover frequency. The phase anomalies associated with the additional poles are attenuated by the primary filters, and the summed phase response shows phase anomalies which are only a couple of degrees away from perfect.

This is the method that I had used 5 years ago when I was forced (by circumstance) to build a two way system with passive filters, although I don't advocate the use of passive filters for HiFi speakers, the results of the pseudo 2nd order implementation worked extremely well. (Good power handling with no audible or measurable phase distortion, have a look at the accompanying slides)

Jack Bouska

[Ian Mackenzie]

So are your drivers going to be in phase (polarity or 180 degrees out of phase)?

The earlier Dynaudios worked like your proposed idea and possibly the Duntech's, but they were true 1st order designs.


Yet another way, opted by Peter Garde (JBL 4430) was to shift the driver crossover co ordinates by rotating the L/C constrants to enable both drivers to be in phase at the crossover point and sum flat. This approach offers minimal group delay. I read about it in an AES article while sitting in the loo in the State Library once. Neville Thiele also has a new crossover design worth considering if you can understand his paper.

[jack_bouska]

So are your drivers going to be in phase (polarity or 180 degrees out of phase)?

All drivers should be wired such that an applied impulse generates a positive acoustic output pressure.

On most transducers, this means connecting the red terminal to the amplifier positive output, but on JBL's it means connecting the black terminal to the amplifier positive output. To avoid confusion, in the following discussion, I will refer to both the black post on JBL's, and the red terminal on other drivers (such as the TAD), as the device positive terminal

Referring to the above schematics, the JBL 2123, and TAD 2002 drivers are represented respectively by the 4 ohm R1 and 8 ohm R2 resistors. (I have omitted all driver reactance for purpose of clarity).

You can see that in both schematics shown, the positive terminal of the transducers should be connected to L1 and C1, which are on the positive side of the signal source (amp).

Polarity flipping of a driver isa "trick" used to fix a phase-induced response anomaly in 2nd order Butterworth filters, but is not needed for the Sandrik method, because the design is not a conventional 2nd order, but rather more like a Linkwitz Reily (cascaded 1st order), with the additional twist of frequency displaced (rather than co-incident) poles.

The basic Sandrik circuit is wired exactly as a first order xover, and I guarantee that when the additional poles are added one octave away (on one side, or the other, or both sides of the xover frequency), you will not be able to hear any difference, other than beneficial effects due to an increase in power handling, or improved out-of-band distortion suppression.

Jack Bouska

[jack_bouska]

So are your drivers going to be in phase (polarity or 180 degrees out of phase)? The earlier Dynaudios worked like your proposed idea and possibly the Duntech's, but they were true 1st order designs.

Thanks again to Ian for raising the question on polarity, and 1st order slopes, It got me to re-check my system, and try out a few more ideas.

Following Ian's posting. I took the opportunity a couple of evenings ago this week, to re-visit the polarity, crossover slopes, and gain settings on my crossovers. In particular, I compared an implementation of true 1st order (6dB/octave) slopes at all four crossover points, against a high order (48dB/octave) Linkwitz-Riely crossover alignment, at the upper two crossover points, between midrange, and my two compression drivers. Both the 1st order, and the 8th order LR require all drivers to be wired in-phase. The first order xover has no phase distortion, but requires high power handling, low distortion, well behaved transducers, and relatively high crossover points, which, fortunately, my system employs. (but I still dare not turn it up anywhere near max volume). The second implementation uses 8th order LR crossovers between the midrange and the big compression driver, and also between the big, and small compression drivers. This has the best power handling performance (I can take the system up to near clipping if I so desire), and also helps to attenuate out of band problems, like cone break-up on the midrange, and diaphragm break-up on the 4" compression driver.

I have not had much chance to listen to the two systems (time for bed when I got done). But rapid switching (using the compare button on the DCX), indicated that the majority of sonic difference is related to the variation in frequency response, rather than phase variation. In other words, the differences were not subtle, and my preference went toward the 8th order slopes, because the 48dB/oct rolloff restricted the interaction/overlap between drivers better than the 1st order. Maybe I can tweak the xover point and response tailoring a little more to restrict the overlap zone in the 1st order xovers from producing a response bump around xover. (which I couldn't cure with simple inter-driver gain settings.)

The first image in this post shows a graphical comparison of the frequency response for the 1st order xover (top graph, both channels shown), and the 8th order slopes (bottom graph). Xover points are indicated with vertical arrows. The biggest effect/problem is at the 1khz xover point, where the 2123 and 2441 seem to sum constructively on the 1st order, and destructively on the 8th order.

Additionally, I could not (easily) implement the Sandrik method, as the Behringer only allows cascaded slopes using the rather cumbersome EQ modules, and the HP and LP shelving options only allow a maximum cut of -15dB. When I applied one of these on either side of the 1st order xover points, I started running into headroom problems in attempting to set output gains between drivers. When I build my next set of dedicated amps, for the compression drivers, I will include a 1st order pole in each amp, around 150-300Hz, so that the low frequency protection is "built in" to the amplifier, and any "fat finger" maladjustments on my part will not run the risk of driving the tweeters with wide band, or low frequency signal.

One other area that I revisited (while I had the laptop/mic test & measurement kit out) was the low frequency compensation for the Altec. After a simple change (2nd order boost instead of 1st order), I managed to get the low frequency -3 dB bandwidth down to an astonishingly low 5 Hz. This is also the approximate low end of the power bandwidth of the system, meaning that the speakers *could* produce around 110dB spl in the octave between 5-10Hz, with the amplifiers near clipping. Of course, there really is no music content down that low, but just in case musicians start producing/recording subsonic tones, I'm ready.

The 2nd image in this post shows the mandatory "eye candy" waterfall plot illustrating the usual room modes (under damped below 35 Hz), and the prodigious extent of amplitude down near the 2Hz end of the scale.

The final image shows a low frequency linear scale amplitude plot (generated using a 5s MLS signal, and 400ms analysis window over the impulse response), for both channels. I'm still amazed that the speakers, (and Panasonic mic), have response down that low.

Jack Bouska

[John W]

Jack,
I was wondering if you could share some thoughts on mounting your 2 in horn flush with the cabinet face, like on the Zingali loudspeakers.
Thanks,
John

[jack_bouska]

Jack,
I was wondering if you could share some thoughts on mounting your 2 in horn flush with the cabinet face, like on the Zingali loudspeakers.
Thanks,
John

If done properly, i.e.: designed with the specific mounting arrangement in mind, then either flush with a cabinet face, or hanging in open air, will work equally well. However, the design of an open air horn, or conversely, the design of both the cabinet and horn in a flush mount arrangement, is far from trivial. I will post a more comprehensive answer later this weekend. If you would like to elaborate with some specific questions, then I can compose a better set of answers, rather than trying to cover too broad a topic. - Jack

[John W]

What I am considering is building the top end to a 4-way system. Basically a box with
a 12in 2202H midrange driver running from 290hz to 1.2khz,
a 2in 2445 using the same profile horn described in your initial post from 1.2khz to 10khz,
and a 2403 cats eye from 10khz up.

There would be a separate box housing the woofers.

I want to mount the horn flush with the face of the cabinet, the tweeter would be flush too, and the 2202H protruding the typical 1/4 in or so. One reason I want the horn flush is I may want to experiment with adding a slant plate lens over the top.

[jack_bouska]

What I am considering is building the top end to a 4-way system. I want to mount the horn flush with the face of the cabinet, the tweeter would be flush too, and the 2202H protruding the typical 1/4 in or so. One reason I want the horn flush is I may want to experiment with adding a slant plate lens over the top.

I understand the question better now, and can answer it quickly, while also taking the liberty to make a few comments specific to your design (in a subsequent post).

Your first question is related to the mounting arrangement for the waveguide on the front baffle. The quick answer is simple; with a slight modification, the contour I posted will work just fine when flush mounted on a front baffle.

The 2” throat waveguide contour is a composite of three curves;
1) Oblate spheroid (3-40deg), plus
2) tractrix (40-90deg), plus
3) quarter round radius (90-180deg).

To construct a flush mount version, simply omit the 3rd section, stopping the waveguide flair at the 90 degree point. You need to keep only the heavy colored lines, and omit the “rams horn” circles on either end. This will make the width of the waveguide mouth exactly 20cm (my diagram is erroneously labeled 10cm, and should read 20cm, sorry). You will need to work out the rebate/recess-flush mounting arrangement cabinetry yourself, depending on your carpentry skills and tools.

Experimenting with a slant plate lens sounds interesting, but I anticipate the results to be disappointing. These lens devices were designed to be placed in front of traditional (non-constant directivity) exponential horns, which exhibit strong narrowing of directivity beam width with increasing frequency.

G. Ausburger described the JBL lens’s as having a non-frequency dependant directivity increase, which means that they are best placed in front of narrow beam width devices (i.e. + - 20deg). The +/- 40deg oblate spheroid waveguide that I use behaves as a reasonable CD device across the full 80deg, (axially symmetric) coverage pattern. Placing a slant plate acoustic lens in front of this horn would expand the coverage in the horizontal plane (really wide), and interfere with the coverage in the vertical plane.

Various other aspects, such as internal reflections in the slant plate, and acoustic impedance contrasts at the front and back of the lens may also impart unwelcome tone to an otherwise simple, and relatively reflection free device. If you need a coverage pattern which wider, or asymmetric, I would advise designing a different shaped waveguide (e.g.: elliptical oblate spheroid), rather than using a lens.

A good analogy would be a trip to the optometrists, where a well meaning individual gets a new prescription for glasses. Based on the observation that these new lens’s improve his vision, he decides to buy two pair, and wears one over top of the other. While it’s true that this will alter the viewer’s perspective, it’s unlikely to be for the better.

However, there may be some benefit to employing a “home made” lens in front of the 2403 cat’s eye, but I’ll cover that topic in my next post.

Jack

[John W]

Thanks Jack. Sounds like I need to dust off the lathe and spin up a couple to give it a try.
I appreciate your explanation for the slant plate lens. I wasn’t sure on the exact function of these, but like the way they sound on some other horns I have. Leaving them off would be the logical first step.

[jack_bouska]

Thanks Jack. Sounds like I need to dust off the lathe and spin up a couple to give it a try.
I appreciate your explanation for the slant plate lens. I wasn’t sure on the exact function of these, but like the way they sound on some other horns I have. Leaving them off would be the logical first step.
I have uploaded a ppt file with the waveguide profile, this should make it easier to print off a 1:1 version for your lathe turning.

As promised, more thoughts on your design as a whole:

Generically, your design looks workable, and is similar to the top end of the 1st system described by D. Daniels: http://www.audioheritage.org/html/perspectives/drews-clues/audiophile.htm (http://www.audioheritage.org/html/perspectives/drews-clues/audiophile.htm)

First, bear in mind that I have not worked with any of the three drivers that you mention, so I will base my analysis purely on specifications, my own use of similar drivers, and some measurements made by Mr. Widget.

Second, my comments are my own opinion, and recommendations based on my experience and personal bias towards audio system design. You are cautioned to apply your own critical judgment before implementing any well intentioned suggestion, even my own. It’s your system after all, and you will be living with the sonic results.

You may benefit from some additional tweaking of the design specific to the topics of: Driver Sensitivity matching, directivity matching, distortion and optimum crossover points, as well as driver layout and cabinet shape for lowest diffraction and best imaging.

Starting with the 2202, this driver appears to have good sensitivity (47dB@1mW@30ft = ~ 97dB@1W@3ft) however the 2445 and 2403 may need to be padded slightly to balance overall frequency response. Do not calculate driver padding until after applying all required response tailoring in the crossover, as the additional filtering will have significant frequency dependant attenuation, which will reduce the overall need for driver padding.

The 12” driver has approximately a 10” diameter cone, and will start to exhibit directivity narrowing at about 500Hz, narrowing to a beam width of +/- 50deg at 1.2 kHz, where you selected the crossover point. My oblate spheroid horn exhibits fairly wide coverage between 1 to 1.5 kHz, rapidly narrowing to +/- 40 above 1.8 kHz, which implies that the coverage patterns of the two devices through the crossover zone will be reasonably smooth, and both direct and reverberant sound fields can both be balanced.

I don’t know where the 2202 starts cone breakup, but crossing over at 1.2 kHz should provide a reasonable margin, using 2nd order, or steeper crossover.

The 2445 is an industrial strength device. Use of the waveguide crossed over at 1.2 kHz will relieve low frequency related stress on the 2445, which allows the use of a conical or oblate spheroid horn contour. If you wanted to use the horn in the octave below (i.e.: down to 500 or 600 Hz), then I would suggest using a more traditional exponential, or tractrix contour (with a much bigger mouth), to provide better low frequency throat impedance, in order to keep the diaphragm from bottoming. For home use, with the 2202 running up to 1.2 kHz, and the 2445 on an oblate spheroid waveguide crossed over at 1.2 kHz, you should have a wide margin of safety.


The JBL specification sheet for the 2445 shows a detailed graph of the raw driver mounted to a 2” plane wave tube. Inspection of the graph reveals diaphragm break up commencing just under 10 kHz, and continuing for at least a half octave. I have not auditioned the 2445 in a home setting (but have heard lots of them in concert PA’s), however based solely on the graph, I believe you are wise in choosing to employ a dedicated high frequency device for the top octave.

My first suggestion would be to move the crossover point down in frequency to around 7-8 kHz, so that the distortion associated with the diaphragm breakup will be further down-slope from the xover point for better attenuation.

Above 10 kHz, the 2” exit on the 2445 will start to exhibit directivity which is narrower than +/- 40 deg. The oblate spheroid contour on the throat of the waveguide helps to expand the useful CD frequency range, but the diaphragm breakup at 10 kHz and above makes this effect unpredictable. Best to avoid problems (as I do) and move the xover slightly lower in frequency.

I would also suggest using active crossovers, (dsp based, preferably). If you must use passive crossovers, bear in mind that the response of the 2445 on the waveguide will require significant response tailoring to get the frequency flat in your listening room. If you have a pc, soundcard, and Panasonic based microphone, you will need to allocate a day or two to get the response sorted using a dsp based xover. If you are using analogue active xover, then allocate a week to the task. If you are boiling a passive crossover, it might take several weeks to get everything right using soundcard measurements, and tweaking by substituting components. If you plan to adjust the system by ear, then you probably need to allocate a lifetime to the job.

I am unable to locate a specification sheet for the 2403 cat’s eye ring radiator, however Mr. Widget has kindly posted a selection of directivity graphs for this, (and other ring radiator) devices. See: http://www.audioheritage.org/vbulletin/showpost.php?p=60871&postcount=4 (http://www.audioheritage.org/vbulletin/showpost.php?p=60871&postcount=4)

The device appears to have reasonable bandwidth down to as low as 2.5 kHz, with probable power bandwidth starting at around 5 kHz and above. A 7 kHz crossover would probably be quite safe from both power handling, and distortion standpoint; however you should apply caution and increase the volume slowly during the initial testing phase if you choose a lower crossover point than your original 10 kHz.

The use of a 7 kHz crossover is also viable from the standpoint of directivity control, as this frequency appears to be the breakpoint from aperture dominated directivity, below 7K) transitioning into horn controlled directivity (above 7k). This would be consistent with a device having approximately a 3” diameter mouth. (I’m curious; can you measure the 2403 mouth and let me know if I’m close?). A 7 kHz crossover would match the directivity of the +/- 40deg oblate spheroid waveguide below, and provide (according to Mr. Widgets measurement), narrowing dispersion above 7k.

I’m not all that keen on the +/- 15 degree coverage pattern above 10k for the cat’s eye, but there are a couple of things you could do to help alleviate this issue:
1) buy a couple of the quadratic diffraction phase grating devices, (as seen in the first picture in Don’s post: http://www.audioheritage.org/vbulletin/showpost.php?p=125026&postcount=1 (http://www.audioheritage.org/vbulletin/showpost.php?p=125026&postcount=1) ) and place them on the rear wall (behind you) directly in line with the path of the sound from the cats eyes. This will help increase the reverberant energy in the room by randomizing the 1st reflection from the HF devices.
2) Design and build your own version of a slant plate lens, and put this in front of the cat’s eyes, instead of the waveguide/2445. A sheet of aluminum, and a pair of tin-snips, plus a vice (for bending the crinkle shape) is all you need to build your own. One long skinny through bolt on either side, with nuts to space the plates will hold the shape and a can of black spray-paint will make the unit presentable. see Ausburgers article on this forum for operation and design information http://www.lansingheritage.org/images/jbl/reference/technical/lens/page01.jpg (http://www.lansingheritage.org/images/jbl/reference/technical/lens/page01.jpg)

My preference for best imaging and frequency response smoothness is to place the drivers in vertical alignment, minimizing the front baffle area around each driver. For an example of vertical alignment, see Mr Widgets design in the following post:
http://www.audioheritage.org/vbulletin/showpost.php?p=61015&postcount=19 (http://www.audioheritage.org/vbulletin/showpost.php?p=61015&postcount=19)

All baffle mounted transducers which have wide (180+ deg) directivity will suffer from diffraction related frequency response anomalies due to acoustic impedance changes at the baffle edges. The easiest way to minimize this in your design is to build a trapezoidal shaped front baffle, where the sides tapers such that the excess baffle area is minimized around each individual driver, and each driver has a variation in distance to all four edges of the front baffle. For examples of this design, see the following Forum web pages:
http://www.lansingheritage.org/images/jbl/specs/home-speakers/1990-250ti/page1.jpg
and
http://www.lansingheritage.org/html/jbl/specs/home-speakers/1999-tik.htm (http://www.lansingheritage.org/html/jbl/specs/home-speakers/1999-tik.htm)

That’s all for now, best of luck and please start a new thread in the DIY section detailing your progress on this design, with pictures and text.

Jack

[John W]

Wow! Thanks for thorough examination. It appears you’ve spent more time researching this than I did, and it is very generous of you to offer these suggestions.

I am still trying to digest all the information and arrive at a final plan. I’ll break things out into a separate post as things progress.
I recall seeing a post about a year ago on some acoustic lenses for the 075 bullet tweeters, but couldn’t find it. Honestly, I will probably try your first suggestion with the diffraction treatment and see where that leads. I will definitely bring the crossover point down to around 7 kHz.
Anyway, here is a link to a detailed drawing of the 2403 that I made some time ago that should answer your questions about the mouth size.
http://audioheritage.org/vbulletin/showthread.php?p=62755

[Amok]

Jack,

Very interesting design indeed. I have not been here for a while due to a combination of workload, plus bored to death of the same ol' on many speaker forums. Sure looks like I have been missing out:) . I use a waveguide in my system (DCX controlled also), built a few years back, but nothing quite this extravagant. Mainly to match the directivity of the dipole mids. I can only dream of woodworking equipment like you have, much less skills.
You have taken things quite a bit further. Your output/bandwidth requirements are rather drastically higher :spchless: .
Have you considered adding a few more (smaller, lower bandwidth) subs around the room http://www.harman.com/wp/pdf/multsubs.pdf to help with modal issues?
We share a great deal philosophically, but not with neighbor proximity LOL. I'm totally with you and Linkwitz with using cylindrical enclosures. I must also agree that the days of the big passive boxes should have ended decades ago...but I digress.
Excellent work with showing via measurements your design work. Bravo:applaud:

- Amok

[JoshK]

Absolutely inspirational! Everything about your research, plan and execution are a marvel. I thought that I have done a load of research on my planned projects (>1.5years and counting) but you have given me a few things to think about still.

speechless....

[noah katz]

Jack,

An amazing accomplishment.

One thing that puzzles me is how you can get such low extension from a TL of that length.

I calculate 1/2 wavelength corresponding to 4 m as 86 Hz, which I thought would mean LF starting to roll off at 43 Hz.

Thanks

[jack_bouska]

Jack,

An amazing accomplishment.

One thing that puzzles me is how you can get such low extension from a TL of that length.

I calculate 1/2 wavelength corresponding to 4 m as 86 Hz, which I thought would mean LF starting to roll off at 43 Hz.

Thanks

Pipes (or any acoustic cavity with planar ends) that are closed at both ends exhibit their lowest resonance mode at a wavelength equal to ½ of the pipe length. (The same is true for room or box mode resonances). This is because both ends of a closed pipe (or a rectangular enclosure) have a positive reflection coefficient, (they act like an acoustic mirror), so half a wavelength looks exactly like an infinitely long wave, for F=1/2 L (and harmonics)

Pipes which are closed at one end, and open at the other end work differently. The closed end (with or without a loudspeaker) acts just like an acoustic mirror, as described above, however the reflection from the open end will see a negative reflection coefficient, caused by the abrupt drop in acoustic impedance at the open end. The reflected wave will have a 180° phase shift (polarity flip), the equivalent of an acoustic “anti-mirror”. This means that the lowest resonant frequency will now be F=1/4 Length of pipe (plus a slight end correction). Overtones will be at odd multiples of quarter wavelengths. (3/4, 1-3/4 …) An open ended pipe (transmission line) is tuned to ¼ of the pipe length, and is often described as a quarter wave (transmission) line speaker. In practice, the ¼ L 1st resonant mode, and harmonics will be shifted slightly upwards in frequency, depending on specific box construction.

In the case of my 18” Altec transmission line the tuning is as follows:
¼ * (334mps / 4m) = ~ 21.5 Hz.

The calculated quarter wave resonance may be suppressed or enhanced by the specific construction details of the transmission line cabinet, and TS parameters of the driver. In my case, I undersized the internal volume, and chose a diameter at the driver end which is very close to that of the speaker cone. Given the low QT and high Vas of the Altec 3182, this resulted in an over damped system (Q~ 0.5), with a LF roll off similar to that of an undersized closed box. In other words, the driver dominates the enclosure. The small internal volume also significantly reduces the port output at ¼ L, and higher (with stuffing). The Martin J. King TX simulation for my arrangement is shown below in the first figure in this post. The lower graph shows the cone output in red, and the port output in dashed-blue. The port contribution is minor compared to the cone output (not typical of transmission line speakers in general), resulting in a strong roll-off of the low frequencies. The device acts as a critically damped closed box of relatively small size, compared to the Vas of the driver.

The Martin J. King simulation indicates the requirement for approximately 12 dB/ octave low frequency compensation, but in practice, in my room, I have applied slightly less than 6 dB/octave, between 20-200 Hz in my system, so the question is: what mechanism(s) provide the additional required gain? (I use 15dB boost, while the simulation suggests >25dB is required at 20Hz)

As mentioned above, because of the small volume, my system is driver dominated, however this particular construction also generates a Helmholtz resonance at a frequency much lower than the ¼ L internal mode. The column of air behind the speaker cone acts as ported enclosure, with the air-slug mass working against the compliance of the driver. To my knowledge, only Martin J. King has reported this phenomenon in relation to transmission line loudspeakers. I first encountered the effect when I assembled my un-stuffed 4m line, and then pushed + released the Altec cone. The un-damped driver visibly oscillated with high amplitude for several seconds, at a few Hz. I was quite taken aback, never before having seen a driver do this without being connected to an amplifier!

Subsequent relative impedance graphs confirmed the lowest resonance frequency to be 4Hz in the 4m transmission line. (I used a CROWN DC300A for accurate response down to DC).

I also measured the transmission line relative impedance with only the internal tapered tube, equivalent to a 2m transmission line. The relative impedance plots for the un-stuffed and stuffed 2m and 4m lines are included at the bottom of this post. The 2m plots clearly show that the un-stuffed transmission line is behaving more like a ported enclosure, with the entire tapered tube acting as a very large & long port, to generate a very low tuning frequency (~8 Hz in the 2m tube). Note that the overtone distribution only vaguely adheres to simple quarter wave pipe theory due to the influence of the acoustic reactance associated with the ported enclosure (Helmholtz) response

The relative impedance graph for the 4m transmission line shows that the frequency has moved down to 4 Hz. In both cases, long hair wool damping reduces both the fundamental resonance, and overtones resulting in a smooth acoustic output from the cone. Although well damped, I note that the 4Hz resonance is still buried within this system, and so the port output at these frequencies probably augments the lowest frequencies below 10Hz.

Another contributor to low frequency reproduction in my system comes from “room gain”. Although many posts cite this effect as important for low frequency extension, in a typical room the effect is mainly one of near field proximity to acoustic boundaries, which limits radiation to ½ space, ¼, or 1/8th space, depending on distance from a corner. True room gain is common in small listening spaces, such as automotive interiors, where the cavity volume is so small that all bass waves are much larger than any of the internal dimensions. Domestic listening rooms are generally large, and commercial speakers have typical bandwidths which restrict LF or subsonic output such that true room gain is relatively rare.

Rare, yes, but not impossible, as explained in the following text, extracted from my website:

“Overall bass efficiency is likely due to the proximity of each driver to the three way corner (masonry walls and cement ceiling) such that over their entire bandwidth (below 110Hz) the Altec drivers are radiating into 1/8 space, i.e., the three rigid acoustic boundaries are within 1/8 of a wavelength for frequencies below 140Hz, which makes the apparent efficiency of the drivers about 9dB higher than if they were radiating into free space. (aprox 3dB for each boundary, ignoring absorbs ion and transmission losses.)

Furthermore, the dimensions of the room are such that for all frequencies lower than 35Hz, the shortest dimension, the 8ft ceiling height is less than 1/4 of a wavelength, and for all frequencies below 20Hz, the medium dimension of 14ft. will be less than 1/4 of a wavelength, and finally, for all frequencies lower than 15hz, even the longest dimension of 18.75 feet will be less than 1/4 of a wavelength. As the frequency drops below 35hz, each of the room boundaries will move through from the range of beyond 1/4 wavelength, down to less than 1/8 of a wavelength (at the frequencies of 17.5Hz, 10Hz, and 7Hz respectively). In other words, for frequencies below 140hz, the ceiling is within 1/8th of a wavelength, and augments the sound pressure level by almost 3dB, when the frequencies go below 35Hz, the floor is also close enough to start contributing to the SPL of the sound, and this increases as the frequency is lowered to 17.5Hz, where the floor is 1/8th of a wavelength from the driver, and the phase of the waveform is nearly identical, regardless of height, and the SPL will get almost a 3dB boost below 17Hz. Likewise, for frequencies below 20Hz, the side walls start adding in phase, reaching maximum contribution below 10hz (another 3dB), and the longest dimension contributes below 15Hz, ramping to 7.5Hz at which point the room will be in complete isophase condition.

Of course wave fronts still propagate throughout the room at the speed of sound for these low frequencies, however the rate of change of pressure amplitude will be so slow, and the wavelengths so large, that each point in the room will appear to have identical pressure amplitude, and phase for these low frequency waveforms. At frequencies below 7Hz, essentially all six of the room boundaries will be closer than 1/8th of a wavelength, and this will yield an extra 9dB of gain, for a grand total of 18dB extra SPL when the corner placement is included. The natural progression across the 1/4 to 1/8 wavelength (zero dB to +3dB increase) combined with the staggered dimensions deliver a gradual ramping up of room gain, which mimics the missing boost below 20hz. The trick is particularly effective in the current room, because of the full masonry construction of the four walls, and the uncommon use of prefabricated concrete floor and ceiling, such that the low frequency reflection coefficient is very near unity. “

Jack Bouska

[Flodstroem]

Hello Jack
Im amazed and impressed. What a great work. I must confess that I was hit by a lightning and couldnt move from my seat before reading all your posts till the end (all night long) I have read all your posts with great interest and everything you write ("say") seems to be so obvious to me that I think I must try some of your ideas and suggestions.

Though I have a lot of and also some similar drivers as you have been using in your project Im going to try to build a similar audio stage. But maybe not as complete as yours because of the costs involved and I also lack most of your skills regarding technical education etc.. (Im a biologist)

But my first step will be to buy some audio measuring equipment (and I have already done some buying) so I could do my own empirical verifications on how different parts really do behave acoustically in my home and also as an instrument to verify my acoustically calculations.

Next step will be to buy a Behringer DCX2496 digital crossover for to be able to set different types of filter slopes etc.

When reading all your posts I couldnt find any information regarding how you built those round "hat-boxes" for your speakers (for the 1401ND, and the 2123H). I am aware of that, they was made out of two boxes, one in another, with air in between them for to not get acoustic resonances from standing waves from the inside to the outside. But how did you built them?

My own drivers from the beginning will be two 2245H, two 2012H, two 2440 (with 2445 Ti diaphragms) and two, either 2421 or 2405s´ I dont have any 1401ND but could use two 2215H as a start.

Have planned to build a large ML-TQWT pair but this (yours) approach seems to better fit the very restricted WAF-factor in our house.

In DiyAudio Forum Im involved (and has been) in the construction of a Krell KSA100mk-II Clone and I plane to build 4 mono blocks for the purpose: 120W/8 ohms, ca 220W/4 ohms (120W pure class A each). Including my two NAD, 6 x 30W (3 x 90W bridged) I will end up with a power source of ca 1.4 kW to start with. This is not enough so I plan to sell those NADs´for to be able to build four or six GB300 (Gregg Ball) MOSFETs´power amps later on. Total power source will then be ca 2.6 the twice of what I will have from the start.

And yes, luckily I have a lathe for the turning the two horn shapes.

Regards

[jack_bouska]

I have read all your posts with great interest and everything you write ("say") seems to be so obvious to me that I think I must try some of your ideas and suggestions.

Thanks for the feedback, I hope my comments on system philosophy and horn design will be of some use to you in your speaker building projects.

Also, apologies to the Lansing heritage membership for my recent inactivity. For the last three months, it seems my spare time is measured in negative numbers! I seem to be inordinately busy at work, with my evening and weekend free time consumed by preparation for some unexpected external commitments to the European Association of Geoscientists and Engineers (see:
http://www.eage.org/index.php?Menu_Code=DLPCourseDetails&EVS_Id=197&ActiveMenu=85&Opendivs=s19,s36 (http://www.eage.org/index.php?Menu_Code=DLPCourseDetails&EVS_Id=197&ActiveMenu=85&Opendivs=s19,s36)
and
http://www.eage.org/index.php?Menu_Code=SCPCourseDetails&EVS_Id=191&ActiveMenu=85&Opendivs=s19,s41 (http://www.eage.org/index.php?Menu_Code=SCPCourseDetails&EVS_Id=191&ActiveMenu=85&Opendivs=s19,s41)


When reading all your posts I couldn't find any information regarding how you built those round "hat-boxes" for your speakers (for the 1401ND, and the 2123H). I am aware of that, they was made out of two boxes, one in another, with air in between them for to not get acoustic resonances from standing waves from the inside to the outside. But how did you built them?

A picture is worth a thousand words, so rather than write another series of long posts (which I still cant afford the time for), I am including a series of photo's taken during the construction of the speakers. (in this, and a subsequent post.)

The pictures are numbered and captioned, and should be mostly self explanatory. If you have specific questions, or want more info related to some aspect of the process, you can post your question with the photo number as reference, and I can comment appropriately.
Jack

[jack_bouska]

Post number two (of two)
The pictures are numbered and captioned, and should be mostly self explanatory. If you have specific questions, or want more info related to some aspect of the process, you can post your question with the photo number as reference, and I can comment appropriately.

[Flodstroem]

Jack, thanks for the excellent pictures of your hard-work. Amazingly. Yes it was worth more than "thousand words" and it looks as it could be possible for me to built a system like yours.
Maybe not exactly due to I lack some of the parts and also, it must fit our listening room too (and the WAF factor not mentioned).

My first question (of the 101 I have -no just kidding):D:

What the h.....l is "RAPIDOBAT", newer heard of it before. Could you explain it in details

Regarding the construction of the speaker boxes this is absolutely clear for me by now, very smart solutions.

For the moment Im doing a research for to how to find the parts and materials for to build the boxes. I think this is the basics start for me. Then when I have found out I could start to order materials and some hardwares too that is not to find in my garage and working shop.

Regarding your design, you are using two 14", two 10" one 2" (2441) and one 1" (Tad 2002). Was your intention to build a MTM- concept or was there any other reason for to use two 14 and two 10" per side? loobings etc..?

My system at the very first stage would be to use one 15" (2215H), one 10" (2012H) one 2" (2445 Ti) and one 1" (2421) /or 07(2405) per side. Also, two 2245H for the two subs.

I dont know if its possible to build a system out of what I have by now but Im planing to buy some more speakers, alternative would be to sell some and buy other models that is a better match than those I have planned to use.:)

Thanks.
Regards

[aust-ted]

Jack, You have done a grand job on this project and your write up is inspirational. Having been out of action for a while i just read it and will attempt some JBL 2441 waveguides.

I am thinking of a simpler system than yours with dual JBL 2235s in large reflex boxes (already in use), a single JBL 2123 (I only have a pair), a single JBL 2441 and a JBL 2405. I also have a DEQX acquired after advice from Mr Widget (Thanks Mr Widget, have never regretted taking up that excellent advice).

I have recently acquired a wood lathe and expect to be in a position to construct horns using templates much as you have described on this thread after some further tuition at the end of this month.

I am proposing using the DEQX to xover the 2123, 2441 and 2405 with restricted box design to assist the low end cut-off of the 2123. Have not worked out the xover for the 2235s but will probably try both passive and active. Will use a 4 amp configuration. The DEQX provides a lot of flexibility in the xovers and equalisation for the 3 top speakers but my limited budget does not permit me to purchase a second one for the bottom end. Any comments on this would be welcome.

Jack I have a question which arises from your work. I note you are xovering the 2441s at 1KHz using a modified oblate spheroid with a tractrix mouth to save space. You have also given advice on the benefits of xovering it higher at 1.2KHz with a 2202H to "relieve low frequency related stress on the 2445". Sounds like good advice that I had not previously thought about.

I had been thinking about trying to extend the 2441 down to 500Hz as they work in some JBL horns or even trying the JBL phenolics down lower but I suppose with the 2123s there is no need to stress the 2441s.

Is there any software available to generate your own oblate spheroid waveguides with different cutoffs like there is for tractrix? As I propose to only use a single 2441 I will not have the need to use the modified profile you used. I am thinking of making a number of different ones partly to hone my turning skills, to achieve a higher WAF but also to experiment to see which one gives the best from a sound perspective.

Regards
Ted

[JBL 4645]

Hello Jack
Im amazed and impressed. What a great work. I must confess that I was hit by a lightning and couldnt move from my seat before reading all your posts till the end (all night long) I have read all your posts with great interest and everything you write ("say") seems to be so obvious to me that I think I must try some of your ideas and suggestions.



I’m kinder feeling immobilized I’m in perpetual daze. :blink: I can grasp some of the posts the rest is like light-years over my head.

I like the Altec 3182, wow looked it up on pdf file nice. I looked around eaby, nope not one single Altec 3182 sub around.

Do you still run the same system as this thread is few years old now and I’d be greatly disappointed if you not running it, after spending great deal of attention to detail, that is a labour of love for ones Hi-Fi.

I can imagine the tightness in the low end supporting the mid and high range with great strength of authority.

Well done Jack :applaud:

[Flodstroem]

Jack, You have done a grand job on this project and your write up is inspirational. Regards Ted


I’m kinder feeling immobilized I’m in perpetual daze. :blink: I can grasp some of the posts the rest is like light-years over my head.

Well, I felt the same way when first reading those hieroglyphs by the excellent writer Jack :crying:

But later I have come to understood more and more about his concept :)
Due to costs etc there have been a long journey to come to the position that Im in at right now: I could build a similar concept only with some "small" substitutions of the drivers
The components I intend to use and what I have in hand are those:

2 x 2245H
4 x LE14H-1
4 2123H (and/or 2012H)
2 x 2440
2 x 2421
some Rapidobat tubes (hehehe)
also 2 x DCX2496

Amps: some "small" building project:
1 Gregg GB300D (that you Ted might know anything about)
6 x Krell KSA100 mk-II clones.

including a great lathe :blink: for the Tractricx horns