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| Build-Your-Own | Main Panel | Dipole Woofer | Crossover/EQ | Supplies |
| System Test | Design Models |
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System Test

This page gives you measured frequency response data for the PHOENIX system. I also added a few pointers for constructing a tone-burst signal source and a receiver with microphone and fast peak meter for those of you, who would like to build some versatile acoustic test equipment.

I must point out that steady-state frequency response curves - magnitude and phase - present only a limited view of a loudspeaker. Hidden in them, and often not discernible to the eye, are further linear distortion data. These are better seen in shaped tone-burst measurements in the time domain. Also, non-linear distortion, which adds its own signature via new spectral components, that were not present in the original, must be made visible with single and multi-tone test signals. 
    Stored energy and non-linear distortion measurements were performed on a variety of drivers and influenced the selection made for the PHOENIX. The 9800 tweeter exhibits increased decay times above 12 kHz due to the trapped air between dome apex and plastic cup forming a mechanical spring-mass system. Removing the cup rapidly rolls off the response (-8 dB at 20 kHz). The D2905/9700 soft dome tweeter is, in these respects, a more accurate transducer and can be readily substituted. Regardless, the metal dome tweeter was initially chosen for this design based on a slight listening preference. I have since gone back to the 9700 as the more accurate tweeter. 
    Frequency response must also be evaluated by how it might generate resonances, reverberation and reflections in the playback venue. 
    And finally, the eye and the ear do not respond in the same way. Often too much, in terms of audible significance, is read into the wiggles of a graphical presentation of the response from a single microphone to the sound pressure at a single point in space. On the other hand, more subtle details and broad trends are often overlooked. Our ear-brain system has its own value scale for sound perception, one that has been honed over a few millennia by what is important for the survival of the species.

With that precaution in mind, here is a limited set of data describing the frequency response of the PHOENIX system.

 

Crossover / equalizer

The crossover/equalizer corrects for trends in the frequency response. No attempt was made to deal with the remaining fine grain structure in the response, which is primarily caused by diffraction effects. Those were minimized by the shape of the main panel (FAQ8). The drivers were selected for their inherent smooth frequency response, in addition to their low distortion and volume displacement capability.
1
- Buffer output, woofer, midrange, and tweeter drive signal frequency response. 

Woofer and tweeter level adjustments are set to midpoint (12:00 o'clock).  x1.gif
       Note 1 - If you wanted to build the PHOENIX with passive crossovers, i.e. inductor, capacitor, and resistor networks between power amplifier and driver terminals, then you would have to replicate the above frequency response curves - magnitude and phase (non-minimum!) - for the woofer, midrange and tweeter terminal voltages when measured at the output of the corresponding filter network loaded by the driver. This is not a trivial design task. Even if you came close with this part, there would still be another difference, because of the impedance seen by the drivers. In the active crossover/eq case each driver looks back into the very low output impedance of the power amplifier. This provides voltage control over the voice coil motion at all frequencies. In the passive case each driver sees the highly frequency dependent output impedance of a filter network. At larger signal levels, where driver non-linear distortion tends to set in, the passive and active systems will have sonic differences due to the impedance differences which affect the flow of distortion currents. 
        Note 2 - It would be interesting to build the PHOENIX with current control of the drivers to investigate further distortion reduction. I have observed that the distortion products in the acoustic output spectrum of a driver are highly correlated with the distortion components of its voice coil current waveform, when driven from a voltage source. Using feedback it should be possible to linearize the voice coil current and thereby reduce distortion to some degree. The driver frequency responses would have to be re-measured under current drive conditions and the crossover/eq redesigned accordingly. 
2 - Crossover/eq frequency response curves as above but without 100-200 Hz shelving high-pass filter and 2760 Hz notch filter.   x2.gif

 

Main panel

The main panel reproduces a frequency range of about 8 octaves, 4 by the midrange and 4 by the tweeter.
1 - The impedance seen by the midrange amplifier is about 3.5 ohm and 4.5 ohm by the tweeter amplifier. The amplifiers selected must be able to supply the required current without limiting and have a low output impedance. Connecting the 8554's in series would have given a 14 ohm impedance and would require large voltage swing capability from the amplifier, i.e. a high power rating.  p1.gif
2 - The following data were taken outdoors with the tweeter 108" (2.74 m) above the ground and the microphone 52" (1.32 m) in front of the tweeter to approach an echo free measurement setup. 
In this first measurement the relative output of the drivers is seen. The midranges are about 12 dB higher than the tweeter. The difference is easily corrected with different gain settings in the active xo/eq channels. 
    For a passive crossover the midranges would have to be connected in series to drop their output 6 dB. The remaining difference would have to be dissipated and reflected by the passive xo. 
    The measured response curves are the result of driver response, panel diffraction and proximity of drivers to each other. The periodicity of about 1 kHz in the tweeter response is due to the panel width.  p2.gif
3 - The horizontal off-axis response of the midranges below 400 Hz shows the typical dipole cos(angle) behavior, e.g. -3 dB at 45 degrees and -6 dB at 60 degrees. 

Around 700 Hz dispersion widens due to the phase of the rear radiation having been modified by the basket and rear cavity acoustic low-pass filter. It also causes the bump at 500 Hz and the roll-off at 6.9 dB/oct instead of 6 dB/oct. Above 2 kHz dispersion narrows, but the drivers are not used in this frequency range.  p3.gif
4 - Horizontal off-axis response of the tweeter on the panel. Widest dispersion is at 4 kHz. It narrows above this frequency and shows response peaks around 16 kHz. These are due to the cup in front of the dome. 
    Caution!! A measurement like this, without any high-pass filter, can easily damage or destroy the tweeter and should be performed with adequately low amplifier power. The only safe test signal is the shaped tone-burst. It can actually be used at very high amplifier output voltage swings to hear the onset of clipping distortion and to determine the maximum SPL capability of a driver.  p4.gif
5 - Response of the 1400 Hz crossover between midrange and tweeter using the active xo/eq. There is about a two octave wide region of overlap between the drivers where both contribute to the total sound output.  p5.gif
6 - The off-axis response of the equalized main panel is dominated by the midranges at the low end and the tweeter at the high end. Note that the dispersion follows essentially dipole behavior up to 1400 Hz, then becomes wider before finally narrowing above 6 kHz.  p6.gif
7 - Effect of the 2760 notch filter upon the frequency response.  p7.gif
8 - Horizontal off-axis response with the notch filter in place.  p8.gif
9 - Off-axis response measured at a larger 86" (2.18 m) distance from the tweeter shows smooth, well controlled behavior of the panel.  p9.gif
10 - The vertical off-axis response above and below the tweeter axis is dominated by the overlap region between midrange and tweeter and the relative spacing of the three drivers to each other. The data was taken with the microphone at 52" (1.32 m) in front of the panel and the panel being lowered by 4.5", 9" and 14" to increase the angle from zero to 5, 10 and 15 degrees. 
        Note the relatively wide dispersion of the system even in the vertical plane. Never-the-less, the tweeter height should be at or slightly above the ear when seated. For psycho-acoustic reasons that I do not fully understand, one can always tell when a speaker's acoustic center is lower than ear height. Tilting back the speaker, as is done for some products, will not correct that impression. We seem to use the floor reflection even to localize a virtual image. It immediately sounds unrealistic to me when the orchestra's sound comes from below.  p10.gif

 

Dipole woofer

The woofer reproduces a frequency range of about 2 octaves.
1
- The impedance seen by the woofer amplifier is greater than 6.5 ohm and includes the driver resonance in the cabinet. Some expensive tube gear has difficulty driving such load. Only solid-state amplifiers should be used.  w1.gif
2 - The following data were all taken outdoors on a concrete floor. 
First, the acoustic output at the front opening plane of the floor mounted cabinet. No dipole diffraction effects are seen, because the microphone is too close to the source. 

The response follows what one would calculate (Ref. 12) from the impedance curve for a closed box woofer with a -3 dB point at 13 Hz for Qts = 0.7. It is also the response of the dipole woofer in the far-field, after the +6 dB/oct equalization has been applied.
The 270 Hz peak and the following dips and peaks are caused by the acoustic impedance mismatch between the transmission line inside the cabinet and the front opening into half-space. The effectiveness of the 290 Hz notch filter in attenuating the first peak can be seen.  w2.gif
3 - Response at the same location as above, but with 100 Hz crossover low-pass filter and +6 dB/oct dipole equalization applied.  w3.gif
4 - The crossover between woofer and midrange as measured at a horizontal distance of 96" (2.44 m) from the panel and with the microphone on the ground. The tweeter height is 39" (0.99 m). The measurement is influenced by reflections from nearby objects and by ambient noise as seen in the lower traces. The SPL from the midranges is doubled at the ground plane relative to free space.   w4.gif
5 - When the microphone is raised to 39" the midrange output drops about 6 dB on average, after smoothing out the floor reflection oscillations in the response. By applying the 100-200 shelving high-pass filter the woofer and midrange levels are gradually lowered relative to the midrange at higher frequencies. If merely the woofer level was lowered by reducing its xo/eq gain by 6 dB, then a step in the frequency response around 100 Hz would result, because the transition from half-space to full-space sound propagation is too abrupt.  w5.gif

 

Microphone

A high quality measurement and recording microphone with flat frequency response could be constructed using the Panasonic omni-directional back electret microphone cartridge WM-60AY. It was priced at $23.41 for 10 units(!) under Digi-Key part number P9959-ND, but is now obsolete. A replacement is the WM-61A (P9925-ND).
When the cartridge is connected as described by Panasonic it produces fairly high distortion at moderate SPL's and is marginally suitable for serious recording and measurement purposes. The microphone itself is extremely linear, but the built in FET amplifier stage is not configured optimally.  

Fortunately, is it possible to modify the external connection to the FET. This involves some delicate work of cutting a trace on the tiny pcb in back of the cartridge and soldering thin, flexible wires to the standard two hookup points, and making a different connection to the capsule housing. The pcb has a circular ring of copper on its outside, which can be exposed by shaving off a small section of the aluminum roll surround with a sharp blade. Then solder a wire from the copper ring to the hookup point that was originally the positive capsule terminal and make it the new ground wire. The terminal with the cut trace becomes the new active output. You might sacrifice some cartridges while learning the process. It is important not to overheat the cartridge when soldering the wires to it, because it destroys the electret. Also, after wires have been connected, the whole backside should be covered with Silicone II Glue to avoid low frequency roll-off due to a broken air seal which might have been caused by the modification. Heat shrink tubing can be used to attach the capsule to a wooden dowel, or the capsules can be glued into the end of a 1/4 inch ID thin metal pipe. Look at the microph1.gif  page from my notebook for further details. 

The output voltage from the modified WM-60AY capsule can swing 5 Vpp (!) max at 141 dB SPL using a -9 V supply, and easily overload a preamplifier with too much gain. The microphone noise floor measures about 36 dB SPL, 80 Hz to 20 kHz, with a 1/f corner of around 100 Hz. With a sensitivity of 8 mV/Pa (= 94 dB SPL) this translates to 10 uV of noise, or 71 nV/sqrt(Hz) noise voltage density. A-weighted noise is estimated as 30 dB(A). By comparison the equivalent input noise voltage of the OPA2134 opamp is 8 nV/sqrt(Hz), corresponding to the noise voltage from a 3000 ohm resistor at room temperature. Thus the opamp noise will be swamped by the microphone noise as long as the opamp bias and feedback resistor values are not too large.



The above circuit provides buffering of the relatively high output impedance (1-2 kohm) of the microphone capsule. At 140 dB SPL the output voltage is 4 Vrms or 11.2 Vpp and will not clip with a +/-9 V battery supply. Such high SPL levels could be approached when measuring close to the radiating surface of a driver during non-linear distortion tests. The input blocking capacitor can be a small size polar capacitor (Tantalum) because of the applied 9 V bias voltage. The two 200 ohm resistors define the output impedance, provide opamp protection and maintain balance for the XLR connection. Output DC blocking capacitors (e.g. 2.2 uF, 50 V, non-polar capacitors for forming a 7 Hz highpass filter in conjunction with a 10k ohm input impedance) may be needed in series with each of the two 200 ohm resistors, if the connected equipment has no input blocking capacitors or if it puts out phantom power. A single OPA134 or other opamps could be used, a LED could be added to indicate power on/off. If the lead length from the batteries to the opamp is within a few inch, then no supply bypass capacitors are necessary, but if an AC derived supply is used, then 0.1 uF capacitors should be placed from the opamp supply pins to ground. Batteries are preferred, because such supply could cause hum problems due to ground loops.  

A more sensitive capsule with a few dB lower noise of 26 dB(A) is the WM-61A. It outputs a maximum 5 Vpp at about 134 dB SPL.

The wiring diagram for a  stereo microphone with output attenuator, if needed, and using the modified capsules is shown on the microph2.gif page. Attenuation of the microphone output degrades the signal to noise ratio and should only be used, when the preamplifier gain can not be reduced.
A discussion of microphone issues can be found at the MicDIYers forum.


Lyman Miller, March 2008
I first learned about the microphone modification from Lyman Miller, Palo Alto, a friend and former colleague at HP, who came up with some of the most natural sounding recordings I had heard. Now retired, he is still much involved with recording at Stanford University, mostly for their archives. He uses a sphere microphone of his own design, initially with Panasonic capsules, later with Schoeps and now with Sennheiser commercial microphone elements.

Inspired by Lyman and Russ Riley, I have used the Panasonic capsules in an easily portable and unobtrusive stereo microphone setup for recording a variety of sound events in order to obtain reference material for evaluating my loudspeaker designs. A sampling of such recordings is available on a Demo CD.

 

 

Shaped toneburst generator

The shaped toneburst generator is a gating circuit which counts off five cycles of an incoming sine wave and applies a cosine amplitude envelope to them so that the output signal gradually builds up from zero level to a peak and then decreases gradually back to zero. Since the number of cycles is fixed, the signal occupies a constant percentage frequency bandwidth, in this case about 1/3rd octave, centered at the sine wave frequency. The duration of the signal is long at low frequencies and becomes shorter as frequency increases. 

The mathematical expression for a 5-cycle burst with cosine envelope is:

f(t) = [0.5 - 0.5*cos(w*t)]*sin(5*w*t)      

where  w = 2*pi*f   and   0 < t < 1/f   

Preferably, the burst is repeated once every second. The burst frequency should be stepped in 1/6 octave increments (1.1225*f), starting at 20 Hz for full frequency coverage.

The signal is uniquely suited for speaker and room measurements, because when observed on an oscilloscope, it shows both frequency and time domain aspects of the system under test. Ref. 13.  It is also a safe test signal to explore maximum output capability of drivers and power amplifiers. 

There are several ways in which you can generate shaped burst signals:

  1. Purchase my Toneburst CD. It provides 4-cycle bursts in 1/3rd octave frequency steps from 20 Hz to 20 kHz.
  2. Create your own CD-R by using the "Expression Evaluator" function of the GoldWave Digital Audio Editor. A procedure for generating burst files was provided by Dave Reite.
  3. Build a simple gating circuit which approximates the cosine envelope as shown in brst_gen.gif. Some of the specified IC components, though, may no longer be available. The circuit has been updated to the extend necessary by Jean Hernan. The gating circuit must be driven from a signal generator, for example the Neutrik Minirator. Burst and signal generator, together, form a very practical, portable test signal source. This is not a pure burst source, because the approximation generates lower amplitude harmonics of the burst.  
  4. The most versatile solution for creating shaped tone bursts with all kinds of envelopes and numbers of cycles is provided by the Agilent 33120A  Arbitrary Waveform Generator in conjunction with the HP 34811A BenchLink Software. To me this is the ideal solution in conjunction with a storage oscilloscope or MLSSA. It allows safe dynamic tests of speakers and electronics to expose their limits without destroying them.
  5. Download or listen to a 1 kHz, 4-cycle burst with Blackman envelope, 1kblkman4.wav, a test signal which is optimized for time domain observation of the response decay. A 16000 point data file of the waveform is available as blkman_4.prn. Point and amplitude values are separated by a tab. You can open the file in Excel and create a graph. It should be possible to generate from this *.wav files of different frequency.
  6. A signal consisting of four bursts with cosine envelope over 10 cycles of a 100 Hz sine-wave is useful for evaluating low frequency room acoustics, 100mbst4.wav. The loss in audible articulation of the test signal modulation is a measure of the masking due to room modes and other resonances excited by the speaker. Ref.1 & 13. A 16000 point data file of the waveform for a single 10 cycle burst is available as cos_10.prn.

You can download and install a virtual Minirator MR1 signal generator, apparently obsoleted by NTI, to work with your computer's sound card. It provides a limited set of frequencies and signal types but can be useful nevertheless.

Sometimes you may need to know the frequency of a sound. To match a pitch and find its frequency download the Virtual Piano from Mike Marzalek's "CI Theory" website.

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| Build-Your-Own | Main Panel | Dipole Woofer | Crossover/EQ | Supplies |
| System Test | Design Models |
Prototypes | Active Filters | Surround | FAQ |

 
What you hear is not the air pressure variation in itself 
but what has drawn your attention
in the two streams of superimposed air pressure variations at your eardrums

An acoustic event has dimensions of Time, Tone, Loudness and Space
Have they been recorded and rendered sensibly?

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Last revised: 03/19/2014   -  1999-2014 LINKWITZ LAB, All Rights Reserved