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Understanding speaker basics.

Discussion in 'd.i.y.' started by The Captain, Mar 21, 2009.

  1. The Captain

    The Captain ~~~~~~~~~~

    Im always glossing over the graphs, Hz numbers and other driver characteristics- so could anyone give an idiots (i mean real layman's) guide so I can try and understand the basics of speaker drivers, &/or the basics of a x-over?

    I'd like to try & work out a suitable upgrade for my vifa d19 tweeter, so some page 1 understanding i think its about time i tried to learn.

    cheers capt.
  2. 337alant

    337alant Negatively Biased

  3. mudlark

    mudlark nearly half a clue

    My own view is if you are going to have a play about then try going active. The experience is very worthwhile. The ESP audio site gives lots of information about how to do it.

    the easiest way to try this is to get hold of a behringer or similar active filter and use your stereo power amplifier in a two way set up. The CX2310 is less than £100. Experiment with one of your speakers and compare it with your old speaker. If you don't get anywhere then sell the behringer and go back to passive. Sell filter on Ebay or here.
  4. 337alant

    337alant Negatively Biased

  5. The Captain

    The Captain ~~~~~~~~~~

    Thanks chaps, going thru the links. Thing is its all page 1 stuff, ie in the wiki it goes

    'the x-over is a circuit that divides input signal' and other really basic stuff like 'a full-range driver is.. a tweeter is' etc etc: then suddenly it goes to.. "steep slopes such as 24dB per octave" which is complete gobledegook to me. The previous link too was basic explaining, then suddenly into "low-pass filters" and "either a 6dB, or a 12dB table" without explanation as to what the dickens they are, why 2 and not 1 (or 1201)..

    edit: 337 your last link maybe the ticket..
  6. mudlark

    mudlark nearly half a clue

    In dividing up the signal between the tweeter and the bass driver you have to use a filter. (Passive) The filter is a combination of resistors, capacitors and inductors. these filters don't work by stopping any signal at a certain point they start to work at a frequency and as the frequency changes they can be more or less effective. the logarithmic ratio of the input signal to a filter and the output gives the reduction in signal. No reduction is 0dB and a reduction of half is -3 dB. (I think)

    A filter that reduces the signal sharply reduces the signal quickly with a change in frequency. A gentle filter is 6dB per octave and a sharp filter is 24dB per octave.

    A gentle filter generally has less electrical components than a sharp filter. Some people say the more components the worse the performance. Others say that the sharper the filter the better. Others try to use drivers that naturally have reduced output matched between the high and low frequency drivers. Other hate any filters at all. I use a filter between the preamp and the power amp.

    Filters also cause phase differences which can make the sound muddy in character. 24dB active filters can be designed to have a small phase shifting performance.

    its all very staright forward, i don't know why you don't understand speakers.

    Oh in forgot to mention problems that occur with the impedance of the drivers, resonant frequencies and the other problems with enclosure design and construction oh and room positioning, amplifier performance as well.

    The above gives a slight incite into why speaker design is a bit of an issue.
  7. The Captain

    The Captain ~~~~~~~~~~

    many thanks mudlark- effort appreciated, and i do get occasional bits.. but lost me after 'logarithmic ratio' and why the word 'octave' suddenly appears from nowhere, and what et etc: its hugely complicated to even grasp your reply. Am I right in assuming the word 'filter' there is what I know as the 'crossover', as a whole? or is a section of the x-over referred to as a filter? ive not heard this term before.. as for 'phase shifting' I couldn't even hazard a guess as to what it is, although Ive read reference in Mr Ackroyd's spiel on his speakers (which always seemed like a science lecture to most prob 95% buyers who'd not the foggiest what these words/terms mean!)
  8. mudlark

    mudlark nearly half a clue

    In sound the difference between loud and quiet is rather great so when measuring sound pressure levels we measure the quietest sound we can hear and then use that as a basis to give an indication of the loudness of a sound. We use the ratio of the squares of the logarithm of the quiet sound to the sound to be measured. You then multiply the result by ten. The ratio is a Bel and the deci is the ten. (sort of...all you knowledgeable types)

    Try looking at the wiki of decibel.

    because the sound types use decibels for sound pressure level it is also easy to compare volumes of amplifiers as a ratio of the input signal in volts to the output signal.

    If you are filtering some sound you are reducing the sound and this is usually expressed as a dB.

    Also sound pitches are heard a bit weird by the ears holes such that the difference between sounds of low pitch are more easily detected than sounds with higher pitch (frequency). A sound that is double the frequency sounds to have twice the pitch, if you see what I mean. If a sound has a frequency of 500 hz then reducing it by 250 will reduce the apparent pitch by a half. Increasing the frequency by 250 doesn't sound like a doubling of the pitch, 1000 is doubling. This increase is a geometric not arithmetic and again can be represented using logarithms. The doubling of a frequency is roughly what sounds "nice" on a a western piano with eight notes between each note of the same name. The difference is called an octave (8)

    So with filters we need to use a way of describing them that works with our ears and how we perceive the sounds. a Filter with equal filtering above and below a frequency point needs to be biased towards the higher frequencies so instead of a fixed frequency difference we use an octave. A 6dB per octave filter which is low pass would have the following result if based on a 50 dB sound at 500 Hz. At 250 Hz the filter would pass the low frequency, but at 1000 Hz the level would be reduced to 44 dB. At 2000 Hz 36 dB. A high pass filter centred on 500 hz would have 44dB at 250 Hz and ....50 dB at 1000 Hz. In the case of a 6dB per octave band pass filter designed to pass 250 Hz to 500 Hz would produce levels of 44 dB at 125 hz and 44 dB at 1000 Hz.

    If you can understand that drivel then I am amazed.
  9. James

    James Lord of the Erg\o/s

    The electrical circuit (which may be as simple as a single capacitor or a complex network of inductors, capacitors and resistors) that connects the amplifier to a loudspeaker driver (or multiples of the same driver) is correctly called a filter. In a passive loudspeaker system, the electrical properties of the filter works in conjunction with the impedance of the driver to which it is connected to effect the change in amplitude response of the driver. In the case of a tweeter, the filter (which is called a high-pass filter) attenuates the signal getting through to the tweeter as frequency drops. The higher the order the filter, the more steeply the signal gets cut. A first order electrical filter cuts at 6dB per octave, a 4th order at 24dB per octave.

    A crossover is the outcome when you apply complementary filters to a pair of drivers, e.g. low-pass to midwoofer and high-pass to tweeter, so that the net acoustic result is a seamless and flat 'frequency response' over the crossover range. This, unfortunately, is easier said than done.

    Why? Three simple reasons.

    First, it is a acoustic result that matters. In other words, the 'order' of the electrical filter does not translate directly to the 'order' of the crossover result. This is because loudspeaker drivers do not have flat frequency response outside their intended operating range. For most tweeters, you will see that the acoustic response drops from around 2kHz and below. Unless the high-pass filter is applied at least two octaves (i.e. at 8kHz) above that range, the natural roll-off of the tweeter will add to the 'order' of the crossover. So a simple capacitor, which is a first order electrical filter, combined with a rolling-off tweeter at 2kHz, is anything but a first order crossover. More likely a second order or third. The lesson from this is actual crossover order = electrical filter order + natural driver roll-off order.

    Second, a passive filter works in conjunction with the impedance of the driver to which they are connected. That impedance is rarely flat, which is what all crossover formulas assume. This means that the electrical filter that promised you a second order electrical roll-off will deliver something a little less predictable. If the driver, particularly tweeters and midrange drivers, has a huge impedance spike within the crossover range, then all bets are off. Unless you can implement another circuit to flatten that impedance first. The lesson from this is textbook filters rarely work as intended, especially in the passive filter domain. A properly designed filter must take into consideration the natural acoustic response of the driver and its impedance characteristics. This is almost impossible without measured data and extensive modelling. Computer software such as LspCAD helps significantly.

    Third, acoustic phase is problematic. That's the bad news. The good news is that phase is directly linked to frequency response and that can be 'predicted' with good modelling tools. Why is phase a problem? In a worst-case scenario, the resulting output from the drivers cancel each other out over some or all of the overlap because they are at 180-degrees to each other. This results in a crappy sound and very little music. The idea for phase alignment is so that the outputs of the driver pair in question sum flat over the crossover range. For odd-order crossovers, the phase is 90-degrees out of phase, but the acoustic response of a properly designed loudspeaker should still sum flat. There are well established crossover 'types' such as Linkwitz-Riley, Butterworth, Bessel, Chebychev etc., which trades amplitude response vs power response vs phase alignment depending on what's more important to you. Oh, the other thing is that actual phase alignment also depends on the relative position of the driver pairs. All of the crossover theory assume that the drivers are acoustically aligned, which is NEVER the case for a loudspeaker with a vertical baffle and flush-mounted drivers. However, these can be compensated for in the crossover, but you will need actual measurements and modelling tools. The lesson from this is loudspeaker design is more miss than hit, unless you have actual* measurement data.

    Hope that helps.


    * published graphs from manufacturers don't count because these are normally measured on an IEC baffle or infinite baffle, which your loudspeakers are unlikely to mimic. The size and shape of baffles affect the acoustic response of drivers.
  10. mudlark

    mudlark nearly half a clue

    Elegantly put james.

    I wonder how much The Captain has understood. I think I get the rough idea about how speakers work.

    Now if you could just explain in the same straightforward way how a transistor amplifier works and why there are so many damned bits everywhere and why some bright spark hasn't cracked the job yet.
  11. The Captain

    The Captain ~~~~~~~~~~

    I'd say probably around 3%. of both the last 2 replies, unfortunately. thing is terms just spring from nowhere, like "2nd order", "attenuates". having no idea what they mean Im at a loss to understand even a fraction. oh dear I guess I'll leave this then! (i think james must be pulling my leg anyway: I mean #1.. an idiot's guide/ layman's terms-?)

    thanks all the same, I'm sure it'll be useful to some folks.
  12. James

    James Lord of the Erg\o/s

    No, sorry, no can do. I haven't the foggiest how transistors work.

    No, I wasn't. Some subjects are naturally more complex than others, and it's hard to explain it all in lay terms. Whilst loudspeaker tweakery/design is not exactly rocket science, it's not as simple as replacing a tap washer either. There are many variables at play, and if you don't fully appreciate them, then you are likely to be disappointed with the results (at best) or blow something up*.


    * for example, a second order tweeter filter is a series capacitor followed by a shunt inductor. If the shunt inductor is put before the capacitor, you will effectively short your amp.
  13. The Captain

    The Captain ~~~~~~~~~~

    yes I can appreciate getting something inherantly the wrong way round would likely blow a spkr coil-

    really I was wanting to grasp the basics of the two curves on a graph corresponding to the HF and LF in a 2-way design.. and therfore to understand what and what would not work as a replacemnt tweeter. In my case of the Monitor Audio r352 the x-over only comprises 1x r, 1x cap, and 2 inductors so that's at least a simple circuit.
  14. James

    James Lord of the Erg\o/s

    If you are simply looking to replace the tweeter, then the safest bet is to find one that has a similar* response curve to the original, and more importantly, a similar impedance curve. If the replacement tweeter has a lower resonant frequency (fs, typically at the peak of the impedance curve), that is better than one with higher. The lower fs is less likely to get in the way of the crossover. As I said in my earlier post, the filter works in conjunction with the driver impedance. If the latter is significantly different, then the response curve will be different. Try and get a tweeter with the same sensitivity too.

    I don't recommend playing around with the filter component until you know what changing the values does.


    * because different manufacturers display their data differently, you might need to replot the curves on graph paper to compare different tweeters on the same visual basis.
  15. stackowax

    stackowax pfm Member

    This may be totally off point but why are the filters always multiples of 6db (6db, 12db, 18db, 24db)?
  16. andyr

    andyr Registered User

    I wanted to rebuild the XOs of my Maggie IIIa about 15 years ago and faced the same problem. So I found a night course at a local technical college (1 night a week for 10 weeks) which took participants through all the basics, in terms of XOs and driver parameters. The (electronics) lecturer was on the staff at the tech college but he was a long time speaker builder/hi fi nut! :)


  17. 337alant

    337alant Negatively Biased

    Thanks for your input it really helps me try to understand this stuff;)
    Could you explain why my tweeter is inverted (as per design) after the high pass filter?:confused:.

    Captain, I thought the Troelsgravesen stuff was interesting as well.

    Edit James guess I should give you more info on my speakers
    They are a Volt / Scanspeak design by wilmslow audio, they where originally home studio monitors, stand mount but I have changed the cabinet to a floor stander, maintaining the same volume of course.
    Volt BM220.8 and Scanspeak 2905/950000
    Bass Xover is 2nd order and Tweeter is 3rd order filters

  18. andyr

    andyr Registered User

    And even 48db, 96db etc - if you're using digital filters like the DEQX has. :)

    "Xdb" is the slope that a 1st/2nd/3rd/4th etc. order filter has, if you use the theoretically-calculated component values for the nominal impedance of the driver you've chosen.

    But you can in fact produce lower-dB slopes by using component values which diverge from the "standard" ones ... eg. the nominal 12db slopes for second-order filters used in Maggies are often about 10db.


  19. James

    James Lord of the Erg\o/s

    That's most likely because it has a second order crossover, which results in phase that is 180 degrees from the woofer. Reversing the polarity of the tweeter brings it back into alignment. My Ergo-IIIR has 2nd order acoustic crossovers, and as a result the midrange is inverted to align with the woofer and tweeter.

    The other reason for inverted tweeters is wayward children's fingers, but I suspect that is not what you meant.

  20. 337alant

    337alant Negatively Biased

    Thanks for that, Its actually a 3rd order on the tweeter, and sorry, yes I meant out of phase.

    In this article there is a discussion about exchanging one scan speak tweeter for another without adjusting the cross over, the interesting thing for me is that in my speakers the Xover is similar to the first example but I note that he achieves a much flatter response by modifying the Xover to last example.
    My question is could I use that same Xover design for my tweeters???


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