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Another trick...High Pass / Low Pass

Discussion in 'Axe-Fx II Discussion' started by jzucker, Jun 16, 2016.

  1. Hubi

    Hubi
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    Thanks for explanation - but:
    The Speaker canĀ“t produce higher frequences than maybe 4-5khZ, but they are there ? who produces this frequences ?
     
  2. ben ifin

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    See here the freq chart here:-

    http://celestion.com/product/1/vintage_30/

    Guitar speakers "dont cut" any frequencies.

    A V30 is quoted at 70hz - 5000k .... however if you look at the freq chart, it simply "tapers" off quite quickly after ~5000k .... and tapers off quite quickly below 70hz .... the freq's [ and other 1st, 2nd order etc... harmonics ] are there, just nowhere near as pronounced or " obvious " as the primary freq's for which the speaker is designed to reproduce.

    If you had a guitar speaker and configured / set it up so that it did not recieve and therefore did not reproduce any freq's at all below 70hz and over 5000k .... it would sound very very weird / bad.

    Ben
     
  3. Rex

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    This. The sound you get from the Axe is closer to what the audience hears from your amp than to what you hear. when you're standing in front of the cab, with the cab on the floor.

    Put another way, the sound you get is what you would hear if you had ears in the back of your knees. :)
     
  4. FractalAudio

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    Those types of graphs are typically made with the microphone 1 meter from the speaker. The classic method is "1W / 1m" which is to apply 1W and measure 1 meter away. When you get the microphone close to the speaker the response is much different and you usually get more highs and lows. This is "close mic'd" and is the technique normally used in studio recordings. During mixdown the producer/engineer will then often highpass and lowpass the signal to remove these excess highs and lows and to make the guitar "sit in the mix".

    IRs are almost always made using the same close mic'd technique and, hence, will sound like a raw recording. Far-field IRs are possible but very difficult to obtain requiring a large facility and special techniques.

    Our primary goal is to model an amplifier and speaker as accurately as possible and the latest modeling is astonishingly accurate. We do not purport to be producers or mix engineers and leave the choice of low cut and high cut frequencies up to the user. Furthermore many users rely on the soundman to apply the filtering at the board, just as they would when mic'ing a "real" amp. More importantly the choice of frequencies is highly dependent upon the IR used.
     
  5. ben ifin

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    Thanks Fractal.

    I was only wanting to point out a real misnoma - a lot of players think that if a real speaker [ say a V30 ] is quoted by Celestion as being 70hz <-> 5000hz that it will produce no freq's / sounds / tones / harmonics etc.... of any kind either below 70hz or above 5000hz .... when in fact that is very much not the case .....

    Thanks again,
    Ben
     
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  6. HarrySound

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    Well this is all very interesting indeed.
    Recently I started cutting low freq my monitors arn't able to produce (according to the spec 49hz)
    I'm going to experiment with the high pass later and see if that sorts my ear fatigue issues.
     
  7. simeon

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    the axe fx already has a global eq (2 of them, in fact). just use it to cut the frequencies you don't like. job done.
     
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  8. VegasGuitar

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    Yep! RTFM, as they say ........ ;)
     
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  9. greiswig

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    Which is one of several reasons I've requested a 5-band parametric EQ there instead of a graphic. If you could switch the outer bands to be 6db or 12db filters, that would end up fixing a lot of my live mix issues better.
     
  10. VegasGuitar

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    That has always made we wonder about the comprehension skills of fellow players ...... the chart NEVER goes to ZERO, it simply drops towards zero. As Cliff pointed out, get up close and you will get more of those frequencies ....

    Outstanding thread/topic BTW!!!
     
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  11. born2lift

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    You high cut it 6-8k FROM 20000? Or set the high cut AT 6-8k?
     
  12. dpeterson

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    High cut in cab block, stuff is packed so going from memory. 6-8k. At low volumes it's a little darker but brightens up when you wind it up some.
     
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  13. #33 mr_fender, Jun 17, 2016
    Last edited: Jun 17, 2016
    mr_fender

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    The cutoff frequency is only half the equation. Filters always have a slope. In reality it's very difficult to simply cut off all frequencies dead above or below a certain point. The cutoff frequency for a filter is usually the frequency that is at the -3db point in the roll off. How much frequencies beyond that get cut depends on the slope of the filter. That is usually given in decibels per octave (dB/8va). A high cut filter with a slope of 6 dB/8va means that frequencies above the cutoff frequency will be 6dB lower for every octave or doubling of the frequency. So such a filter set to 3kHz means that at 3kHz the cut is -3dB, one octave up (6kHz) will be 6db lower or -9dB, two octaves up (12kHz) will be 6db lower than that or -15dB, and so on. A 12 dB/8va high cut filter set to 3kHz would be much steeper with 6kHz at -15dB and 12kHz at -27dB, and so on, so the highs drop off much faster.

    The different slopes look something like this. Notice the point where they all intersect, the cutoff frequency, is at -3dB and is not where the roll off starts or ends.
    [​IMG]

    In the graph it shows 1st order, 2nd order, etc. That refers to the stacking of filters to achieve a steeper slope, so each additional filter order doubles the slope. Also notice, the steeper the slope, the sharper the knee gets and the flatter the cut up to the cut off frequency. An ideal cutoff filter would have a hard 90 degree angle at the cut off frequency, but that's a tall order. (pun intended :p)

    In this example:
    1st order = 6 db/8va
    2nd order = 12 dB/8va
    3rd order = 24 dB/8va
    4th order = 48 dB/8va
     
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  14. Perdikament

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    Exactly... Those frequencys are there,this is precisely why mic placement makes such a huge difference...
     
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  15. bobmichigan

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    Live and Learn, hopefully.
     
  16. GM Arts

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    I use the GEQ page in the amp block to boost mids / cut lows & highs. Saves using an extra block and allows me to use the master GEQs for minor adjustments to the playing environment.
     
  17. dr bonkers

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    What amp in the room guys fail to consider is that there is always a mic on your cabinet... it is called the human ears.

    Human hearing is very subjective because we all have different frequency response curves with each ear, which are usually very different than most mics.

    If you are curious about the frequency response curve of your particular ears, have a test with an audiologist and ask them to graph the testing results of each ear and the composite curve of both ears.

    If you are still driven to get your amp in the room sound, then seek out mics with those curves, position the mic or mics in your same unvarying listening position in that perfect room, use a neutral power amp with mic+di method, create your ir files, and you will be closer to your dream tone.

    Of course you will need to seek out perfectly flat monitoring in an anechoic chamber for the listenin experience to be spot on and it will probably only work for you in that environment YMMV.

    The other issue if you do this though is other people may not like your listening curve in their playing or monitoring environment.
     
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  18. Ed DeGenaro

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    Im with Simeon...i cut with global eq...but it's the same Axe...or recording chain.... Close Mic can get ratty quick.

    Side Note, the top Note of the guitar is 1200 Hz, bottom 80 some...Assuming E to e 24th fret.
    The stuff above is what makes the difference in Timbre... I.e. overtones.
    Rolling off hard at 4k will attenuate everything but the Octave and fifth on the high notes. With gain IdL certainly Sound more stable.
     
  19. Wolfenstein98k

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    This makes perfect sense... But would there ever be the ability for the user to set defaults other than the factory defaults? I realise there are workarounds like saving a block, but I'm just curious.
     
  20. mtmartin71

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    The challenge in all of this is that there are a million and 1 ways to get to the same place with FRFR EQing/. GEQ, PEQ, AMP hi/lo, cab hi/lo, global GEQ, the IRs, TMB, etc. My goal is to only cut or shape it once but that's not where I land. What I do is use the OH suggested IR positions with the SP2 mix IRs, cut in the cab at 100Hz and 10KHz, and then let it go from there. I could go around 120Hz and 6500Hz for a fully shaped sound with no cut at FOH but 95% of the time I'm playing through our band's board and the bassist is running sound. In the end, you just have to let some of this sh!t go and roll the dice and let someone else tell you if you need to change it.
     

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