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Requirements of Transducers for Loudspeaker SystemsThorsten Loesch A bit further down (in the Forum) someone made an interesting statement along the lines of: "It is more or less established there that a real fullrange driver goes from 20 Hz to 20 kHz +/- 1.5 dB" I find this very a very interesting illustration of what preconceived notions can do to the way we view a subject. First of all, the 20Hz-20kHz "audible" range is something concocted on a green table, by essentially drawing a line on a piece of paper. There was some reasoning and research behind this for sure, but that was also a lot that was ignored. I would like to see some discussion of what matters in reproducing music and musical instruments, instead of waving specifications and numbers written on paper around and saying "It must this otherwise it's not Fullrange or Hi-fi or whatever....". Now I am well known for having substantial "prejudices" about Audio Technology. I often quite dogmatically declare certain technologies to be superior. Much of the background for these dogmatic declarations comes from a study of music itself, long terms of working professionally in recording and reproducing music and in recent years from carrying out many "all else being equal" blind tests in a relaxed, unpressured but very controlled environment. Moreover, much of I find to matter from a rational view of the subject, based on physioaccoustic research and psychoacoustic research as well as electronics has been born out in listening tests, indicating usually a quite substantial correlation of theory and reality… But not, I have not yet formulated the GUT (Great Unified Theory), not even for music reproduction, nor do I intent to. I'm an Engineer in the end, all I want are parameters to which I can design equipment so it makes a better job out of reproducing music. And that is where we come back to transducers, especially in our case Speakers. And one of the things that matter in absolute terms least, but which are being used as primary criteria in establishing the quality of Speakers is Frequency Response. Now before we have a close look at what REALLY matters in speakers, let's first look at a few connected and highly intriguing statements (both are from memory and correct in content, but not verbatim): 1) Spectral analysis on a Harpsichord shows that a played Harpsichord will produce significant energy above around 8Hz reaching up to beyond 72kHz. (someone actually researching the behaviour of musical instruments). 2) The Frequency range needing to be reproduced for a music lover (as opposed to an Audiophile) is 100Hz to 9kHz. (a well known critic of classical music, - he later revised his view as to mean 50Hz - 12kHz in the light of hi-fi Equipment allowing this range easily to be reproduced). 3) The 3m in room smoothed response of the Speaker fitted into a commendably tight 10db Window between 100Hz and 8kHz. (D.B. Keele in a review of a well known audiophile Speaker in "AUDIO". The accompanying graph from a Crown TEF showed appx. +15db and -20db extremes for the unsmoothed deviation from 0db in the same 100Hz to 8kHz band. The same speakers measured almost perfectly flat anaechonically. The Speaker also was judged quite favourably subjectively in this and many other reviews). So from the above it appears that we have two substantially competing or even contradicting views. Many people will agree that the 100Hz - 10kHz range is the range within which most of the musical "information" is encoded. Yet we will also find that in the real world and at our listening position a perfectly flat response Speaker will be subject to many narrow band rises and drops, overall making about 35 db between the lowest and highest point and that is only in the musically "relevant" 100Hz - 10kHz range. At the same time, many Instruments (If we take into account "parasitic" sounds - like the mechanisms on the Harpsichord giving rise to extremely low tones at quite high intensity) will produce sounds over an incredibly wide range of frequencies. So, do we need speakers capable of 8Hz - 72kHz (+/-1db), or is 100Hz - 10kHz (+/-10db) more than good enough in the real world? And is the Frequency response actually particularly important as measure of quality, particularly in the context of current 2-Channel Audio? Allow me to take as an example one specific Instrument in order to illustrate the kind of demands made by a 2-Channel (say ORTF Array miked) recording of the instrument chosen. For this example we shall use the "Kettle Drum" or the "Timpani". This a very large Drum, tuned to a fairly low resonance, depending on a number of issues often in sub 50Hz Range. When struck by the "mallet" the impact will move the Skin quite far off it's resting position, instantly generating a strong pressure wave of very low frequency, which is actually only a "spike" into one direction, having a very steep rise. In addition, the interaction between mallet and skin will produce a kind of "white noise" wideband sound reaching up to above 2kHz. Both the pressure wave and the noise originate from the same point in space, the point of the Skin where the mallet struck. We will then (after the initial Impulse, when the mallet has been literally bounced back by the restoring force in the Skin and has left the skin) see a starting oscillation at the tuned resonance frequency of Skin and Drum "Kettle". This Oscillation was stared by imparting a Impulse to the Skin and will reach quickly reach a maxima and fade out slowly unless damped out. The origin of the resulting sine wave (and it's harmonics) will be much less localised than the first, as the whole skin now participates as well as other parts of the Instrument. If we where to place a flat response transducer at the position of the Drum and where to take a Frequency response sweep with the two Mic's of the ORTF array placed as usual we would find a severely non-flat frequency response for all of this. Yet, energy between a few Hz and a few kHz is being produced, even with the relatively low tuning of the drum. This is because a non-linear resonating system (like a string, a Drum Skin and so on) produces not only Harmonics of the Fundamental, meaning tones with twice trice and so on Frequency but also Subharmonics of the Fundamental, meaning tones with halve, a third and so on Frequencies. Moreover, a particular phenomena in Human hearing (it is based on a mixture of the physioacoustic behaviour, or so to speak the mechanics, of the ear and some psychoacoustic components, or so to speak the signal processing in the brain) also has some impact. If we hear the correct set of Harmonics from a tone with the fundamental (and even subharmonics) fully suppressed we will still hear the actual tone (or note) as our ear and brain system "reconstructs" the actual note from it's harmonics. Similarly, the absence of one specific harmonic again has little impact on the perception of the tone of a note. At the same time, the perception of the placement or position of the Instrument is actually being determined by initial wideband impulse and pressure wave, which hence if not reproduced time coherent will result in a diffuse perception of the Instrument. Interestingly it seems that also our perception of dynamics is involved here. BTW, the perception of the note itself is of course based on the actual fundamental Frequency, while the perceived "tone" of the instrument is based purely on the harmonics (and to a degree subharmonics). Lastly, it is interesting to note that all natural events follow a roughly gaussian distribution, with usually quite identical shapes of the distribution curve to both sides of the peak. This also holds for instruments and their harmonics and subharmonics, which are also forming (on a linear frequency scale) the usual common "bell" shape known to anyone ever having studied statistics. All this is quite complex a subject and one that it would have enough material for a few dozen dissertations in it, so don't expect to even make a full outline of what all this implies. Let me just come to some of my conclusions here.... A Transducer intended to reproduce music (including human voice and the like) should have a fundamentally balanced response over a fairly wide frequency range (as wide as possible) but it MUST UNDER ALL CIRCUMSTANCES be balanced over the 100Hz - 10kHz range, using 1/3 Octave averaging. Narrow-band deviation from a flat response are fairly harmless, especially notches. It is desirable to have the transducer to offer some Energy outside the 100Hz - 10kHz band, ideally on par with the 100Hz - 10kHz range. By only reproducing the 100Hz - 10kHz range it is still possible to capture all relevant fundamentals and harmonics, including the harmonics needed to "restore" the relevant very low Notes below 100Hz and the Subharmonics to "restore" the noise type happenings above 10kHz. Indeed, all the music, all notes and phrases will be able to be followed and pretty much all nuances will be captured. The slight impact is on the tonality of certain instruments, especially at the high end with such as Harpsichord, Percussion and the like, as well as the physical sensation form the fundamentals and pressure waves of the lowest notes. While these items contribute to a closer "illusion" in many cases, it is entirely possible to enjoy music without them. It appears however quite important that our Transducer shall not alter the distribution of Harmonics, so any distortion should be low, of the type giving a simple Gaussian spectrum and ideally of very low order. Furthermore, if any semblance of correct dynamics, timing (I know, I did not cover this earlier) and positioning are desired, it appears that the larger part of the 100Hz - 10kHz range should be reproduced in a manner and fashion that is time coherent. This usually implies the use of a single transducer (even multiple transducers driven in unison seem to lead to some blurring due to their no longer being in unison at higher frequencies. Anyway, the above points strongly towards the absolute need of using high(ish) Sensitivity, wideband Transducers if our goal is the musically correct reproduction of the Music. The reasons for this are as mentioned the time coherence (which even coaxials of the Tannoy or Altec mould cannot manage, though the old RCA LC-1A 15" Coaxial did). Also the presentation of two point sources (matching the 2 point "pickup" from our Microphones) and the low Distortion (there is a link between distortion and efficiency using current technology, more about this at another time) are relevant. As we have seen, Frequency response as such is not a primary issue at all. IF the "sensation" of "being there" is desired also on a physical Level (impact) it appears the use of well designed Subwoofers (or the use of wideband drivers with a capability to go low enough) seems desirable. In addition, for those people who still have a decent HF hearing (myself being a case point) it might be desirable to have drivers with more extended HF or indeed Supertweeters. I prefer to use and can hear the difference made by a Supertweeter crossed in at a nominal 22kHz [-6db] covering up to about 40kHz. Anyway, back to some numbers so we can start designing something. These numbers focus on the MINIMAL ability to reproduce acoustical Jazz and classical Orchestra, as well as Vocal Works like choral music and opera. Modern styles like rock and pop which are usually fully electronically created and have no "acoustical" Yardstick against which to compare are not being considered, yet they are usually also well served by Transducers covering or exceeding the noted specifications. Some of the requirements are not based on material discussed here, I'll come back to this later. The Transducer(s) should be able (IMHO) to reproduce 96db SPL at the listening position with less than 1% THD of the simple type in the 100Hz - 10kHz range (this includes also the amplification devices) and with less than 0.3% THD in the 300Hz to 3khz range. (For stereo this applies to a pair of transducers in typical positioning, being driven by equal amounts of power). It should be free from significant Power Compression for 96db SPL at the listening position. (For stereo this applies to a pair of transducers in typical positioning, being driven by equal amounts of power). The Transducer should have a even energy balance in the 100Hz - 10kHz range. The Transducer will desirably have notable amounts of energy in the whole range covered by the recording, meaning 30Hz to around 50kHz on LP, 10Hz to about 12kHz on CD. (Note: above 12kHz anything coming from the Output of the DAC/CDP playing a CD was not part of the original recording, but is spurious sound generated by the digital Filter or simple Distortion if no oversampling is used). The transducer should become more directional with rising frequency to reduce the room influence by promoting direct sound over the reverberant field. The transducer must be capable to preserve the fundamental Waveform of a triangular wave and a squarewave at least for the 300Hz - 3kHz range, a much wider range being desirable. At any extend, it should be clear that as far as I'm concerned a truly flat 20Hz - 20kHz frequency response is nothing that would be required for the reproduction of music. Indeed due to room influences having such a transducer would actually result in severely lifted Bass response in most rooms and hence an ill defined and boomy response. As often a narrow HF dispersion is also being criticised, I would like to add that wide, uncontrolled dispersion of the upper midrange and treble destroys imaging. It actually promotes the rooms reverberant field over the direct sound and at the same time it promotes a overly bright sound as the usual "housecurve" for higher frequencies is not promoted, instead the energy balance in the reverberant field remains flat. Lastly, it should also be clear now that using multi-miking and multitracking during recording will severely compromise all aspects of the musical performance recorded even WITHOUT massive additional "doctoring". I hope all these slightly unconnected ramblings and notes help to clarify some of the reasons why I feel that wideband Drivers MUST be used and why I feel that the nowadays very common 3 or 4-Way, flat response, low efficiency Cone/Dome speakers are mostly incapable of producing "high fidelity", if fidelity to the musical event is our goal. I welcome constructive criticism and questions on clarifying specific points made, where no explanation was forthcoming or where it was by far too sketchy. I will aim to address these eventually. Horn LoudspeakersNow, a while earlier I pointed at all the hidden intricacies in the TQWT Design and how this kind of design could quite clearly be optimized very well for a given driver using either inventiveness and knowledge or cut & try with dumb luck. And I promised I'd come to horns too. So, let's actually look at a "horn". We know two fundamental types, Rear-loaded Horns and Front-loaded Horns. Frontloaded Horns usually have all the Sound Radiation available from the Front of the Driver only and almost exclusively. All Compression Drivers and the like operate on Front horns. So does the ORIS 150 Front horn, Avant Garde Audio and a few others. Let's for a moment talk about "front horns". They are IMHO better termed "Waveguides" as what they essentially do is to "guide" the sound waves so that they are fairly tightly "bundled". There are few shortcuts when designing such horns. Regardless if the driver is compression loaded (increases sensitivity at the expense of bandwidth) or not, Front horns generally need to have a smooth expansion curve, sharp bends tend to make a mess out of the sound. A typical example of Horns with severe problems are all so called "constant directivity" Horns using multiple conical sections. While this indeed results in a very tightly directivity over a wide frequency range, the Frequency response usually suffers substantially, requiring complex equalisation. All in all Front horns are indeed real horns, with a Horn mouth area equal to that requires and a suitable depth. Sometimes Front horns can be too short, resulting in the Cancellation (just like with open baffles - same principle) to set in at frequencies higher than the Horn mouth Cutoff. This is better avoided if we are horn loading Wideband or Fullrange Drivers. Rear Horn-loaded LoudspeakersHere the Cone of a conventional (wideband or not) Driver radiates a significant degree of the sound and especially the upper octaves above a few 100 Hz. There are many examples here for such horn, These Include the Carfrae and Hedlund Horn, also typical Lowther factory enclosures (Accousta, Medallion, Fidelio, Bicor types) as well as many of the enclosures designed for the Fostex Sigma Series, the Lamhorn, the Edgar/Moth Horn and the PA Style "Scoop" as used in the Koechel Speakers. In any case, the frontwave of the Driver remains undisturbed but the rearwave is applied usually to a modest size compression Chamber (with modest compression ratios) and then coupled to a horn, which may be straight (no idea about an example), curved (Carfrae, Hedlund) or Folded (Lowther Factory Enclosures and the like). Again the Horn really operates as "waveguide", focusing the sound waves and hence in it's operating range offering acoustic gain (this depends upon the profile, but as much as 9db are possible). HOWEVER, at least in their lowest octave or often even in the two lowest octaves, our "Horn" does no longer operate as Horn at all. How so? All domestic rear loaded horns known to me are "foreshortened". This means they are shorter and have smaller mouth than demanded by classic design theory. This is being done simply to keep enclosures manageable. The customer is told that placing this Horn in the corner will "extend" this horn. This little fairytale has been told since times immorial and various "corner horns" have deliberately used this. BUT, please give me a break and look at the coupling and the acoustic Impedance matching.... No way this all falls together as a Horn (excepting for a moment the Klipschorn which is really a folded Frontloaded Horn with the driver loaded into a compression chamber). So what is happening here? Simple. Let's take a Horn having an internal length of 2.5m and a 0.16 square meter Horn mouth. This Horn BTW is one of the few examples known to me that seem to offer a decent operation and good LF bandwidth. Now I'm hard pressed to do the math for the Horn mouth related cutoff at this time of the evening (someone, please help), but I suspect at the best we are looking at around 100Hz (I suspect more like 150Hz) or thereabouts. Yet for this specific horn the Plots show a (not particularly smooth I might add) response down to 33Hz!!!! How so? Okay, here is the secre t.The Horn is a (folded) horn down to around 100Hz or whatever the real cutoff is. The "image" of the horn on the floor contributes effectively to doubling the Horn mouth, but with some loss of efficiency in the octave below the nominal cutoff. But that still doesn't get us anywhere near the 33Hz. But what if we actually (for a moment) view the Horn as an unstuffed 2.5m long quaterwave transmission line? What would our (undamped) pipe resonance be? Any takers? No. 334m/s sound speed divided by 2.5m divided by 4… 33.4Hz Is it dawning yet? At the low end we have a simple quarterwave resonator, undamped. Now this still can nowhere near match the SPL generated form the Horn loading the rearwave, so what more is happening? Okay, our "Horn" stands in or very near a corner. So the Corner will "load" the output just as it would load a sealed box or reflex Subwoofer or indeed a Transmission line. We will get about 6db acoustic gain from that at 30Hz or so Hz, enough to match the horn loaded upper Bass. Now, there are a few interesting additions here. If our horn avoids any sharp bends or folds (Carfrae, Hedlund) we have one resonator and the Horn. The additional harmonics of the resonator will cause problems similar to the TQWT of course, but will (hopefully) be somewhat alleviated by the Horn operation .So, decent Bass Extension, especially if the Horn vents (like seen in the Carfrae Horn) into a corner. But WHAT is happening in our folded horn… A huge can worms, crawly, slimy worms actually. Each individual sharp folded section will have a distinct pipe resonance. And harmonics. There will also be helmholtz resonances at relevant frequencies in the smaller (nearer the throat) sections. The whole results from this combination can be very unpredictable and messy. Try measuring the impedance some of the better folded horns and be look at the number of major resonance phenomena in the whole thing. Now again, such things as various additional Helmholtz Resonators can be employed to deal with the major problem modes (as suggested in the TQWT thread) and this is done in the Lowther Bicor enclosures and the Side Vivace (a modified "schmacks" horn). Anyway again from all said above it is clear that if we take care to have a complete mathematical analysis of all resonance modes present in the Folded Horn we might succeed in playing off some of them against others and overall achieve a pretty decent balance with a highly specific Driver and for pretty much solidly prescribed positioning. Or, like (I believe) the Edgar/Moth Horn for Reps and Lowther we can simply make a hyperbolic almost straight and very short horn that is a real 100hz or 90Hz horn and operates with a Subwoofer. Or we can of course take existing designs and use the "cut and try while depending on stupendous amounts of dumb luck" approach and hope for the best. To give some idea, the "Virtuoso" from Eric Thomas took about 2 1/2 years to develop to a degree that all major problems where dealt with. And the Virtuoso uses sections of shaped Polystyrene to form the Horn and has a minimum of bends (only one major one at about 90 degrees). Given the methods used and the basic design with already fewer problems than most (and a huge Box for on a good day 50Hz Low end) this shows some of the problems faced by those who try to solve these issues empirically. I can certainly now understand why Hiroyasu Kondo prefers different arrangements (like the Western Electrics Double bass Reflex) and the use of Drivers which do not require horn loading to balance out a weak low end over the classic rearhorn loaded high sensitivity wideband Driver. Kondo San always talks about "obedient sound". He means sound that is easy to control, easy to sculpture to "taste". This is clear in his Amplifier circuits. Now, front horns are fairly obedient (only fairly, they can still be messy), but systems like the DBR and conventional Bass Reflex are VERY obedient sound wise. Having meddled with anything from huge stage sound 4-way horn loaded PA's to micro monitors and Home speakers I too prefer obedient sound. Now folded rear loaded horns and TQWT Enclosures are NOT obedient. I wish those hoping to cut their own both the best of luck, success and would like to give a parting note. Remember, it is the journey that counts, not the destination at which you might perhaps never arrive. Have fun, but beware of the disobedient sound.... Tapered Quarter Wave Pipes (TQWP/TQWP)As for TQWT I perceive these as real problem childs. I have several Single Driver TQWT. Some of these DIY and at least two commercial ones (I think there was one more but I'm not sure it was a TQWT). Heard were variants of the "Everest" (straight 2m tall Pipe), TQWT using various Fostex Units, one or two Lowther TQWT and even one unit (also in the Everest format) using conventional Drive units 2-Way. Sorry. They all sounded terrible. The Bass on most was so crap that it defies description. Ill defined and lumpy. Certain Bass notes would be near inaudible, other would boom way too much. Up till today I have not heard with my own two ears a single TQWT that I would consider to be acceptable. So all bad news then? No. The TQWT is really good at a few things. It manages very well to combine a Horn and Transmission Line (and also some reflex loading). The main problem of the TQWT is that it will produce a comb filter effect due to line resonances at lower Frequencies. The only way to reduce these Line resonances is to stuff the line severely, which also elongates the line acoustically, but at the same time it really kills efficiency. The notches are caused by line harmonics canceling the Output from the Drivers front. They correspond to Impedance curve peaks which can be easily seen on a plot. Hence the advice "experiment". Now if really, really well optimised I suspect that the Helmholtz resonance of the port can be used to "suck" the energy out of the line at the worst one or two harmonics of the line. Also, the port work as lowpass attenuating the higher harmonics of the line. To get this to really work requires exceptional abilities in tuning or loads of cut & try sessions coupled with a load of dumb luck. The trick is to have a major port resonance at the lowest frequency desired (essentially the Drivers Resonance). The Pipe Resonance will likely be somewhat higher than that. The port then needs a second "parasitic" (pipe type) resonance at the second harmonics of the line resonance. This means the "parasitic" and normally unwanted resonance of a classic Port is desired. What it does is to simply remove energy from the Pipe at the 2nd harmonic. Normally this (2nd harmonic) resonance is out of phase with the Drivers front radiation and hence creates a cancellation, the source of the deepest and lowest notch. This is (I believe) why all the French TQWT designs I have seen are unstuffed and have a very defined port. If this is done well the result should work "by design". But all this obviously implies that a TQWT is a highly tuned Device that will work only with the driver it was designed for. The EXACT Driver. And as said, even some commercial manufacturers have gotten it wrong. Very wrong indeed. There is of course another option. Instead of going trough all the tuning simply use two (folded) TQWT in one enclosure, possibly even with different Drivers. One Tube should be tuned exactly 1/2 Octave below the other, once the minimal stuffing in the throat (the narrowest point in the tapered tube) is taken into account. If identical Drivers are used there is nothing more to it (apart from ideally use 12 Ohm or 16 ohm Driver so that that they when paralleled offer a 6 - 8 Ohm load to the Amp). If non-identical Drivers are used the Driver with the Lower resonance should be used in the Tube with the lower resonance. In any case, now one pipe will have a "peak" wherever the other has a Dip. This is well calculated will deal well with the resonances, leaves a much flatter impedance curve and a fairly flat frequency response. I know of two commercial Implementations of this Principle, have heard both and found the results pretty good. One company (Castle Acoustics UK) uses identical Drivers (Howard and Harlech Models also the top of the Line "Inversion" Speaker), but really only very bog standard cheap HiFi Driver fare. The Speaker is simply open at the bottom and is height adjustable for "fine-tuning" of the bass. Still, sensitivity is quite decent and the bass is fast, clean and pretty even. Another (Hoerning DK) uses different Drivers, with a Lowther front firing and various 12" or 15" Pro Audio Driver rear-firing, each driver with it's own Pipe and finally ending in a common outlet with very distinctive set of Ports. These Speakers also remove the Lowthers Whizzer cone (a major source for the very ragged midrange and the notable Lowther "Sting" and Honk) and add a small cone Tweeter above 6kHz.... Very nice I must say. I'm supposed to get a review pair "really, really soon now".... They where delayed as Tommy Hoerning decided to retune the port system.... Anyway, I hope that sheds some light on the TQWT and it's problems. BTW, if you think the classic rear loaded Horn looks better, wait till I take that one apart.... ;-) All I can suggest, if you can find the money (around $ 1,000 the pair) try finding a pair of Axiom 80 10" Drivers from Goodmans. If you have really a lot of the Green stuff locate a pair of Axiom 120 (I think) their almost never seen 12" Brothers.... These work in a reflex Box (albeit a largish one) and If you compare an Axiom 80 (Alnico Magnet, Cantilever suspension) against the latest Lowther there is no competition… The Lowthers gets wiped out on all counts. Crossover-less Twin Full-range Driver SystemA crossover-less twin driver system gives me an idea of something to play with for fun. Difficult to find a match, axioms are a good one. Thinking about it… Actually, finding a match isn't really difficult. The same principle could be done with a Lowther or Fostex, apart from the Fact that the Lowther has a much lower Xmax and hence more limited max. SPL. The "trick" goes like this: Driver 1:
Driver 2:
Using currently readily available Drivers I'd be tempted to try a Lowther DX2 in 15 Ohm (or Fostex FE208? is there a 15 Ohm Version?) with a Eminence Beta 12LT in 15 Ohm Version. Make the box around 150 Liter, no subdivisions, 30 to 40Hz or so tuning (this Vent is not so much a "Bass-Reflex", it is a resonator to limit excursion on the Drivers). It makes at low power for a Speaker system with very "obedient" sound. Unlike Horns or TQWT all parameters are easily controlled and conventional methods of managing resonances and the like work well. Power at 30Hz is limited to under 10W/8Ohm, but the whole Speaker should manage a real around 100db/W/m with a nice 8 Ohm Impedance. The Lowed is likely more limited than using the Goodmans Axiom Drivers I used, but should still go down into the lower 40's or upper 30's of Hz. Top
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