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Thread: Tricky mic position create random response mesure

  1. #1
    Senior Member B&KMan's Avatar
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    Tricky mic position create random response mesure

    Hi everybody,

    After small controverse in regards of position mic for evaluation calibration SPL, I perform for members, a small experience of danger of mic in near field for coherent response and amplitude value.

    I put my microphone in near field : exactly same space mic-driver of all 3 mesures but just keep mesure in arroud top edge of driver center of driver and low edge of driver...

    Ok if you look pict the change position is really small maybe 2 inch for all 3 position in vertical and no in deep and no in lateral...

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  2. #2
    Senior Member B&KMan's Avatar
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    In according of picts the result of 3 mesures is catastrophic random response in frequences and amplitude...

    conclusion at this closeup all positions is too critical for good evaluation and reproductible datas. each 1/16 inch back, angle, down or side affect strongly mesure... do not put your SLM or mic too close... inside of 2 feets the close-up is too critical and push a complexe procedure for extraction of fairs stable datas.

    I hope this help to better calibration..






    Jean.
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    Senior Member B&KMan's Avatar
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    So At frq 16K and at exactly same power output, the difference value is more than 12DB depend just 1 inch vertical position!! at 1/8 inch higher position; differente curve and power response....

    So if you put your big slm into near field for evaluate SPL add the reflextion of instrument + the type of mic, plus if you load in weight "C" or "Lin"...

    Indianas jones jungles sound heres baby....

    If you consider any this type of response for calibrate high or med or Low Frq, what your chance to drop notch right on the good value ??


    No surprise many sceptical is claim : hey my ear is the best (youafafafaf).



    Jean.

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    RIP 2011 Zilch's Avatar
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    Oh, Jean, Jean...

    I BELIEVE Ian was measuring INTEGRATED near field SPL, not FR, and he advised using a tripod such that the microphone could be precisely positioned at the desired location for most reliable determinations.

    In any case, as he subsequently reiterated for clarity, the measurement and adjustment is RELATIVE, NOT absolute.

    [I'm thinking OTHER things, too, but I'll just stifle them for now....]

  5. #5
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    I think a much more meaningful test would be at the standard 1 meter distance, which allows for normal wave development and propagation, especially with a slot type of radiator. Up so close, you're probably dealing with speaker "quantum mechanics" effects. And standard 0, 15, 30 and 45 degree offsets would seem to relate much closer to what you're actually hearing and listening for.

    John

  6. #6
    Senior Member B&KMan's Avatar
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    Quote Originally Posted by Zilch
    Oh, Jean, Jean...

    I BELIEVE Ian was measuring INTEGRATED near field SPL, not FR, and he advised using a tripod such that the microphone could be precisely positioned at the desired location for most reliable determinations.

    In any case, as he subsequently reiterated for clarity, the measurement and adjustment is RELATIVE, NOT absolute.

    [I'm thinking OTHER things, too, but I'll just stifle them for now....]

    hey hey Zilch... talk your all your point of view with no reserve ( )


    for spl with sinewave:

    mmmmm what is interest to keep relative SLP of sinewave of 16K is any 1/16 inch change on couple Db the SPL ??? any mesure in interesting if reproductible... this is a basis of any datas...

    the interest to expose the many differents fr response in DB and expose the yoyo SPL if you choose generate one frq or other...

    choose any fr and check the difference in reagrds at same fr but just little difference keeping...

    Look the Global spl line in white and the total of thi difference is gange more than 4 db in pink noise !! (this is result of mesure exposed )...

    so the measuring INTEGRATED near field SPL in this method is very very tricky...

    So maybe I have completely in field but I do not understand what is interest record arbitral value with very tricky reproduction value for the same driver and for other...

    so the result is
    not conform to power spectral energy of driver at all fr and at specific fr
    and not according to environnemental area listening
    Low reproductible value

    So open my eyes and explain what good aspect is possible in this set-up

    This is my 2+2 cents point....
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    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
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  8. #8
    Senior Member B&KMan's Avatar
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    Quote Originally Posted by Ian Mackenzie
    Okay,

    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.

    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.
    The Doctor
    Hi Ian,


    Ah I feeling this great member put light on this phenomenon...

    Now I understand and I realize the reliability and theorical approach...





    but I'm not understand again....

    1--- If It Is the interest is put 3 dB (or 4 for horn) down spl why JBL put Lpad ??? just put resistance and voilā !!!

    2--- Well any driver is a specific variation on regards of reference driver (this is true for the etwork too )...
    But I Assume the tolerance factory is tight but not identical... So if your driver is drop 0.75 dB than reference... I assume you have necessity to upper correct with L-pad for boost current for compensate low efficience...

    Assuming 2 same test of same type driver in aternativlly connected in same network:

    If you put L-pad in max for driver A and for Driver B but .75 Db difference power on driver response and if you work your method you fix L-pad at same place for each and your report the exact 3 Db down response electric and acoustic but the start level is difference for A to B ??? your result is same differencial power and arrival... No ?? normally you arrival at same L-pad position but absolute SPL result is different right ???
    If you add the network imprecision tolerance: the differencial result is more no ??

    On other hand if this result is good in theorically and perfect in anechoic chamber ?? but it is good in any room...



    Okay

    3--- your method push a big question Why JBL put L-pad ?? If just for fix exactly -3 dB (4) electric power for driver... I loose probably another aspect...

    4--- My litterature expose the interest of L-pad is for opportunity to place SPL and ideally spectrum fr resp in flatness possible response on room and reagards of sweet spot listening...

    Because any room is different and many materials absord , diffract, diffuse, and reflexe, the necessity to tune speaker on regards of this is important...

    the final result is flat spectrum as possible arround your ear , in regards of all environnemental variables...

    5--- I'm not shure the perfect down 3dB acoustic response is corrolar to ideal electric balance on network... and I'm not shure if the network is better " happy" in this set-up and produce best transfert...


    6--- I'm feeling the interest if this method is fix the unprecision of scale arround L-pad only...


    aouch !! my skin is iritate by too scratching head...

    Please doctor dont drop me ...


    Jean.


  9. #9
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    Okay,

    The L Pads are there at the end of the day toi do what ever you like with them.

    I don't understand your other assumptions. Assuming you follow JBL standard test rig and measure the voltage drives there should be no issues.

    If you are hearing impaired, have a very dead room or a very live room you might want to adjust for a variation or like deafening skreachy sound turn the horn up full by all means. Some owners resort to using a 1/3 octave equaliser in an attempt to make the response as smooth as possible at the listening position. In small moderation this can be useful.

    However if the latter is true the problem is the room and not the speaker.

    In my 25 years experience with 4343's the single biggest problem with listening satisfaction is setting the L pads of all the drivers to attain a acoustically correct setting. You can spend months or even years and not get it right.

    After performing the above procedure I have also measured the on axis response using MLS and obtained the smoothest overall response.

    Where the response of individual drivers is uneven this is extremely difficult. Even with smoothing and you have to account for room interference and micophone location.

    The method above is simple and accessable for most owners.

    Go here for further information.

    http://www.audioheritage.org/vbullet...&postcount=141

  10. #10
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    Ian, Jean, between this thread and the 4343 mod thread would you both consider getting serious about sound ...You both make a lot of sense in your methods. Jean, with the multiple mic positions and Ian with the Tandy. I can relate to adjusting for MORE than 25 years, as well as using an EQ. Just about the time I thought I had that "Magic" setting, I would most always get out of the chair and diddle the knobs once again. It seemed that every song or new music source would beg for a "Minor" adjustment. Finally, I boxed the EQ and let the knobs alone. NOW, I have to have a Tandy. Thanks guys....

  11. #11
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    Rec,

    That is the most sensible post about this topic I've read for a while and I appreciate your thoughts.

    Asbolutely, even in the vinyl days I recall only one or two albums sounded correct, Billy Joel's 52 Street was one of them. This is another dilema of owning accurate monitors. Everything that's wrong tends to show up and only complicates the issue of tonal balance.

    I do find less insanity however in knowing that the pads at least are set at a true reference point to start with.....bass and treble knobs on your amps anyone. Were'nt they a thing of the past?

    The Doctor

    Edit.Sorry I missed you first point.

    I suppose I got serious about JBL's sounding right at the time when I heard my 4343 at my brothers place. At that point a changed out the 2344A bi radials, sold them off to Cyclotonguy and spent 12 month rebuilding a set of 4345's.

    However my quest for serious sound started really when I switched to Passlabs Class A amplifiers about 4-5 years after indulging Cyclotonguy, Karen and Neslon Pass . But getting the L Pads set correctly is a good place to start.

  12. #12
    Senior Seņor boputnam's Avatar
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    I guess I don't get it.

    Measurements at those proximities will have little bearing, IMO, on what is heard at the listener position - that is, unless things are REALLY outa phase / time alignment. There is so much sonic/acoustic reaction in the near-field, that to measure things well inside that field would seem to be ignoring real affects outside that domain. But, I'm confused...



    Assuming the engineers/makers had things fairly close, FFT will reveal what is occuring at the listener position and guide your fine tuning.
    bo

    "Indeed, not!!"

  13. #13
    Senior Member B&KMan's Avatar
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    Quote Originally Posted by boputnam


    Hi,
    first I dont try the Ian Method so I'm not put judgement in now...

    actually the question is SPL in regards of spectral density, spectral power, power vs energy and corellation with SPL. a FFT and DFT approach and quality of each in regards of time capture ... problem of each approach, the precision of one record to average, room response, cepstrum, convolution and deconvolution, hilbert transform, Coherence, zwicker's loudness, zoom window, etc, etc, my acoustician pratice is push me work with many standart internationnal ISO and ANSI and create a path of relatively same repetitive methodologie...

    But is getting too serious so I equalizing my response a little bit by exemple to come of JBL Engeener...

    If you buy a beautyful high level grade synthesis sytem ( 100K USD) a Jbl gentlemen representation is go at your home with biggest box....

    Is extract 4 mic (1/2) pressures field (so the mic is vertical position) for best random response in quasi difuse field and connect this 4 mics at console to integrate sum signal response. The mic is different positions in height and distance of wall center, for better average... the pseudo-pulse train is produce the lattop deliver the result and calculate the compensation curve. In same time you look curve the operator decide if apply correctif , if yes the curve correction is sending at the processor sound include in system synthesis... after this another pulse train is emitted by system for look the result... this loop test is run 5 minutes, hours or days, in regards of precision and problem appear in curve response...

    Many professionnal RTA (DFT approach) offert a extension module for connect 8 or 16 mic... but a millionnaire set-up !! so I keep one and call function average... so each record is sum calculation... I'M decide if I try 3, 5, 6, 9, or 100 record... of course at certain point the average difference is too small and irrelevant...

    this is ISO and ansi method for calculate echo, room response and mony other parameter difficult to catch with precision...

    conclusion : For non ideal environnement: the average is the key... because the position mic is all the time affect by 3 types of mode occuring in the room, axial, tangencial, and oblique.

    this is the professionnal approach and recommendation, Of course now the sound intensity, and mapping sound device is exist and fix relatively this problem, but one sound intensity, analyser and calibrator is same price of synthesis ( ... )

    I hope this cue is not really help many guy for fix problem but is put light on why and ... Help consideration...

    Best regards.

    Jean.

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    Senior Member B&KMan's Avatar
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    Quote Originally Posted by rek50
    Jean, would you both consider getting serious about sound
    .

    thanks for reply,


    Well, well well, the shock of idea create light and sound !!!

    I'm consider this forum with serious and many member help seriously with my network and other tips so I just return the elevator... but it is possible to joke a few time no ???

    manys guy read this and other tecnical tips so if start on my recommendations and finish in dead end : is it seriously not respect...

    jean...

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    Quote Originally Posted by boputnam
    I guess I don't get it.

    Measurements at those proximities will have little bearing, IMO, on what is heard at the listener position - that is, unless things are REALLY outa phase / time alignment. There is so much sonic/acoustic reaction in the near-field, that to measure things well inside that field would seem to be ignoring real affects outside that domain. But, I'm confused...



    Assuming the engineers/makers had things fairly close, FFT will reveal what is occuring at the listener position and guide your fine tuning.
    I guess I might be able to imagine why you are confused.

    Its a case of discipline of the mind to analyse specific facts out for some's desire for a dust cloud of confusion.

    We are not talking about a whole Lab case study here on designing a system from stratch nor are we searching for micro changes in air density to locate zenomorph's on the edge stratasphere.

    Think of this a striping down the paint layers of colour on a car till you get down to the bare metal. The top layer is the visible image but as you go deeper and deeper ...Perhaps the reconing and clean down to the bare metal is a better analogy. We know how important that is!

    Anyway, as I posted originally one needs to understand the relationship b/n voltage drive and net spl. When a speaker is designed one key aspect of the design is matching driver sensitivities...we know that from Project May. But we are not talking about a two way where you can itteratively compare low and high and eventually get it right. This is s bloody 4 way!

    One needs to know that the net spl of each driver is matched between all four drivers BEFORE one can start making claims to the effects of near field reflections, room reverberation and all this other crap. Phase anomallies and what happens latter on is not the point.

    Attempting to look at the entire response curve of a whole system will send you cross eyed trying to work out which aspect of the curve is right., what's up, down etc and then you say okay I better boost that with equ and then you wind up with a complete mess.

    This is why I am say to those particularly who are building their own systems and networks to consider the approach I outlined.

    If you engage yourselves in the practicallity of such an exercise the benefits of what appears to be an over simplification become very obvious. As you turn the L Pad and the needle moves to zero 0 dba mark you will understand why you are doing this.

    Studies have been done that support evidence that small amplitude broad band deviations of 2 - 3 octaves over the entire spectrum are accutely more audible as shifts in tonal balance than irregular or random peaks and dips. This is why RIAA curve accuracy is so important, a +- 0.75 db variation or less is quite audible. So obviously the better the match of the 4 driver spls the more true the system will perform, doing that at 3 metres in your room is lunacy.

    And so if we apply ourselves to the task of finding a way to determine the correct level match of driver sensitivity in a multiway system it is more likely to sound correct.


    The Doctor.

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