HighPass filter

Discussion in 'Effects and the DSP' started by blitter2, Jul 3, 2011.

  1. blitter2

    blitter2 New Member

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    Hi.

    When a HighPass filter with fcenter = 125.6 Hz and BW = 8.0 is used, the range of the effect exceeds peak as it showes on EQP1 monitor.

    But please does this mean that the sound clips and distorts (because over peak effect), or that you just using a filter and its working range gets over peak as well as under peak?
     
    Last edited: Jul 3, 2011
  2. Russ

    Russ Well-Known Member

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    It depends on the signal level going in. The graph display shows what happens with a signal at a level of 0 dB at each frequency. So for example:

    at 125 Hz, the signal level shows approx. +15 dB.
    at 250 Hz, the signal level shows approx. +3 dB
    at 62 Hz, the signal level shows approx. -10 dB.
    at 1000 Hz, the signal level shows approx. 0 dB.

    The actual level shown is only relevant for a 0 dB signal, but because 0 dB is the reference line, it is easy to see how much gain or attenuation is applied at each frequency.

    e.g.
    at 1000 Hz, there is 0 dB gain/attenuation, so these frequencies are not effected by the filter.

    at 125 Hz, there is @+15 dB gain, so any frequency components at 125 Hz will be boosted by +15 dB. So, if there original level (of the 125 Hz component of the signal) is 0 dB, then the level out (for that frequency) will be @+15 dB (as shown on the graph), which would cause clipping. But if the level going in is -12 dB, then the level out will be around @+3 dB, etc.

    Also, as I said in one of your previous posts, there is some headroom in the DSP, that allows you to exceed 0 dB temporarily without clipping.
     
  3. Russ

    Russ Well-Known Member

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    BTW: I thought I should add that in general you will not know what the level is going to be going in for every possible audio source or their individual frequency components, etc (other than it should be somewhere below 0 dB). For this reason, it is probably a good idea to use the In/Out gain knob to offset any large gains (e.g. if you are boosting some frequency by +15 dB, then you may want to reduce the signal level by some amount using the In/Out gain knob to prevent clipping, etc).
     
  4. blitter2

    blitter2 New Member

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    Russ what I do is, wiring a Peak4 level meter in paraller with K1lt (ending of DSP) and making sure everything is bellow 0db.

    Won't this way is supposed to work?
     
    Last edited: Jul 3, 2011
  5. Russ

    Russ Well-Known Member

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    That should be OK, but I do not know if the settings that you mentioned are the actual settings you are using, or if they are just given as an example for the purposes of your question. There is only 12 dB of headroom, so a 15 dB boost could cause clipping (depending on the signal level going in) within the EQ plugin itself (so even if you reduce the level before k1lt, it is not gonna fix it, as the clipping already happened).

    Assuming that the signal level in the DSP (or within any plugin) never exceeds +12 dB, and the level right before k1lt never exceeds 0 dB, then you should not have any problems.
     
  6. blitter2

    blitter2 New Member

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    These are my actual settings.

    Very good Russ, fully understood.

    Thank you.
     
    Last edited: Jul 3, 2011
  7. blitter2

    blitter2 New Member

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    Russ, is it 12db of headroom for each plugin, or for all plugins used as a total value when all gains are added together?

    For example if I use two pass filters with 11db gain on each, the total gain is 22db.

    How does it work? Is it that any plugin's gain shouldn't exceed 12db, or 12db of headroom as a total for all plugins used?
     
    Last edited: Jul 4, 2011
  8. Russ

    Russ Well-Known Member

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    It is headroom for (basically) the whole DSP (but for each signal).

    Signals entering the DSP are divided by 4 (assuming input sliders are kept at 0 dB or 100% (where applicable)).
    This is what gives the headroom.
    Signals leaving the DSP are multiplied by 4.

    You would not want to do this to the same signal (or at least not the same frequency component of that signal (again depending on what the original level was)) without some plugin in between reducing the gain.
     
    Last edited: Jul 4, 2011
  9. blitter2

    blitter2 New Member

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    Thank you Russ.
     
  10. blitter2

    blitter2 New Member

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    With a HighPass filter with fcenter = 63.4 Hz and BW= 1.22 I don't notice any over peak effect at EQP1's monitor.

    Please does this mean its indeed under or matches peak ?

    Please when a filter matches peak , does it have negative effect?

    I am asking because it has been said that a max BW for filters to be under or maybe to match peak could be BW = 0.71 but I now notice on my EQP1 monitor that maybe it could be the same with BW up to BW = 1.22 .
     
    Last edited: Jul 5, 2011
  11. blitter2

    blitter2 New Member

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    As well please how can I boost centain frequency 400 Hz, 800 Hz or 1.6 kHz which is the belongs to the bass guitar one?

    And I would like to attenuate the 50 Hz one.

    Is it the method , Notch and Bell filters?
     
    Last edited: Jul 5, 2011
  12. Russ

    Russ Well-Known Member

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    The graph used in these plugins is pretty small, thus has low resolution (not a ton of data points). Any values that fall between it's data points will not be seen (in full accuracy) on the graph. (e.g. maybe it is rounded to nearest data point (e.g. a boost of 0.4 might show as 0 dB, etc)). [EDIT: maybe not the best example for this graph, but hopefully you understand]

    If the graph shows a line at 0 dB, then those frequencies are probably not affected (much) by the filter.

    I am not sure what you are asking there, but you would probably want to use bell filters to boost/cut specific frequencies like that.
     
    Last edited: Jul 5, 2011
  13. blitter2

    blitter2 New Member

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    Russ about the graph's accuracy, can 1 db be the largest divergence?

    Or more like 2 db?
     
    Last edited: Jul 5, 2011
  14. Russ

    Russ Well-Known Member

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    Sorry, that was probably not the best example as there is more resolution on the amplitude axis than there is on the frequency axis. e.g. The amplitude axis might be accurate to @0.5 dB (as there are @12 pixels per 6 dB), but the frequency axis is logarithmic so certain parts of the graph (lower frequencies) will have more resolution than other parts of the graph. If a boost/cut happens on some frequency data point that doesn't fit exactly on the graph, you might not see it (the graph is gonna try to fit the curve as best it can, but it's resolution is limited).

    I do not know what the largest divergence is, but I think it is meant more as a visual aide to use along with you ears to help you find a setting that you like, etc.
     
  15. blitter2

    blitter2 New Member

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    Thank you Russ.
     

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