تبلیغات
برق. قدرت. کنترل. الکترونیک. مخابرات. تاسیسات. - سوالات عمومی در مورد سیستم صوتی و فواصل بلندگوها و نویزگیری و اندازه سیم ها و آستانه شنوایی3

برق. قدرت. کنترل. الکترونیک. مخابرات. تاسیسات.

دایره المعارف تاسیسات برق (اطلاعات عمومی برق)

 

Q21 - Why does SPL increase 6 dB for two drivers in parallel, when the electrical power consumed only increases by 3 dB?

A21 - The acoustic power Pa radiated from a small source at long wavelengths is the product of the radiating piston area Ap, the square of the piston velocity vp, and the real part of the piston's radiation impedance Zp:

Pa = (vp)2 Ap Re{ Zp }

Since the real part of Zp is also proportional to the radiating area Ap, it follows that the radiated power is proportional to the square of piston velocity and the square of piston area:

Pa ~ (vp)2 (Ap)

When two identical drivers are connected in parallel, each piston moves with the same velocity as the single driver, because the current through each voice coil is the same as before. The total radiating area has doubled, and the radiated acoustic power has increased fourfold (10 log (4) = 6 dB) over that of the single driver. The electrical power consumed by the two drivers has merely doubled (+3 dB). 
With 4x acoustic power for 2x electric power you have a 3 dB increase in power conversion efficiency.

The sound pressure p at some distance from the source is proportional to the square root of radiated acoustic power.

p ~ (Pa)1/2 ~ vp Ap

When the piston area Ap is doubled and the piston velocity vp stays unchanged, then the pressure doubles (20 log (2) = 6 dB). 

In summary, when two identical drivers are connected in parallel and driven with constant voltage, then twice the electrical power is consumed (+3 dB), the radiated acoustic power is increased by a factor of four (+6 dB), and the free space sound pressure level is doubled (+6 dB) at a given distance.  

Note that piston velocity and displacement are proportional to each other and both are directly related to the current through the voice coil. With two identical drivers connected in series, piston displacement and velocity decrease to 1/2, but the piston area doubles, which leaves the sound pressure and radiated power unchanged (0 dB) compared to a single driver connected to the same voltage. Electrical power dissipation is now 1/2 (-3 dB) and again the power conversion efficiency has doubled. 

A source that is small compared to wavelength and radiating 1 W acoustic power into free space generates a sound pressure level of 109 dB SPLrms at 1 m uniformly around it. The SPL is 103 dB at 2 m and 89 dB at 10 m. 
If the source has a power sensitivity of 86 dB SPL/1m/1W, then it takes 10(109-86)/10 = 200 W of electrical power to generate 1 W acoustic and 109 dB SPL/1m. The power conversion efficiency is 1W / 200W = 0.005 = 0.5%.
Two drivers with 86 dB SPL/1m/1W connected in parallel will produce 92 dB SPL/1m when driven with the same voltage, but consume 2 W. At 1 W input the driver combination generates 89 dB, and it takes 100 W to produce 1 W acoustic power, thus the efficiency is now 1%. 
The two drivers could be a tweeter and a midrange and have very different piston areas Ap. If they have the same 86 dB output for 1 W input, then the tweeter must have much higher piston velocity vp than the larger piston area midrange. The volume velocity U = vp Ap , though, is the same for both drivers when they have the same sensitivity. 
    For power efficiency of the PHOENIX see FAQ20 and for piston excursion requirements to obtain a desired SPL see Ref. 4 in Publications
  Top

 

Q22 - How would you describe the sound of the PHOENIX?

A22 - As very open, transparent and detailed like that of a good electrostatic. Un-colored. Very dynamic, fast and with such resolution that you can easily follow the melody of an individual instrument in a mass of others. Imaging is very realistic if well recorded. All this also applies to the bass. You might first think that the speaker lacks bass because it does not give any hint of its capability until there are low frequencies in the program material. The low end is in proportion to the rest of the spectrum, rather than droning on as with so many box speakers. The PHOENIX is designed to be played at fairly realistic levels. It will sound thin when played at low levels, as expected from the Fletcher-Munson curve.  The speaker will give you total involvement with the music when running at the proper sound level and you will not forget the experience. You might even learn to pay much closer attention to sounds around you, just like someone into photography becomes much more observant of the visual aspects of his world.

The sound of the PHOENIX is directly comparable to the Audio Artistry Beethoven-Elite and Beethoven-Grand. Both are 4-way active systems, with additional drivers for the frequency range below 250 Hz, and intended for larger rooms and even higher playback levels, or lower non-linear distortion when at the same volume level as the PHOENIX. In all other aspects, like timbre, resolution, imaging and dynamics the three speakers are almost indistinguishable. 
The Audio Artistry Beethoven, which is a bi-amplified, 4-way active/passive crossover speaker system and forerunner of the B-Elite and B-Grand, was voted best "Loudspeaker Product of the Year 1998" by the staff of Stereophile Magazine. It was reviewed in Vol.20, No.11, November 1997 and is included in the Stereophile Archives .  

You could listen to the Beethoven-Elite in John Field's living room. He operates the Tanglewood House B&B (+1-707-996-5021) in the California wine country north of San Francisco.
Top

 

Q23 - Can you point me towards people who have built the PHOENIX?

A23 - I know of a number of people who are in the process of building the PHOENIX and some who have completed the project. 
I invite anyone to share their experience and observations with me. If you allow me, I will publish your web or e-mail address here, so that others may see what you have done or get in contact with you. Thank you for passing on what you have learned, so that others may benefit from it.

1 - PHOENIX, Dave Reite , Oct. 2000
2 - PMTM1, Dave Reite, Dec. 2000   
3 - Phoenix Builders Website, Jim McDougall, Jan. 2001
4 - PMTM1 with active xo/eq, Steve Ashe, ashe@dim.com , Jan. 2001   
5 - Two-way with Diatone full-range, Markus Karner, Mar. 2001
6 - PHOENIX with 830500 woofers, LM3886 amplifiers, Steve Ashe, ashe@dim.com, Sept. 2001
7 - PHOENIX with heavy stone and slim steel stands, Douglas McKnight, douglas.mcknight@mcknight.to, Oct. 2001
8 - Digital Crossover Filter Construction, Don Maurer, June 2002
9 - PHOENIX and Dirt Cheap Dipole Loudspeakers, Stephen W. Moore, Oct.2002
10 - BOB, variation on a theme, Steve Dodds, Feb. 2003
11 - Dipole woofer, Charles Tidwell, ctidwel1@tampabay.rr.com , May 2003
12 - PHOENIX variation and many other dipole projects, Monte Kay, February 2004
13 - PMT1 on Bruce Coppola's website, February 2004

If you are looking for opinions and occasional facts on any loudspeaker subject, then participate in the Madisound Audio Discussion .  
The  topica dipoles forum is for DIYers building dipoles of all sorts. 
In the Planar Speaker Asylum you will find owners of dipole loudspeakers from various manufacturers discussing their merits. 
The hifi_dsp forum discusses digital crossover techniques. 
Top
 

 

Q24 - What is your process for designing an open baffle speaker?

A24 - I start with a set of drivers that I have extensively tested and measured. I listen to test signals, music and voice without baffle. The close-up frequency response, harmonic distortion, multi-tone distortion, burst distortion and volume displacement have been measured.

1 - Decide on a 2-way, 3-way or 4-way system depending upon output volume requirements, allowable size or intended application.
2 - Design and build a proto cabinet(s) with dimensions based on estimates of necessary baffle size, diffraction and aesthetics.
3 - Mount the drivers in the cabinet(s) and measure the free-space frequency response on-axis, off-axis, horizontally and vertically.
4 - If the response is not uniform, then change the cabinet dimensions. Repeat steps 3 and 4 until the response meets the target.
5 - Design and build prototype line level filters to equalize the drivers for flat on-axis response.
6 - Measure on-axis response and repeat steps 5 and 6 until target is met.
7 - Design and build prototype crossover line level filters.
8 - Measure on-axis response and repeat steps 5, 7 and 8 until targets for in-phase and out-of-phase response are met.
9 - Measure the free-space frequency response on-axis, off-axis, horizontally and vertically. If the result is not adequate, then go back to step 4.
10 - Listen to the speaker in mono using a wide range of program material and test tones.
11 - If the results in step 10 are acceptable, then build a second cabinet(s), otherwise go back to step 2 or 5.
12 - Assemble a second speaker.
13 - Measure the second speaker as in step 9.
14 - Build line level electronics for second speaker.
15 - Compare both channels of electronics to each other for identical measured performance.
16 - Set up both speakers in the intended listening space.
17 - Listen to the system with pink noise in single channel, dual mono and stereo for symmetry of sound. 
18 - Listen to a wide range of program material.
19 - Pause, rest your hearing, relax for a day or more.
20 - Listen to material which you have recorded with omni-directional microphones and with which you are very familiar.
21 - Pause, ask yourself how realistic the speakers sound.
22 - Make small adjustments to the frequency response if deemed necessary.
23 - Listen to a wide range of program material over an extended time period. Note if you get tired or want to hear more.
24 - Go to a concert. Listen to non-amplified sound, a symphony orchestra, jazz group, chorus, people talking, singing.
25 - Listen to your new speaker system and assess its capability.
26 - Sit back and enjoy or go back and start over again.
Top

 

Q25 - Why do you not use a rear firing tweeter or a dipole ribbon tweeter?

A25 - A rear firing tweeter for the PHOENIX would increase the speaker's high frequency 4p acoustic output by 3 dB for an even more uniform power response. In earlier designs I actually used such a tweeter. Because of the physical size and separation of the two tweeters - in terms of wavelength - it is impossible to obtain a simple cos(a) polar response. Instead, the off-axis response will consist of multiple beams, where the outputs of the two drivers see each other, which is primarily off-axis. On-axis, in front, the rear tweeter contributes very little to the sound. Overall I have not found a real benefit resulting from a rear tweeter. The off-axis response of the single tweeter-midrange combination is well behaved and actually has its widest dispersion around 4 kHz. Above that point the normal increase in tweeter directivity comes into play. Also, a rear tweeter may cause reflections from nearby objects which affects imaging and makes speaker placement critical. I am therefore using only a front tweeter.

A dipole ribbon tweeter would have the same potential problems as a rear tweeter. In addition the low 1440 Hz crossover requires large volume displacement capability. Typical high frequency ribbons would generate severe non-linear distortion under this condition.  A ribbon of greater length would narrow the vertical polar response too much. The large amount of air movement due to the close proximity of the 8" drivers might modulate the movement of the low mass ribbon and lead to further distortion. I have not used a ribbon driver, because the ones that I am aware of do not seem to fit this application. 

See the ORION+ and my AES paper for my latest understanding about the importance of a rear tweeter for making an acoustically small loudspeaker system less sensitive to room reflections, provided it is set up symmetrical and out in the room.
Top

 

Q26 - Why use active crossovers when passive crossovers require fewer components and amplifiers?

A26 - Active loudspeakers make very efficient use of amplifier power, allow drivers of vastly different voltage sensitivity to be used together, and give maximum control over the motion of each voice coil. 

A 2-way speaker with passive crossover, for example, may run into amplifier clipping on a big bass note. In this process the high frequency tweeter signal, which is riding on top of the bass waveform, gets chopped off in the amplifier, causing large amounts of very high frequency distortion. This may lead to the unexpected result that the tweeter overheats and is damaged by a 50 W amplifier, yet it would be unharmed if the speaker was driven by a 200 W amplifier. If an equivalent 2-way active system was driven by separate amplifiers with 25 W for the woofer and 25 W for the tweeter, then the woofer may well clip and produce distorted output, but the tweeter with its own 25 W amplifier would never know about it. Often a higher power amplifier is assigned to the woofer to avoid the clipping problem all together. A 100 W + 25 W 2-way active speaker is capable of much higher undistorted output, than a passive speaker driven with 125 W.

The drivers in a passive system should have about the same voltage sensitivity. It is the driver with the lowest sensitivity that determines how much the crossover has to attenuate the signal for the other drivers, in order to obtain a flat combined frequency response. Voltage sensitivity is wasted. Amplifier voltage or current swing need to be larger than in an active system to obtain a desired sound level. 
In the case of open baffle speakers, which have a response that rolls off towards low frequencies at 6 dB/oct, it becomes quite impractical to lower the sensitivity of all other drivers to such an extend that the response is flat to a useful low frequency corner. An active system can equalize the open baffle driver by adding frequency dependent gain ahead of the power amplifier.

Finally, in an active system each power amplifier is connected directly to the voice coil of a driver. This provides maximum damping and control of the voice coil motion, without the decoupling of capacitors, resistors and inductors of a passive crossover. The result is lower distortion.
Some people are concerned about the direct connection of a >100 W amplifier to the tweeter voice coil, fearing damage due to low frequency  turn-on and turn-off transients. Such fears are quite unfounded, because the slow transients cause little voice coil displacement below the driver resonance frequency. Of greater concern would be a permanent, large DC voltage at the output of the amplifier, but this can be easily checked with a voltmeter. A 200 mV DC output voltage would cause about 8 mW of constant power dissipation in the tweeter, which is insignificant. The greatest risk for damaging tweeters comes with swept sinewaves or MLS test signals of high power levels. The inaudible low frequency content can overheat the voice coil.

An additional benefit to active crossovers is the easy implementation of a wide range of filter functions using operational amplifiers for crossovers and equalization. An understanding of the transfer functions of drivers and cabinets used for a particular speaker can lead to the design of active networks that precisely compensate for frequency response variations and give the desired overall response in magnitude and phase. Since opamps separate and buffer each filter section the design and fine tuning of the circuit can proceed without interaction. Modern opamps, such as the OPA2134 are sonically transparent, and absolutely no cause for concern.

An understanding and the correct application of active crossovers and equalization, combined with appropriate choice of drivers and good cabinet design and construction, give the DIYer the capability to build loudspeakers with sonic performance that exceeds the best of what is commercially available. The additional components and amplifiers necessary for this task are a small price to pay when excellence is in reach.
Top

 

Q27 - How do I test the PHOENIX crossover/equalizer circuit board?

A27 - Dave Reite contributed the following method for testing the circuit board.

"Once you have your active crossover/equalizers constructed here’s a simple testing procedure using a PC based sound card spectrum analyzer. 

You’ll need two each, standard 3.5mm stereo mini plug / two RCA plug cables, and one each double female RCA adapter.  Connect one cable to your line out jack and the other to the line in jack on your soundcard.  Connect the two white (left) RCA’s together with the double female adapter.  The red (right) plug from the line out jack goes to the Phoenix crossover input and the red (right) plug from the line in jack goes to the selected crossover output.

Download a demo copy of  SpectraPlus . This is a 30 day fully working copy of the best soundcard based spectrum analyzer program I have found. Once you have the software installed, set it up as follows:

Mode:  realtime
View:  spectrum
Options:  settings:    Sampling format:  16 bit stereo
Dual channel options:  Real transfer function (right/left)
Smoothing:  Hanning
Utilities:  Signal generator:  White noise

Note:  If your sound card doesn’t support duplex operation, you’ll have to supply your own white noise source from something externally.  (A test CD with a white noise track would be a good option.)  A “Y” adapter to apply the signal to the crossover input and the white (left) plug from the sound card line in cable is all that’s required.

In the spectrum window set “plot top” to 20,  “plot range” to 60, “averaging” to 5, and verify “peak hold” is unchecked.

Hit the RUN button at the upper left and within a few seconds you should see a very smooth curve of the output you’re measuring.  Repeat as necessary on all the outputs and settings of the Phoenix crossover to verify its operation.

Check the picture of SL’s MLSSA measurements for reference. Your results should look identical.

I hope this helps. Comments/questions are appreciated.

Dave Reite  - daver@neptune.kpt.arl.psu.edu ."
Top 

 

Q28 - How high in frequency can you push the dipole woofer?

A28 - The shorter you make D, the distance between front and rear openings of the PHOENIX and H-frame woofers, the higher the frequency for which the rear wave adds in phase to the front wave, and the unequalized dipole output reaches its first 6 dB peak.
Unfortunately, there is a resonance that comes into play before this peak. The cabinet structure between the driver cone and the cabinet opening forms an acoustic transmission line. If its length L ~ D/2 is large compared to the opening width and height, then there is a resonance when L = 1/4 wavelength. This is due to the acoustic impedance mismatch of the transmission line at the opening and at the cone.

The resonance peak in the H-frame and PHOENIX W-frame occurs at lower frequencies, because the acoustic structure in front and behind the cone is more complicated than a simple transmission line.
The H-frame woofer with D=12" has a physical length L of 6" for this "transmission line". This should give a 1/4 wave resonance at  f = 13500 in/s / (4 x 6") = 563 Hz. The measured peak is more like 240 Hz. Thus, the effective acoustic length is about (563 Hz / 240 Hz) x 6" = 14". The reason for the difference lies in two places. 
One, the physical line is very short compared to the opening size and an end correction must be added which effectively extends the length of the line. 
Two, the driver frame traps air behind the cone so there is added mass that lowers the transmission line resonance.

For example, the D=16" H-frame woofer with the Peerless 830500 drivers (photo) should have a resonance at (6" / 8") 563 Hz = 422 Hz, but measures with a response peak at 200 Hz. 
The Phoenix woofer with W-frame and D=19" should have a 1/4 wave resonance at 356 Hz but measures 270 Hz. In a similar cabinet with the same D but using 830500 drivers the resonance is at 190 Hz.  Again, these are acoustically complicated structures. Exactly relating the physical length L to the acoustic length and resonance frequency only works if the transmission line is long compared to its cross-section. In practice it is best to measure the frequency response in the cabinet opening plane.

In all cases the resonance occurs below the frequency for which a dipole
with a given D would have its 6 dB peak, i.e. when the rear wave adds in phase to the front wave, because D = 1/2 wavelength. In the close-up frequency response measurement at the opening of the woofer cabinet this 6 dB peak cannot be seen, because the microphone senses only one side of the woofer. It gives the infinite baffle response without any 6 dB/oct dipole roll-off.

Decreasing D moves the 1/4 wave resonance to a higher frequency, and allows for a higher crossover frequency to the midrange. But, it also reduces power handling of the woofer, since now larger cone excursions are required to reach the same SPL as with a larger D.
Top

 

Q29 - How do you measure cone excursion of a driver?

A29 - A simple wedge micrometer that is cut out of paper can be used on larger drivers to measure their peak-to-peak cone excursion. It is a rectangular triangle with 2:1 sides and a scale at its base line. Attached to the dust cap of a driver cone it moves back and forth. The eye cannot follow the movement, but sees a diagonal line that intersect the scale graticule. The axial movement is translated to a sideways displacement of the diagonal intersection and gives a twofold magnification of the excursion. While this is not a precise measurement instrument it is still very useful in relating acoustic distortion data to cone displacement.
Top

 

Q30 - Can a dipole woofer be placed in a room corner?

A30 - A dipole placed in a room corner should provide minimal excitation of room modes or resonances. The corner is a region of maximum sound pressure and it is well known that a typical monopole woofer in this location will couple optimally to all room modes. A monopole woofer would be a high impedance pressure source in the high impedance region of all wave modes (Z = p/v) and be optimally adapted to support them.  A corner dipole, being a velocity and low impedance source, would have minimal coupling to room modes.
Unfortunately, the H-frame and W-frame dipole woofers described for the PHOENIX cannot be placed completely into a room corner, because it would block their rear radiation. A minimum distance of 3 feet (1 m) must be maintained.
L. E. Selmer (larryeselmer@netscape.net) proposes a corner dipole woofer that uses a different baffle arrangement and makes full use of a room corner. He claims that the sound from his first prototype shows exciting promise. I post his idea here for others to experiment with. I cannot implement it in my own room, because I have no clear corners. It should also be pointed out that the midrange panel needs to be placed near the woofer so that the distances from the listening position to the woofer and midrange are different by less than 1/8th of a wavelength at the crossover frequency. This might only work acceptably in fairly narrow rooms unless the signal to the midrange is electrically delayed.
Top

 

Q31 - Is there an optimum room placement for a dipole?

A31 - Yes, there is, from a sound reflection and also from a standing wave point of view. Both are a consequence of the figure-of-eight or cos(angle) free-space polar response of a dipole with its opposite phase front and rear radiation and very low output at 90 degrees off-axis. 
Assume the
dipole speaker is placed at the same distance from the front wall as the listener has to the wall behind him. The negative polarity sound from the dipole's backside is reflected off the wall behind it towards the listener. The sound from the front side of the dipole travels directly to the listener and also to the wall behind him where it is reflected back towards him. The rear wall reflection reaches the listener at the same time as the reflection off the wall behind the speaker. The two reflected sounds have traveled the same total distance for this particular speaker and listener  setup. The two reflections cancel each other, because they are of opposite polarity. (Jorma Salmi, "Dipole source placement in a room", 92nd AES Convention, 1992, Preprint 3327)
A diagram helps to clarify the case. It shows the plan view of a rectangular room with dipole D and listener L. The four hard room boundaries can be removed and replaced by four image sources D1 through D4. The special setup, where a = b, does not help with reflections 3 and 4 off the side walls. Moving the speaker away from the side wall reduces the strength of the reflection, because the radiation goes to an angle where the dipole output is attenuated relative to the on-axis level. In practice it may be difficult to move the speakers far enough into the room to satisfy a = b, but keeping them at least 1 m (3 ft) from the rear wall is recommended.
Occasionally a cardioid free-space radiation pattern is suggested for a speaker, because it would radiate, like a dipole, 4.8 dB less power than a monopole for the same on-axis SPL. Unlike a dipole, the cardioid radiates weakly towards the rear. Thus, there is little reflection off the wall behind it. But, there is no cancellation when a = b. In addition there will be stronger reflections off side walls, floor and ceiling. 
    It is important to understand that the discussion up to this point has only dealt with the first reflection off a single room boundary. The next order of reflections involves both front and rear walls and the image model would have to be expanded with additional sources. Successive reflections always occur and they lead to the gradual build up of stored energy in the form of room resonances or modes. Thus the cardioid speaker, even though it does not radiate towards the rear, excites a strong longitudinal mode between front and rear walls, similar to the dipole. The dipole, though, excites side-to-side and floor-to-ceiling modes only weakly, because they propagate along the null-axis of its radiation pattern. Furthermore, by angling the dipole its coupling to specific modes can be changed, whereas cardioid rotation has considerably less effect. 
The cardioid radiation pattern is the sum of a dipole and a monopole and its in-room behavior lies between the two constituent types of sources. It has the reduced total power output of the dipole into the reverberant field above the Schroeder frequency. It lacks in adjustability of coupling in the discrete mode frequency range where it behaves more similar to the monopole. The low frequency response rolls off at 6 dB/oct like for a dipole and has to be equalized. Unlike monopole and dipole the cardioid produces no first order reflection from the wall behind it.
    The dipole has minimum room mode excitation when it is placed near a pressure maximum (= velocity minimum) of a mode and when its axis of radiation is not aligned with the direction in which the mode travels. Pressure maxima are at the room boundaries and room corners. A dipole woofer should be placed near the side walls, provided the distance from woofer to listener is nearly the same as that from the midrange. Again, most likely practical limitations will have to guide best dipole speaker placement for room modes as well as for first reflections off room boundaries.
Top

 

Q32 - How would you increase the output capability of the Phoenix?

A32 - The sound output of any speaker has reached its limit, when the distortion becomes objectionable. It may sound too loud and intolerable, even when the sound pressure level is not exceedingly high, because we unconsciously judge loudness by distortion. A clean speaker can play at very high SPL and you may not be aware of it, until you try to talk to someone and find yourself shouting. Before contemplating changes to the PHOENIX I would recommend to build it as designed and live with it for a while. It can play very loud. If you think you need more, then consider the following paths for upgrading during a second construction phase.

The first step for reducing distortion in the PHOENIX would be a change from the 1252DVC/X6100 woofer driver to the  XLS 830500. Next would be a change of the 100 Hz crossover from 12 dB/oct to 24 dB/oct, which reduces the excursions of the 21W/8554 midrange drivers at low frequencies, but makes woofer placement and integration more critical because of reduced frequency overlap. The steeper crossover also reduces the effective source size in an important part of the spectrum and makes the sound more subject to room effects. Finally, the number of woofers could be doubled. 
I would recommend these approaches rather than going to a  4-way system like the Audio Artistry Beethoven-Elite or Grand, because it will be difficult to realize their full potential with the tools and working knowledge typically available to the DIYer. I think the ultimate performance of the 3-way PHOENIX will be quite comparable or even superior, if the 4-way design is not executed perfectly..

Changing to the XLS 830500 for the dipole woofers requires the addition of a single WM1 or MT1 printed circuit board. The equalization is described in the documentation that comes with the pcb.

Changing to a 24 dB/oct LR4 crossover is more involved, but not too difficult for anyone who has modified a printed circuit board before and understands something about filter design. The PHOENIX pcb has three notch filters for room equalization. Two of these can be changed to a 12 dB/oct highpass and a 12 dB/oct lowpass section to be wired into midrange and woofer channels. In addition, the 90 Hz 1st order highpass filter section in the midrange channel must be changed to a 2nd order highpass to obtain the other half of the 100 Hz, 24 dB/oct highpass filter. All this must be done while preserving the dipole roll-off compensation in the 10-300LP and the 90-500LP stages. The 90-500LP boost must be extended to 30 Hz to minimize its influence upon the 100 Hz crossover. 

Alternatively, unmodified circuitry from a single MT1 pcb can be patched into the PHOENIX midrange and woofer channels. This solution for obtaining a 24 dB/oct crossover is detailed in the MT1 pcb documentation. The same MT1 pcb can also provide the additional equalization that is needed for the XLS 830500 woofer.
Top

 

Q33 - Is a pre-assembled crossover/equalizer available?

A32 - Loading and soldering all the required electronic components onto two Phoenix printed circuit boards takes me about 8 hours. It is not difficult to do, but requires attention at every step. I do not sell pre-assembled boards, but I can put you in contact with someone who will load boards. I can check your assembled boards for proper functioning, should you want to have this assurance, in case you lack the necessary equipment to test them yourself. The cost for this service, including correction of minor errors, is $50 plus return shipment. As part of this test you will receive the measured frequency response for each output and a check list.
Mail the assembled circuit boards to:
Linkwitz Lab
15 Prospect Lane
Corte Madera, CA 94925
USA
Top

 

Q34 - What is the optimum Qts for the drivers of a dipole woofer?

A34 - The low frequency roll-off of a woofer and its associated group delay are optimal, from what I have observed, when they follow the response of a 2nd order highpass filter with Q = 0.5. When a driver is mounted in a dipole W-frame or H-frame its mechanical resonance frequency Fs decreases to Fd, due to air mass loading, and Qts increases by a similar percentage to Qtd. 
For example, a driver with very strong motor, Fs = 18 Hz and Qts = 0.2  might have Fd = 16 Hz and Qtd = 0.22 as determined from an impedance measurement of the baffle mounted driver. With Qtd < 0.5  the low frequency behavior of the woofer is characterized in the complex s-plane by real axis poles at -69 Hz and -3.7 Hz and by 3 zeros at the origin. One of these zeros is due to the front-to-back dipole cancellation with its 6 dB/oct  low frequency roll-off. The frequency response of this 3rd order acoustic highpass filter must be equalized to obtain a flat response. A suitable target response could be Fd = 20 Hz and Qtb = 0.5. It is easily realized with two shelving lowpass filters. The first filter with a pole at 20 Hz and a zero at 69 Hz corrects for the low Qts of the driver. The second filter with a pole at 20 Hz and a zero at 400 Hz compensates the 6 dB/oct roll-off due to dipole cancellation. The 3.7 Hz pole is low enough in frequency so that the response is dominated by the 2nd order roll-off below 20 Hz. An advantage of a low Qts driver is the ease with which it can be equalized for an optimum response with Q = 0.5. 
A driver with a smaller motor might give Qtd =  0.7 and Fd = 20 Hz, which leads to a pair of complex poles in the s-plane. This can be readily changed to a 3rd order Bessel highpass response by using a shelving lowpass filter  with a pole at 20 Hz for the necessary dipole roll-off compensation. 
Likewise, if Qtd = 1 and Fd = 20 Hz, then an additional pole at 20 Hz, from the dipole equalization, leads to a 3rd order Butterworth acoustic highpass response. Third order filters introduce more group delay than 2nd order ones. It is therefore advantageous to use Qts < 0.5 drivers, even when they require driver roll-off equalization in addition to the normal 6 dB/oct dipole correction.
Top

 

Q35 - How does the new ORION compare to the PHOENIX?

A35 - The ORION evolved from the PHOENIX out of a desire for a smaller size speaker, one that is acceptable in many domestic situations. The PHOENIX has peak output volume capability that is just not needed in many homes, especially in living rooms of less than 400 ft2.  Also, I had been searching for drivers that might match or exceed the performance of those used in the PHOENIX in terms of linear and non-linear distortion. More recently I had convinced myself that a LR4 crossover in the 100 Hz region has inaudible group delay distortion, which allows me to use a single 8" driver without sacrifice in output capability. Some ideas about softening the cabinet appearance worked out nicely acoustically. Combine all this and and more and you have the ORION
As it turned out, I believe that this speaker is the most sonically refined of all my designs. So much so that I do not need the physically much larger Beethoven-Elite for my personal listening enjoyment. Yet, I have not given up the satisfaction that comes from reproducing near realistic sound levels and dynamics. The ORION does all I want for the type of listening that I do in terms of frequency response, time response, spatial imaging and amplitude range, and with superb resolution in each domain. It draws me into the music better than any loudspeaker I know.  
If this speaker were used for Home Theater I would add THOR subwoofers below 40 Hz with a LR4 crossover, but not for 2-channel music listening. The ORION reproduces the acoustics of the recording venue so well that I have turned off my surround speakers for ambience recovery much of the time. 
Top

 

Q36 - Are there better drivers for the PHOENIX?

A36 - There may be, but I would have to ask, better in what respect? I investigated alternatives for the Scanspeak 21W/8554 midrange and D2905/9700 tweeter by testing for non-linear distortion and stored energy. This was not an exhaustive search, but I went after what I considered most promising drivers, based on various factors that I knew from experience. I did not investigate ribbon tweeters, since I have no evidence so far that they could reproduce sound more accurately than some dome tweeters. The investigation of alternate woofers turned out to be very beneficial, because I found in the Peerless XLS 830500 not only a lower distortion and larger volume displacement driver than the 1252DVC and X6100, but these drivers are also no longer available. There are now probably additional 12" drivers on the market that would perform equally well. I have not investigated any further. Some have short voice coils and long magnet gaps, which results in very low distortion, as long as the required cone excursion does not force the coil to the edge and out of the gap.  
The SEAS Excel midranges and tweeter had slightly better measured performance in the parameters I tested for, than the Scanspeak units. I did not think the difference was worth redesigning the PHOENIX and, instead, I chose to try out these drivers in a new design concept, the ORION, rather than using the SS drivers again. As it turned out, the ORION is an improvement over the PHOENIX and the difference is greater than what I would have expected from the measured quantities. But there are other differences besides the drivers, which I feel contribute to this: The driver layout is M-T rather than M-T-M, the baffle and driver mounting are more open, the woofer to midrange crossover is different and the woofer is integrated. 
I have no intention to go back to the PHOENIX and to change its drivers. It is in my opinion still a good speaker, and capable of serving a slightly different purpose than the ORION. 
Top

 

Q37 - What cables and interconnects do you recommend?

A37 - I prefer not to recommend any specific product. Cables can have audible effects and some manufacturers make sure they will, either through unusual electrical parameters and/or by suggestion. Weaknesses in the design of the output-to-input interface are exploited. Sounding different does not mean it is also a more accurate transfer from electrical to acoustic domain.
My guideline for speaker cables is to keep their resistance to less than 0.1 ohm for the roundtrip path of the current. This defines the maximum length of a 2-conductor copper cable for different wire gauges.

Wire gauge Max. length in feet
18 8
16 12
14 20
12 30
8 80

I measured the 16 gauge Megacable from Radio Shack (278-1270) that I use. A 10 foot length has 0.07 ohm resistance, 714 pF of capacitance and 1.9 uH of inductance. The line impedance is 51 ohm. A typical tweeter has a voice coil resistance of 4.7 ohm and 50 uH inductance. At 20 kHz this yields an impedance of about  |4.7 + j6.3| = 7.9 ohm. Add to this the cable inductance of j0.24 ohm, and 0.07 ohm resistance for 10 feet, and the impedance becomes 8.09 ohm. This causes a 7.9/8.09 = 0.98 or 0.17 dB reduction in tweeter output at 20 kHz which is insignificant.  The cable effect is even less at lower frequencies.

Speaker cables can act as antennas in the AM frequency band and may cause distortion in the output stage of a solid-state amplifier, if strong radio frequency signals are present. In particular, the cable capacitance in conjunction with the inductance of a driver voice coil may form a resonant circuit for these frequencies. The resonance can be suppressed by placing a series R-C circuit of 10 ohm/2 W and 0.33 uF/100 V across the cable terminals at the speaker end. 
    Coaxial interconnects with phono (RCA) plugs tend to pick up radio frequencies in the FM band. The currents that are induced in the cable shield must not be allowed to enter the inside of the coax. This requires a very low resistance connection between the outer conductor of the phono connector and the chassis (signal ground) of the equipment that it plugs into. The continuity and low resistance of the shield is also very important for hum and buzz currents, so that they will not induce a voltage on the center conductor. The technical description for this is the Transfer Impedance of the cable and connectors, which must be in the low milli-ohm range. Unfortunately I have not seen this specification used by the audio industry. An excellent description of the theory and treatment of hum and buzz problems in equipment setups with mixed two and three prong AC plugs is given in AN-004 by Jensen Transformers, Inc. I have not found balanced interconnections to be necessary for the high level circuits past the preamplifier. But sometimes it requires to experiment with AC outlets in different locations to reduce to insignificant level the buzz that one may hear with the ear close to the speaker cone. So, when choosing a coaxial audio interconnect look for good mechanical construction, direct contact between shield and connector, and well plated contact surfaces.  
I find what is needed at Radio Shack. I solder speaker cables to terminal strips on the speaker end and use dual in-line banana plugs on the amplifier end.
Top

 

Q38 - How do diffraction effects show up?

A38 - This question is best answered by looking at some actual frequency response measurements, though this does not answer to what extend diffraction is audible. Go to the Diffraction from baffle edges page. For an explanation of the physics involved see FAQ 8.
Top

 

Q39 - How much power does it take to drive a dipole woofer?

A39 - It is often assumed that excessive amounts of power are required to drive a dipole woofer, because of the necessary 6 dB/oct acoustic roll-off equalization and additional boost required for drivers with Qts below 0.5. In fact, equalization does not change the power requirement. It is a driver's cone excursion for generating a desired SPL which determines how much output voltage and current an amplifier must be able to provide. I will illustrate this with an example example for two drivers, a Peerless 10" XLS 830452 and a Seas W26FX001, when driven from amplifiers with 180 W and 60 W into 8 ohm specification. Go to the Power amplifier limited SPL for a dipole woofer page to find the results. See also a related analysis of SPL limits for a closed box woofer .
Top

 

Q40 - Can the PHOENIX be build with passive crossovers?

A40 - Practically speaking, the PHOENIX cannot not be driven from a single stereo power amplifier even if you had succeeded in designing a passive crossover filter that applies the same voltages to each driver as shown by the frequency response curves A, B and C of xo_eq2.jpg and the associated non-minimum phase response for C. The problem resides in the large amount of equalization that is required for dipole operation. For example, between 30 Hz and 400 Hz the amplifier power that is available to the woofer and midrange differs by 20 dB or by a factor of 100. Thus if you used a 100W amplifier to meet the low frequency needs, then only 1 W would be available around 400 Hz. That clearly is not sufficient. A wasteful 99 W is dissipated as heat in the resistors of the crossover. The obvious solution is to use a dedicated power amplifier for the woofer with an active crossover/equalizer between woofer and midrange. This leaves a passive crossover between midrange and tweeter. Now  the drive voltage to the midrange needs to be attenuated by up to 14 dB or a factor of 5 in the crossover/equalizer. Again, this is wasteful of amplifier power which would be needed for the midrange. The PHOENIX prototypes exemplify the tradeoffs when using passive crossovers and equalization.

These are reasons for going with line level crossovers/equalizers: 

  1. Effective use of amplifier power. It takes more amplifiers, but these can be of lower power.  For example, the ORION uses one 60 W amplifier for each driver and has a very wide dynamic range. 
    Many more speaker design approaches become feasible by using line level filters.
    Even for a 2-way speaker with a single power amplifier it is advantageous to use line level equalization for the baffle step compensation (4
    p to 2p radiation). It is a more efficient use of available amplifier power to cut the high frequencies at line level, than to attenuate the highs by dissipating power in a passive crossover. At the same time the low frequency end can be extended by a line level equalizer.

  2. Less distortion at high sound levels. The woofer amplifier may clip, but the tweeter does not see the high frequency components that are generated. Distortion may not even be noticed. Compare that to a passive crossover speaker where the clipping of the power amplifier at low frequencies is not only highly audible but can also destroy the tweeter. In such case a larger power amplifier should have been used. 

  3. The power amplifier is directly connected to the voice coil of the driver and exerts maximum control over its motion. In a passive crossover the impedance, when looking back from the driver into the crossover, becomes progressively higher as the stop band frequency is approached. Also any equalization carries with it an increase in impedance and consequently a reduction in control. Usually the first impression of a well executed active crossover/equalizer is one of greater clarity.

  4. With a passive crossover any change in voice coil impedance affects the network transfer function's magnitude and phase and causes a change in acoustic frequency and polar response. The driver impedance may change and recover slowly due to heating (+0.4% per degree C) at high signal levels, or change instantaneously due to large cone excursion and non-linear motor behavior. Such changes in impedance have no influence on the filter function of a line level crossover/equalizer and their acoustic effect is reduced.

Sometimes concern is expressed over the use of operational amplifiers in line level filters and possible deterioration of the electrical signal. This can be completely overcome in my experience by careful signal level flow design and choice of excellent operational amplifiers, such as the OPA2134. My concern is with commercially available "electronic crossovers", both analog and DSP based, which may not have sufficient adjustability for the PHOENIX together with distortion free dynamic range. The finite number range of a DSP unit must be properly allocated to the signal demands of different filter stages at their location in the digital signal path. 
The PHOENIX printed circuit board together with the MT1 and WM1 printed circuit boards offer full flexibility for implementing all sorts of different active line level crossovers/filters.
Top

صفحات جانبی

نظرسنجی

    لطفاً نظرات خود را درمورد وبلاگ با اینجانب در میان بگذارید.(iman.sariri@yahoo.com)نتایج تاکنون15000مفید و 125غیرمفید. با سپاس


  • آخرین پستها

آمار وبلاگ

  • کل بازدید :
  • تعداد نویسندگان :
  • تعداد کل پست ها :
  • آخرین بازدید :
  • آخرین بروز رسانی :