Monitor EQ morphs with output volume

  • It would be nice if the Monitor EQ (ideally parametric ;) ) could be linked smoothly to the Monitor output level in order to compensate for loudness effects. This would allow for sounds to remain more consistent over a wider range of volumes (sleeping wife, wife not home, acoustic gig, rehearsal, club gig, stadium gig...) You program an EQ at low volume, and one at full volume, and the parameters morph smoothly in proportion to the volume. Maybe add the ability to do linear or log functions as required.

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  • This would be near impossible to achieve as the changes found in the Fletcher- Munson curves are neither linear nor logarithmic.
    But for a loudness effect with more bass and treble on low volume this FR could be nice.
    I just can't see any scenario where I had to this by morphing, and what you outlined looks more like a configuration thing to me. Care to elaborate a bit more on this?

  • The FM curves themselves aren't linear or log at SPL=f(frequency), but for a given frequency the phon=f(SPL) relationship is roughly linear. Say that you need to boost bass by 6dB and treble by 3 dB at low volume to maintain "loudness". By the approximately linear phon to SPL relationship, you would need to boost bass by 3dB and treble by 1.5 dB at the halfway point of loudness perception. This would be simple to automate. Each frequency tracks linearly, but at different slopes. To see what I mean, look at the FM curves and note that at a given frequency the steps between phons are about the same distances, but the steps at different frequencies are different.


    So in simpler terms, I'm suggesting a variable loudness feature that could be customized for a given listener and cabinet. Program your monitor EQ at high and low volumes, and the KPA will interpolate at any volume automatically. This should work pretty well with linear interpolation, but I suggested other curves to allow for personal preferences. I often do my programming at low volume, then the sounds get harsher as I turn up the volume. It would be nice to have an EQ that was gradually and proportionally applied as I got louder.

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  • A kind of "automatic compensation of Fletcher Munson Effect" in the Output Section of the Profiler has been demanded several times. I also would highly appreciate this feature !

  • A kind of "automatic compensation of Fletcher Munson Effect" in the Output Section of the Profiler has been demanded several times. I also would highly appreciate this feature !


    Would be nice , but I think the loudness of the sound is determined by the amplifier and/or boxes after the Profiler. :whistling:

  • Sharry, "loudness" as used here is an English idiom for volume compensating EQ, not for the actual volume. I'm not sure what the German word for it is, but it is the same as the button on a stereo receiver that adds bass and treble to the sound to add the same "excitement" as you get from playing music loudly. For me, I really need a "softness" EQ that takes off some bass and treble at loud volumes to make it sound more like the sounds I dial in in my living room. I have much more time to tweak at home than I do at rehearsal.


    The only difference to the normal monitor EQ that I am suggesting is that it be automatically variable so you don't have to program multiple compensation EQ's for different stage volumes. I like simplicity :D .

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  • Reading Ingolf's reply makes me think that I misunderstood Sharry's reply. So I guess you are proposing that there are other "loudness" factors besides volume? I agree 100%. The frequency response of speakers is proportional to volume. They also distort at high volumes. However, these effects manifest somewhat smoothly, if not entirely linearly. This is one of the reasons I proposed having different response curves available.


    I think that dismissing this as not possible because there isn't a perfect mathematical function that can track EQ compensation to volume is a case of perfection being the enemy of improvement.

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  • I think that dismissing this as not possible because there isn't a perfect mathematical function that can track EQ compensation to volume is a case of perfection being the enemy of improvement.


    You're right. I'm not dismissing the idea in general. Loudness switches on older high class hifi amplifiers do a decent approximation already.
    What I was trying to say above is that it (the F-M- curves) are a complex thing.


    OTOH thinking about it more: there are hearing aids on the market already that have different sensitivity for the frequency spectrum depending on the loudness. They are truly Fletcher- Munson compensated.

  • Sharry, "loudness" as used here is an English idiom for volume compensating EQ, not for the actual volume


    Well don,t be sure about the correct word. I mentioned the volume (dBA ore sound pressure or what ever) which influence the way of hearing which is explained by the Fletcher Munson Effect.
    This is set by amplifier and Box and not direct by the output of the Kemper and can not be controlled ore adapted with a morph able parameter. (This is the topic)

  • There's a continuous Loudness adjustment knob on my Yamaha A-500 stereo rack amp. While it can't know (or predict) how loud the signal I am listening to is, it's basically a volume attenuator with a pre-shaped F-M compensation curve.
    It's not based on how high the current SPL is, it just determines how much you want to attenuate it: a good compromise IMO, which certainly works much better than a linear attenuation.
    If Kemper gave us this kind of control with some pre-shaped curves to choose from (in order to approximately take care of the absolute SPL initial value), it would be more than enough for practical uses, provided of course that we'd be out of a linear response anyway.
    This would be a way to implement this function IMO :)

  • My considerations concerning Fletcher Munson compensation are much more simple (therefore maybe pretty wrong) :


    Using a simple analog guitar rig, the increasing of the amp's volume doesn't seem to cause the FM-Effect so intensely as it is with a FRFR-System. This is due to the fact that the guitar-Speaker just only can transmit frequencies up to max 4-5 kHz (beginning around 100-150 Hz / 3dB). So you normally tweak only the bass- and the treble-knob a Little bit to adjust the sound , when cranking the amp . The "frequency-rest" is converted into heat and magnetic Hysteresis.


    With FRFR-Speakers the FM-effect occurs so much more, because the Speakers are not "frequency-limited", so the additional bass- and treble-Response is evident.
    So, in my opinion, a slightly volume-controlled bandpass-filter (similar -but vice versa - to the cab-on/off-filter in the monitor-output) would highly improve the profiler's Response, when cranking the volume.
    (hope you see what I mean)

  • So, in my opinion, a slightly volume-controlled bandpass-filter (similar -but vice versa - to the cab-on/off-filter in the monitor-output) would highly improve the profiler's Response, when cranking the volume.
    (hope you see what I mean)


    I think I know what you mean but I can not imagine a solution which is realized with the Kemper.
    The sound the people hears is normally controlled by the FOH.
    There is no parameter you can control from the KPA to change setting of the FOH exept the Volume of the input signal to FOH. (=KPA output)
    This does not really represent the Volume what is setted at FOH Output (Boxes). And this is the volume which caused the FM-effect.
    As a simple example the output of the Kemper is as high as possible, but the mains of FOH are very closed. It will not be loud.

  • Oh, I'm sure it wouldn't be impossible if you calibrated the monitor settings with a dB meter.


    I was l talking about the FM-effect. The FQ mentioned a wish for indication of the active Monitor output.
    The indication of an absolute value of signal strength at the KPA output (in a defined unit at the best) could be very helpfull.
    But I see now way for compensating the FM-effect with that value. It would need a Phon meter in front of the boxes with a connection to the KPA to make a calculation of the parameter for modifying an EQ.


    The unit dB is a logarithmic relation and as all relation it needs to have a defined base.
    In tontechnique for audiosignals in germany and many countries in Europe, it is established that 0 dBu corresponds to 0,775 V effective.
    This is the voltage which has to be to become 1mW power at 600 Ohm. It is normally the unit which usually is indicated for the mixer output. ( for a audio signal in Europe)


    But this says nothing about the loudness (sound pressure or how ever you want to call it ) you will have at a venue which caused the FM.
    Its easy to believe, that a 30 W active studio monitor will not be so loud as an 800 W active FRFR Box with the same inputsignal from KPA.

  • You guys are making this way more complicated than it needs to be. There is no reason to know the power output or SPL levels or have any reference to the FM curves within the KPA. The user simply needs to EQ the monitor at two different volumes to his/her preference and interpolate the EQ values in between. The interpolation function needs to be chosen such that the morph works correctly. The KPA monitor volume goes from 0 to 10. If this value already tracks SPL linearly than you just need to do linear interpolation on the EQ and you are done. If not, you need to find a function that linearizes the relationship. This can be done just once with either math or measurements by Kemper. Since the KPA already has interpolation functions built in (morphing), the whole process of implementing this shouldn't take more than an hour unless there is some architecture issue in the way.

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  • You guys are making this way more complicated than it needs to be. There is no reason to know the power output or SPL levels or have any reference to the FM curves


    Well, if you want to approach the issue scientifically, reasons are right there: how an SPL variation influences the "average ear" also depends on the absolute SPL value.
    IOW, F-M curves are parametric: the percent perceived level increase\decrease of a given frequency depends on the absolute level it changes from.


    Of course, if an approximation is all you want, then a solution like a continuous loudness compensation (like the one I've described in my previous post) might nevertheless work well.


  • Well, if you want to approach the issue scientifically, reasons are right there: how an SPL variation influences the "average ear" also depends on the absolute SPL value.
    IOW, F-M curves are parametric: the percent perceived level increase\decrease of a given frequency depends on the absolute level it changes from.


    That isn't correct, and maybe it is where some are getting caught up. The beauty of using the relative logarithmic decibel scale is that you can compare relative values without an absolute reference. You only need to state a reference value to convert a dB measurement (a ratio) into an absolute value. Adding +3dB at 1W output and adding 3dB at 100W output are dramatically different amounts of added power, but the ratio of increase to the original signal is exactly the same in both cases. This is exactly why we don't need to know the level of a fader to add 3dB of boost of EQ to it. Now all we need to know to make this idea work is if the required boost level at a given frequency to maintain equal tonality at a louder volume is always constant in terms of dB.


    So here is an example to explore that last question if you are all still following me and I haven't outlived my welcome... :D


    Check out the FM curve on Wikipedia or where ever. Let's start at the 20dB SPL isophon. Assume we have the amp EQ'ed perfectly at this really soft level. Now we want to play at 20dB louder and still sound the same. We need to add about 10db of SPL (proportional to output power) at 100hZ, 20dB at 1K, and 20dB at 5K. But now note that if you were to jump from the 60 dB isophon to the 80dB line, you would be changing by the same relative amounts (dB's) for that jump too. It didn't matter where you started from. You need to add 10dB to the bass, and 20dB to the mids and treble to make the sound be perceived as 20dB louder across all frequencies. The isophons don't always jump equally at every frequency over the entire range, but it is pretty darned linear as far as our chaotic world goes.


    And of course the FM curves don't tell the whole story. Everyone has experienced compression drivers getting nasty at higher volumes. This too can be compensated for by gradually and proportionally reducing treble output as volume increases.


    And regarding FM being only an average for human perception, that doesn't matter either, since the user will be setting the two calibration points themselves based on thier own peeception. All that matters is that the jump from one isophon to the next is roughly constant at a given frequency. If you needed to add +10dB of bass from the 20dB isophon to 40, +15 from 40dB to 60dB, and +5 from 60dB to 80dB, this would not work because you WOULD need to know the SPL your actual cabinet was putting out to make the proper jump. As long as the jumps at a given frequency are approximately the same, then we don't care how loud the cabinet is when we change volume.


    Does this make sense? I have a BS in Engineering, so some stuff that might seem obvious to me might not be to all. I'm trying to explain my point the best I can, but I can't always tell if I'm doing a good job.

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  • In principle, I think an indication of output level and an EQ in the output (already exists) are useful.
    I think that an automatic compensation of the FM-effect by morphing EQ parameters does not work in practice in my opinion.
    There is no exact formular for conversion from SPL to dPA - the connection has been established empirically with sinus signals. (I think there exists approximation formulars )
    Therefore there is no linear (or other mathematical) relationship between dB Kemper-output value to dBA-value at the box.
    Then one would have to determine the average distance of the audience. This would give the figure for loudness feeling of the audience to be compensated. (even not linear.)
    Furthermore, one would have to consider the 'average age of the audience. There are empirical curves taken into account the age.
    Then probably the size and nature of the hall will play a significant role.
    I think the good known practice to make a frequency analysis to get some information about the venue and decide ultimative with the ears can not beaten at the moment.
    I do not see a simple solution with a reasonable ratio of expenses to effect.


    Peace ;) - Harry