Originally Posted by YoSoyPincho
Wow sir, first of all i would like to thank you immensely
You are very welcome, sir.
You are absolutely right about the reverb low ends on synth, i did notice the melody was getting lost in between something, but didn't recognize the reverb as the possible suspect here! It does makes sense as i experimented a lot with wet soaked reverb sounds, trying to make it sound "spacey/wide". I will be more aware of the reverb frequencies indeed. Thanks a lot!
OK, if you are looking for some ideas of how to accomplish a spacey sound, here's my 2 nuyen!
Don't turn to reverb for spacey sound.
It seems like that's where you would go, but unless you have a really nice [Only registered and activated users can see links. Click here to register]
, you're just not going to get something that works for a moderately fast melody line, with lots of active supporting tracks around it. You'll instead end up with too much for the human brain to easily sort through at the speeds per second that the quantity of sound information is delivering. The net effect will be an audible fog with highlights plucked out here and there because our brains are impressively fast and will still hang in there with most of the notes and we'll still get the general gist of it.
What I would suggest is to drop the reverb entirely.
Grab a flanger effect and turn the rate all the way down to minimal, and the saturation/affect level to maximum and then dial it back until it makes a pleasing metalic buttery sound...almost like talking inside of a culvert.
Then grab a delay effect from your tools. Grab one that has the ability to be short, well pronounced, and that you can easily dial in to roughly around 8th note repetition.
Play with it around a little, but basically you'll be looking for JUST a slight bit of repeat/delay and yet a QUICKLY fading return so it doesn't pile up over time (just to where you can hear 2 to 3 times after the initial note, but still well below the actual note volume levels so that if there's overlap, the "real" note takes priority).
Then, and this is where the fun comes in...
Grab a stereo widener and flip the polarization 90 degrees so that rather than "down the middle", you'll be sending the sound "down the left and right".
Make adjustments, but you'll probably want the sound to be spaced out somewhat wide so that it's not super narrow (but sometimes you might find the super narrow works on a melody).
Alternative method - the Giorgio Moroder style.
In this method, you still toss on a flanger and dial everything way back, but you skip the rest.
Instead, you duplicate the track and then offset the duplicate by a 32nd or 64th note or even less...you'll have to zoom way in probably...just a HAIR off..actually, less distance than a hair.
Then you move one track slightly to the left (about 11 o'clock position) and the other to the right (about 1 o'clock position).
This is a painful to edit, but if you already have everything laid down, then this can often be more clean and create the effect more clear.
However, it's all about dialing in that hair of a difference between the two tracks.
If you dial it too far out, then you end up with a literal doubling of every note, and if you dial it too tight then you just end up with a huge leap up in your volume level.
That's another challenge with this; you'll have to readjust the entire level for the melody because you have two tracks spitting out the melody instead of just one.
I go with both of these depending on what's going on.
If it's a really full song, I'll often go with the single-track effect method, but if it's a smaller track song with lots of room, then I might go with the Moroder way.
And about the Spectrograph, it is incredibly accurate as i actually put a brick-wall low-pass filter around 18 kHz on every single track because i normally don't like the really "sandy" bright sounds, and the mid frequencies were also quite boosted up too much originally, which i happened to attenuate with EQ's, but i think i had to lower the level even more, to reach up the -16 dB's as you mention.
Yep; that ("brick-wall low-pass filter") would definitely do that.
The other way around this is to address the sound profile at the instrument level.
This may not be possible with all synths, and in those cases, you'll have to work on it via EQs or a multiband compressor.
Basically, most instruments have some sort of tonal control and you can usually sculpt it into shape around the edges with an EQ or a multiband.
If I'm not working on a square ADSR envelope[Only registered and activated users can see links. Click here to register]
, then I'll most often use a multiband because the nice thing about an MTB over an EQ is that it's much easier to not create amplitude accidents with an MTB.
And EQ can easily punch up the middle in large amount just because you've tried to isolate it and not noticed that it's increasing the dBs rather than just isolating the pocket you like.
An MTB can more clearly shave off the top end and focus the middle ground in without that issue....but if you want to dial a really complicated profile of the tone in, then an MTB isn't going to be what you reach for.
But I really do prefer addressing tone in the instrument sound itself.
Most sounds can be adjusted without using effects to get there, and if they can't be, then a different sample or synth might be worth considering before reaching for EQs or MTBs to shape the tonal profile.
I would, however, suggest stopping the low-pass cut filters.
I know that's a super common thing to do, but just don't bother with it.
If you want to apply a low-pass filter, you could apply that on the master or one of the mix buses, but I wouldn't do that on the instrument track layer very often.
Ultimately, however, I would just not use low-pass filters to build a sound profile, and instead stick with EQs and MTBs.
Frequency pass filters are great for cutting off areas that don't need to be included to make sure you don't have problems when the song is played in different venues and at different (really loud, for example) levels (like big PAs), but even though it's common now, they aren't really intended to be used to instrument tonal profile creation.
I like to think of the pass filters (both high and low) like the walls of a sandbox.
There's just literally no sound after that point.
But that can be very dangerous if used too narrowly, like on every track, because that's like building a miniature sandbox frame inside the sandbox because you want to raise some sand up in one spot.
The problem is that you eventually box yourself in with very little flexible elbow room this way, so there's no real ability for expression to flex "naturally".
It's more like a machined piston of sound - very specific, but very powerful.
Which CAN be cool, but you really have to be more careful, and use it as a tool and not just a default setting because not all sounds play nice with this treatment and you can get conflicts and problems generated.
Anyway, I'm rambling...sorry about that.