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Thread: Tricky mic position create random response mesure

  1. #31
    Senior Member B&KMan's Avatar
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    Ok this is my curve in according of your set-up mic at 1.5 m

    impulse tone, rectangle weignt window... etc, the pict contain all indication mesure...

    the notch of 8k is the result of bad cutting network...

    normally is cut at 9.5K but at 7k the hf is full here and UHF is full here too soo the cummulative is little over 4.7 DB... but impossible to redure this notch with-out loos the rest of sprectrum of UHF... sic !!! this is another reason I build a new network...



    Jean.
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  2. #32
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    Hi,

    Well that is good, it will be interesting to see what your new network does!

    The Doctor

  3. #33
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    Quote Originally Posted by Ian Mackenzie View Post
    Okay,

    I am now Back, the Daleks nearly got me..lucky for my sonic screwdriver I escaped!

    Zilch is correct.

    My reference is in effect to the design of the system with respect to relative Spl pressure response. The drivers respond to a specific spl based on sensitivity and the voltage drive applied via the filter sections. Therefore for each driver to operate at a matched spl of acoustic output from flat amplitude the voltage drive to each driver must be adjusted via the L PADs.

    In order to understand this it is important to realise we are refering to net spl of each driver at a specifc descrete point. The overall flatness and integration is a function of the filter slopes, phase response and location of the drivers on the baffle. JBL have already done this design work so our JOB is quite simply to adjust the voltage drive.

    After some trial measurements about 18 months ago I realised using FFT, MLS and Pulsed Analysers lacked the instringic precison for performing a measurement of net SPL based on differential of the voltage drive.

    So I pulled out my trusty Tandy meter and mounted it on my tripod and fed a sine wave via a PC contolled function generator to the amplifier. By placing the meter a specific distance from the baffle in front of each driver I was able to measure with great precision the net SPL via fine adjustment of the L pads.

    I used frequencies of 600, 5000 and 150000 for the midrange, horn and slot respectively.

    I started with the slot and positioned the meter directly on axis about 2 inches from the baffle and wound down the pad on the horn fully and adjusted the level with the Slot Pad fully open to +4 db on the meter. I then carefully adjusted the L pad till I got 0 db, after repeating this several times I marked the postion on the foil Cal.

    I then place a masking tape over the Slot mouth and repeated the same test with the horn with the meter again on axis with a reference of +4db (and wound down the midrange L pad fully) and then adjusted the horn L pad to 0 db.

    The same technique was applied to the mid come using several measurements to assure precison of the +3 attenuated back 0 db level.


    The location of the meter should be directly on axis but exact location is not overly critical. The reason is simple in the that we are only concerned with the net relative spl after adjustment from fully open L pad back to a desired point for 0 dba. As JBL advise this on their foil cals in Dba and the voltage drives from the crossover filters are already pre determined it is fairly straight forward to adjust the exact levels of each driver.

    I think it is reasonable to assume that JBL with all their vast resources would have found the 0 dba reference point.

    In fact I have found using this technique the only reliable and repeatable method of adjusting the relative balance of the drivers on a 4 way system.

    Mid and far field measurements using even gated analysers do not offer the same precision for this kind of measurement.

    When I then played the system it has portrays the smoothest and most remarkable coherent sound and the imaging is amazing.

    I find the need for further adjustment completely un necessary

    Needless to say my tongue in cheek remark (since edited) in the crossover modification thread is probably apt. It's simply a case of interpreting the available information and applying it with some simple techniques to solve a problem.

    But if we come back here in few years time we will probably find people scratching their heads with the same problem.

    The Doctor
    I know this is an old thread,

    Wouldn't you get *about* the same result if you played some pink noise, medium low volume, then measure the db of the woofer (or driver that is not adjustable via lpad), then measure horn, tweeter, and adjust those to the same db level as the woofer?

  4. #34
    Senior Member B&KMan's Avatar
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    Quote Originally Posted by jfine View Post
    I know this is an old thread,

    Wouldn't you get *about* the same result if you played some pink noise, medium low volume, then measure the db of the woofer (or driver that is not adjustable via lpad), then measure horn, tweeter, and adjust those to the same db level as the woofer?

    Hello


    Well, if you know the exact frequency cut off and only play it on the 15 '', note the level in dB or dBA. Then you increase the power of 8 '' until you have + 3 dB or dBA that goes. Then you redo the same procedure with compression driver and then the tweeter.


    the problem is obtain exact cut off frecency dividing network...

    If you take a pink noise and measure globally in dB or dBA, it will be impossible to find the right level of adjustment because the acoustic energy is based on the frequency range covered.


    ps.s. in fft it is necessary to use white noise rather than pink noise

    ----------------

    for this thread, the fundamental question is what is the ideal position or angle to arrive at having an answer that "ear sound" balanced. In other words where should we place the microphone to have a calibration that corresponds to a good sound balance listening.

    Two approaches: on axis at 1 to 1.5m in the axis at the common height between the 8 '' and the flute or squarely at the sweet spot of listening


    That said, several microphones offer an approximation of balance significantly more performance than a single microphone.




    But that does not take into account the work on phon in a diffuse field.


    The tonal balance changes according to the fact that it is diffuse (no free field) and according to the volume.


    Several approaches give a downward slope as the frequency increases, but by exactly how much?


    And what is the answer of the room itself? In short, not as simple as it seems ...


    my 2 cents

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