Hi everyone.
since I'm not an expert on the subject can you help me understand something?.
lately I have read about the state variable analog active crossover, if what I understand is right it is possible to vary the crossover frequency that separates the high pass from the low pass simply by maneuvering a potentiometer in order to find the most correct and suitable crossing for the speakers.
usually to do this you use a software to make simulations and from that you get the filter and then refine it, I would like to understand if I can do without this method.
Is it possible to use the state variable crossover combined with listening to sounds to find the crossover frequency? but I don't know what range of sounds to listen to in order to do it.
bye thank you
since I'm not an expert on the subject can you help me understand something?.
lately I have read about the state variable analog active crossover, if what I understand is right it is possible to vary the crossover frequency that separates the high pass from the low pass simply by maneuvering a potentiometer in order to find the most correct and suitable crossing for the speakers.
usually to do this you use a software to make simulations and from that you get the filter and then refine it, I would like to understand if I can do without this method.
Is it possible to use the state variable crossover combined with listening to sounds to find the crossover frequency? but I don't know what range of sounds to listen to in order to do it.
bye thank you
well it depends on a few things some are two way, some are three way, and from my days as a soundman even four way stereo.
then it comes down to what drivers are being used, box type and so on...it's not easy to work out and in most cases i would still measure with a mic and look at the response with something like REW.
in my PA days i always ensured that there was an RTA (real time analyzer) in my rack but nowdays software analysis is the new norm.
then it comes down to what drivers are being used, box type and so on...it's not easy to work out and in most cases i would still measure with a mic and look at the response with something like REW.
in my PA days i always ensured that there was an RTA (real time analyzer) in my rack but nowdays software analysis is the new norm.
To understand the theory in active filters, might I suggest reading The Design of Active Filters by Doug Self.
The debate on DSP for active filters is an entire different rabbit hole. The only thing I know for sure is the DCX was horrible.
You find the correct crossover points as slopes by modeling the drivers and measuring the distortion. Evaluate displacement, offset, phase, group delay etc. Many cross the tweeters WAY too low because the see a frequency chart that looks so sweet and try to use the natural rolloff. Big mistake, big distortion. Many cross the mid way too high and can't handle the breakup or are forced to use steeper slopes or additional filters adding other undesirable side effects. Crossover parameters are chosen as the best tradeoff for the measured parameters. We call this "engineering"
The debate on DSP for active filters is an entire different rabbit hole. The only thing I know for sure is the DCX was horrible.
You find the correct crossover points as slopes by modeling the drivers and measuring the distortion. Evaluate displacement, offset, phase, group delay etc. Many cross the tweeters WAY too low because the see a frequency chart that looks so sweet and try to use the natural rolloff. Big mistake, big distortion. Many cross the mid way too high and can't handle the breakup or are forced to use steeper slopes or additional filters adding other undesirable side effects. Crossover parameters are chosen as the best tradeoff for the measured parameters. We call this "engineering"
in the dark ages of my youth we would do it empirically with listening, which is, i think what the OP is asking, just failing to know what to listen for in the process. as much as i advocate measurement i'm reminded of the many soundchecks i did in my PA days, where i've had to quickly access a rental rig for a show, listening to the individual outputs of the x-over low, mid, high and coming to tuning solution in terms of frequency points (assuming they are variable,some where fixed freq) and then establishing a gain structure that provided a good in room response is certainly not easy by ear but when it's the only thing you can do, it's not impossible to arrive at a workable solution.
if you keep levels reasonably low and simply take time to play with the control of your x-over you'll sart to make a correlation to how it affects the sound, which is good , but let common sense prevail, don't run tweeters or horns too low in frequency (that can hurt them knowing something about the drivers your using helps,such as typical frequency range and power handling capabilities/sensitivity) same as not running a sub too high in frequency (it will start to sound boxy) and don't neglect to move around your listening space to get a sense of far field effects. when it comes to midrange that's where, because our sensitivity getting "right" seems the hardest, but ensuring you have a smooth transition to both low and high comes down to just what parameters you can adjust with the crossover your using such as in filter slope and type but that's a deeper rabbit hole for later...so fear not find some music you like and have a go i guaranty you'll learn something and have new questions.
if you keep levels reasonably low and simply take time to play with the control of your x-over you'll sart to make a correlation to how it affects the sound, which is good , but let common sense prevail, don't run tweeters or horns too low in frequency (that can hurt them knowing something about the drivers your using helps,such as typical frequency range and power handling capabilities/sensitivity) same as not running a sub too high in frequency (it will start to sound boxy) and don't neglect to move around your listening space to get a sense of far field effects. when it comes to midrange that's where, because our sensitivity getting "right" seems the hardest, but ensuring you have a smooth transition to both low and high comes down to just what parameters you can adjust with the crossover your using such as in filter slope and type but that's a deeper rabbit hole for later...so fear not find some music you like and have a go i guaranty you'll learn something and have new questions.
"Dark Edges" What I was talking about. Back in the '70s, used to drop in a speaker builder and we would all sit around playing with piles of parts on the floor. There is no excuse for that now. Free software tools that would have cost hundreds of thousands of dollars on a mainframe, PC measurement for an $100 USB mike and free software. Has been since this Millennium started.
The classic state variable cross over is 12dB/Octave only, and you need to adjust two (identical) resistors to move the cross over frequency for both high-pass and low-pass. That's half as many parts as a Sallen–Key cross over and only one value R and C.
https://sound-au.com/articles/state-variable.htm
Many analog cross overs required several different values for the resistors and capacitors and a set for every 6dB of slope for each end of each band, so you get a lot of odd value parts required, and of course, there is no way to move the frequency without changing several parts. But it is possible to use a single value resistor and capacitor in Salen-Key filters if you play with the gain and Q. This is fun to do in SPICE simulations.
https://en.wikipedia.org/wiki/Sallen–Key_topology
But this is the 21st century and you can do a much better job with a DSP. There is no need to cascade many stages for a steep cross over and multiple frequency bands, accumulating noise and distortion in the process. A DSP can use your favorite algorithm if you don't like the results, and it can do phase and amplitude EQ in the same hardware, and the frequencies can be changed easily. Ok, so it's more than most amateurs want to take on, but it is the best solution available today.
https://www.minidsp.com/applications/digital-crossovers/digital-crossover-basics
I should comment that a typical cross-over uses symmetric slopes at each frequency, typically Butterworth, where each band is -3dB at the cross-over frequency. That looks nice on a graph, but loudspeakers are messy. Due to phase shift, analog filters only sum to flat if you use a very gentle slope of 6dB/octave, etc, etc. The best crossovers are finely tuned to the speakers by people with the best of analysis equipment and knowledge.
https://sound-au.com/articles/state-variable.htm
Many analog cross overs required several different values for the resistors and capacitors and a set for every 6dB of slope for each end of each band, so you get a lot of odd value parts required, and of course, there is no way to move the frequency without changing several parts. But it is possible to use a single value resistor and capacitor in Salen-Key filters if you play with the gain and Q. This is fun to do in SPICE simulations.
https://en.wikipedia.org/wiki/Sallen–Key_topology
But this is the 21st century and you can do a much better job with a DSP. There is no need to cascade many stages for a steep cross over and multiple frequency bands, accumulating noise and distortion in the process. A DSP can use your favorite algorithm if you don't like the results, and it can do phase and amplitude EQ in the same hardware, and the frequencies can be changed easily. Ok, so it's more than most amateurs want to take on, but it is the best solution available today.
https://www.minidsp.com/applications/digital-crossovers/digital-crossover-basics
I should comment that a typical cross-over uses symmetric slopes at each frequency, typically Butterworth, where each band is -3dB at the cross-over frequency. That looks nice on a graph, but loudspeakers are messy. Due to phase shift, analog filters only sum to flat if you use a very gentle slope of 6dB/octave, etc, etc. The best crossovers are finely tuned to the speakers by people with the best of analysis equipment and knowledge.
Ashly defied the definition of simple 12 db only crossovers as early as the mid seventies and "something" Davies (sorry old memory) introduced 18 db per octave filters as the next evolution but shortly after there was a slue of "active crossover's" that offered 24 db per octave filters and professing better sound quality so despite Doug Self's text there was no correspondence as to what was better....
That depends what you mean by "sum to flat" - total driver power or on-axis pressure amplitude? If you are at normal listening position you might think the latter is what counts as flat, assuming room reflections are small by comparison. If you are at a random location the total power might be more relevant as the average level in the room depends on power emitted. At very high frequencies its difficult to be precisely on-axis, and at very low frequencies room effects probably dominate, so its complicated.
The 4th order LR sums to flat amplitude on-axis and is probably quite a good choice for bass/mid crossover in an acoustically flat space, despite having 3dB dip in power - you are more likely to be in on-axis zone at those frequencies, and reasonably far from room resonances (hopefully).
Basically there is no single notion of "flat" for a crossover once you are in a room (headphones, sure).