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Topics - Khron1k

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Hey all, noticed something a bit odd today when I was applying a high pass filter sweep over a snare drum. I have my snare drum sitting at -10 db, but as I begin to cut low frequencies I end up actually losing headroom, and it ends up sitting at about -3.61 db(well by it, I mean the drum rack that the snare and filter are sitting in). I am quite confused how I appear to be losing headroom, even though I am removing frequency content.

Attached two relevant photos. For reference, my snare is the second element in the drum rack, and the eq is on the drum rack as opposed to on the snare.

1st screenshot is of the filter disabled and the rack metering at -10.10 db(side question, anybody have any idea why this is metering lower than the snare, since nothing else should be in the signal chain).

2nd screenshot is of the filter enable and the rack metering at -3.61 db.



2
Sound Design / Wind up/wind down effect.
« on: March 24, 2016, 04:20:44 pm »
I was listening to the Seven Lions track, Fractals, and was quite baffled by a couple of the sounds I was hearing. I have no idea how to describe them, so going with wind up for the first and wind down for the second seems to be the most accurate description.

Edit: In both clips, it appears like some extra audio got added to the beginning of the clips(even though it's not showing up in ableton), so the effect I am curious about is the one going on in the last half to two-thirds of both clips.

3
A common problem I find myself running into during mixing is correcting for clipping in the master track. I imagine my workflow is pretty similar to most people just starting out:
  • Get all instruments sitting at a level that seems to fit.
  • Attempt to record
  • Notice a spot in the track that is clipping a few percentages of a db on master track
  • Figure out what layers are causing the clipping, and either add a little sidechain and/or adjust the levels of tracks
  • Repeat from step 2 indefinitely until all spots have been adjusted for.

Coming from a Software Engineering background, steps 2-5 strike smell to me like a process which should be automated by software to some degree. At least in regards to identifying the positions in the track which are clipping as well as frequencies which are causing clipping, while leaving the actual adjustment of the tracks up to the producer.

Basically, my ideal workflow would be something like below:
  • Get all instruments sitting at a level that seems to fit.
  • Pass master layer through some tool.
  • Tool spits provides annotation of some sort where clipping has occurred as well as frequencies responsible.
  • I go fix it up.
  • Repeat step 2 to verify that all have been fixed.

Does anybody know a tool which will do something similar, or at least will help partially automate the above process?

Cheers.

4
Going through tutorials, it seems like there are two ways of going about using a sidechain compressor:

1) Individually add a sidechain compressor to each component that needs one. Mr. Bill seemed to go this route in his videos.
2) Run everything that needs to have a sidechain through a single channel/bus/group, and then just add the sidechain compressor there. Virtual Riot seems to go this route in his videos.

For folks that have been producing a while, which do you prefer and why?

5
Sound Design / Confusion around EQ'ing terminology.
« on: March 17, 2016, 03:52:03 pm »
Just wanted to clear something up that's been confusing me. When people mention frequencies that that start high-passing or low-passing at, is that the frequency at which they place the frequency knob in the eq or is it the frequency which should end up getting filtered out entirely?

For the high pass filter in the attached image below, it looks like right about 4k is left unchanged, the frequency knob is set about 2k and then just below 1k is where everything will be filtered out, so which of these frequencies are folks referring to when they say I set a high pass filter at <blah> frequency? Just want to make sure I'm speaking the same language as everyone else.

6
I know that the goal of a compressor is to reduce the dynamic range of the input signal that is passed through it. However, what I can't seem to grasp from a technical level, is why does that (typically) cause an increase in the db of the output signal?

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