wait, there are some contradicting infos in your post. as I understand it transients are those “things” that make up the attack-phase of a sound which in turn make a sond snappy (or not, then there are no transients).
so, what you wrote :
using a gate with a small attack does kill transients.
downward expansion … no clue what you mean. my music is always loud as shit, I don’t have to do this. (just joking, I know what you mean, although I do it differently)
shorter envelopes do actually feed transients, making them stronger and able to fight back. not good.
no reverb : I am with you there. I like my sounds dry.
brickwall compress … depends. Somehow I am too stupid to understand the idea of compressors, all those terms of attack, release and all that. infact, if I use the preset of a compressor (any compressor) I end up with something I don’t want, a sound that “snaps” like hell and then gets soft. So I always use them as limiters (more like a clipper) with adjustable knee and then lower the makeup thus effectively killing my transient this way (by generally lowering them down).
Downward expansion simply refers to using the effect known as an Expander “improperly” as a gate by turning the ratio down instead of up.
Shorter envelopes kill transients because the instrument plays for less of a time… I’m not sure how this would add to your transients.
Compression works by pushing the volume down when a spike in volume (above a certain threshold) is detected… people often take this to the extreme by cranking the ratio up in a poor attempt to make their track sound louder… this is called brickwall compression because it sounds like the volume peaks are being held back by a brick wall that won’t budge. Compressors can be used in creative ways also, but I was specifically reffering to the brickwall compression method.
I’m not quite sure which information was conflicting, but I hope this clears up a bit
a good explanation I once read was with the piano, when you look at a single piano-sample you have these few ms at the beginning where it is kinda lika a total loud, high-pitched noise and only after that the piano sounds evolves, which means the attack of a piano has alot of transients.
A really extreme example (I think because the mic was too close to the hihat) but also a good example why you sometimes need to kill transients. Play that with a kick or snare with enough snap and you wonder why your level suddenly goes trough the roof (or an autolimiter reduces the master-gain to a level that you’d normally need decompression-procedures for when you get back up toward +0db). And I would say brickwall limiting is the best solution for this particular problem, and not only brickwalling the entire mix but this particular sample on it’s own. (altough I would actually manually edit it on sample level for this extreme example (which is btw from a commercial library)).
the point of all this is that you normally you have f.e. the first milliseconds of each first beat where alot of sounds (or their transients) fight for their snap, basedrum, a hihat, some bassounds, some arpeggiated melody and so on and so forth, the question is if all those sounds really need to be that snappy, you normaly wouldn’t recognize the particular snappiness of a melodic “pling” (from the arpeggiated melody) when a huge basedrum kicks at the same moment, you can kill the transient of that “pling”, doesnt matter anyway.
But I may be totally wrong about this, maybe someone got a third opinion.
for natural sounds ofcourse (apart from a botched recording as this one above), but for artificial sounds from synths and similar I think it’s good to edit them from time to time.
OMFG I got my terminology all wrong… not the first time that’s happened. Please replace the word “Transients” with whatever word is used to describe sustained notes (perhaps just “sustain”?)… sorry about that people, I feel like a complete ass now. Where I managed to pick up that twisted definition of the word Transients is beyond me, but at least it’s an antonym to what I actually meant to say
How can you expect to get a mix right if you’re compressing the master output? Compressing individual tracks is fine, but don’t touch that master channel! It was designed to be at 0.0 dB, while the loudest instrument peaks at about -8.5 dB. Unfortunately Renoise only halves the volume (reduces it by 6.02 dB), so you’ll have to reduce the input gain (pre-volume) of each channel by -2.499 dB if you want to have enough headroom. Some styles will even need more headroom, so start pulling those volumes down and mixing the way it was supposed to be done. You’ll start hearing a lot more and your mixes will improve tenfold. I think it’s really time I wrote a tutorial on setting your levels right in Renoise, because it’s absolutely terrible the kind of mixes people are pulling out of it. You can always slam the crap out of it when mastering, but a mix should peak no higher than -3.0 dB.
I can help a lot more just as soon as you upload a proper mixdown. You’ve probably already clipped things beyond repair in the mix, but getting that output down would be a good start.
I could get you the xrns, but nobody compressed the master here.
I don’t what a ‘proper mixdown’ sounds like, so I cant help you there.
Afterall, that’s the whole point of this topic. Duh
sorry atlantis, but what you say makes absolutely no sense. a brickwall limiter on the master set to -.1db and adjusting the in-gain accordingly is the absolute standard and minimum of mastering. I like the new mixing btw, it’s definately not as harsh at the original one. still abit low on bass for my taste, but much, much better anyway.
Or did I misunderstand something? Reading it now, it seems you’re talking about compressing the individual tracks. But read my words: compression is NOT a way to hide peaks and turn the volume up without clipping. Compression reduces dynamic range by turning the volume down above a user-defined threshold. This allows you to turn the peak level back up to where it was before, bringing the average level up with it, but this is only secondary to actual dynamic range compression. Too many people think of compression in a backward sense, and it makes them drive the compressor by pushing the input into a clipping state, believing the compressor will prevent any peaks from going past 0.0 dB. This is only partly true, but it is the wrong mentality. You need to think downward. Turn everything down and make the parts fit well together instead of pushing everything up where it can’t go higher.
A proper mixdown shouldn’t clip for starters, so that you can accurately hear what’s going on in the individual parts.
As for sending me the XRNS, I’d probably be overwhelmed by all your volume wrongdoings. That’s why I said the mixer routing is probably beyond repair, but you can still turn the master post-gain down and work on balancing the post-gains of each mixer track, and EQing the instruments to sound better. If you want to send me the file though, it might help me to visualise what’s going on, but I don’t have any of the VSTi’s.
This isn’t the way to use a limiter. Don’t drive the input gain - lower the threshold instead and set the output ceiling to e.g. -0.2 dB (after you’ve found the threshold you want by keeping the output ceiling and threshold the same).
Is there a difference between pulling the treshold down and normalizing the wave afterwards and leaving the treshold at -.1 and pushing the input gain up?
im sure there was a calmer way of putting all that…tutorials are always gunna be good though if this kind of thing gets ur goat so to speak.
as far as my 2c goes, iv always had bad results when applying compression to the master track. my prefered technique is to put compression on each track (and even then only when i feel it needs it as i believe compression produces a certain sound which i dont always desire) and then just EQ the master and maybe a bit of barely audible reverb/etc just to help things to sit together a little. i admit that last bit is a bit of a cheat though
Atlantis, thanks for the tips! Indeed that was what I was using
compression for, it gets things loud and crispy, especially with ‘digital’
sounds. I’ll look more into compression. I did compress the various
channels instead of the master. At LEAST I knew I shouldn’t be doing
any compression on the master
Captain Woo, I never read any Nagel. Nagel is Dutch for nail.
Thanks for the kind words tho !
I listened to both, I like the Foo? tweak more than the original.
It’s really cool, but gives off a king’s quest vibe. Have you considered working with an orchestra / choir? It’s ambitious but it would really give your musical exploration much more profound depth and notoriety. In the decades past, when i was a student in film/video school, I have had stuff scored for free by classical music students at universities and the sound quality is unmatched by “computer emulation”. You should look into it if you are serious about this direction, otherwise it comes off cool, but works as a sketch, or a shell, to something that could be much bigger. There’s just so much one guy on a laptop can do before you have to take it to the next level of musicality. Really, the first step in “mixing orchestral” is recording an orchestra…
In any case, really awesome as usual, but this could be so much more, evidently with great difficulty.
BotB has got a good thing here, I just let it shine.
For everyone’s info this was my rough workflow:
Take down all the volumes and compressors.
Balance the orchestra tone and compression, as well filtering the verb ‘wet’ to reduce mud.
Re-EQ the bass, added mda-sub.
Re-EQ the guitar, added mda-stereo.
Re-EQ the drums and tweaked compression to reduce dynamics a little.
Added most of the drums to a send and added tube warmth.
Added light mda-sub to the drums.
Spent a while balancing all elements in relation to each other - basically aimed to have everything perceptible no matter how busy the sound got. The ‘pre-mix’ never got much over -6dB.
Built a mastering chain, firstly some reductive EQ off the subs and a light rolled-shelf from 14.2k up.
Maximiser set to about +10dB boost, and the absolute peak volume set to -0.3db.
Referenced boosting with FreeG, aimed four around -8-7dB RMS.
Used SSL X-ISM to check there were not inter-sample peaks.
Yes, so I’ve been trying to say. Though it is more about mentality and the benefit of mixing downward instead of upward. If the threshold is at -0.1 dB, you’re going to have to push the gain about 5.5 dB into clipping, which can cause clipping in some effects. It’s just stupid and really bad practise.
(just joking, I know what you mean, although I do it differently)
