# Reducing Transients Transparently



## Daniel Petras (Dec 8, 2016)

Hi all,

On a recent mix I did, I noticed after the fact when comparing my mix to another that my percussion transients were quite dramatic and significantly reduced the amount of headroom I had to work with. What sorts of things do people do in order to maintain that fat percussive sound (especially when in the context of percussion instruments) while transparently reducing the transients in order to save headroom for the mastering stage?


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## Tfis (Dec 8, 2016)

Maybe compression.


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## muk (Dec 8, 2016)

Compression is a possibility. And try Flux Bitter Sweet (it's free).


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## Puzzlefactory (Dec 8, 2016)

Try Alloy 2, it has a multiband transient shaper.


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## Daniel Petras (Dec 8, 2016)

Puzzlefactory said:


> Try Alloy 2, it has a multiband transient shaper.



I was just reading about Alloy 2. Gonna look into that. Cheers!


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## karelpsota (Dec 8, 2016)

If you're trying to achieve loudness with your percussion, try distorting it with saturation. You loose dynamic range but get more harmonics that we perceive as "loud". You can even take it, a step further by transient shaping the distorted signal to add a bit more punch. Neutron and TransX Wide are good plugins to do it.


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## Puzzlefactory (Dec 8, 2016)

Also don't underestimate how much a high pass filter can do wonders to your headroom, as most of the problems lie with low frequencies.

Failing that, a multiband compressor just affecting the low frequencies can also help a lot.


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## Tysmall (Dec 8, 2016)

Try the volume knob before anything else. Your drums do not have to be hitting 0db to be felt.


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## KEnK (Dec 9, 2016)

Look into using 2 compressors ( or 1 limiter, 1 compressor) in tandem. A very effective approach.
The basic idea is to use the limiter to just lower the peaks, then then the comp to raise the body of the tone.
Just a little of each in this case.
Also parallel compression is a good technique for what you're after.

these techniques can work w/ all the other ideas mentioned here.

k


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## Martini Hill (Dec 9, 2016)

Saturation is your friend here.


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## Daniel Petras (Jan 29, 2017)

KEnK said:


> Look into using 2 compressors ( or 1 limiter, 1 compressor) in tandem. A very effective approach.
> The basic idea is to use the limiter to just lower the peaks, then then the comp to raise the body of the tone.
> Just a little of each in this case.
> Also parallel compression is a good technique for what you're after.
> ...


Isn't the first thing you mentioned more or less parallel compression where the dry signal gives you the transients and the over compressed signal gives you more of the body? Wouldn't your method just be equal to turning the dry knob down slightly?


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## charlieclouser (Jan 29, 2017)

I think the most transparent choice will be something like the Waves "L" series limiters - the L1, L2, L3, etc. I think of these things as "peak shaving" plugins, which just shave off, or push down, the sharp transient peaks back down into the body of the waveform with a minimum of audible change to the sound. No color, no mojo, no character - just reduced peaks on the transients. There are lots of other companies doing basically the same type of plugin, and most will probably be cheaper - but I've got a big Waves bundle and out of all of them my favorite has been L3-LL MultiMaximizer. This is a five-band multi band L3 style limiter, with detailed control over each band, and the "LL" version is a low-latency version of the plugin which means I can leave it on each of my stem sub masters to tame the peaks and provide a consistent, predictable output level. This way, when all of my limited stems are summed into a composite mix, there are no surprising jumps in level as various stems combine. In normal use I don't hear any change in the character of the sound, other than the fact that the pointy peaks are pushed down into the murk by a very precise amount.

For many years I had been using TC Electronic's MasterX5 (which has a 10msec look-ahead delay), and then switched to Ozone v5 and v7 (which have variable look-ahead delays dependent on which algorithm is in use), but in a detailed shootout I was surprised to find that L3-LL Multi won the day. So that's strapped across all of my stem sub masters now. A bonus is very low latency and CPU usage - my template has fourteen instances enabled at all times - CPU hit is minimal and I can easily record and play through the plugin without an annoying delay in the signal. I don't know the exact look-ahead delay but it feels like it's less than MasterX5's 10msec.

Perhaps someone else can suggest other, cheaper alternatives to the Waves L3 series?


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## benatural (Jan 29, 2017)

If this hasn't been suggested yet, I use Fabfilter Pro-L for this purpose and it works great. I also use their Pro-MB for more targeted and dynamic cuts. Very happy with the results.


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## Jediwario1 (Jan 29, 2017)

A limiter is great for taking off the transients like some have said already but another great tool is a Clipper, I use the free one by GVST
You have to be careful because you can quickly distort whatever your clipping, but often I find I get better results on some sources than using a limiter. This will work better with mid and high percussion rather than low stuff.


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## Daniel Petras (Jan 29, 2017)

Jediwario1 said:


> A limiter is great for taking off the transients like some have said already but another great tool is a Clipper, I use the free one by GVST
> You have to be careful because you can quickly distort whatever your clipping, but often I find I get better results on some sources than using a limiter. This will work better with mid and high percussion rather than low stuff.


The KClip from Kazrog is a very good, transparent sounding clipper. I might just give that one a shot for transient control on my busses for the next mix.

Here's the KClip V2 in action:


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## pixel (Jan 30, 2017)

I refer to too loud transients issue only: For fat drums tape saturation/compression is great option but it's not transparent. For transparency brickwall limiters (some are more transparent than others) and transient shapers. Compressors are rather too slow to deal with fast transients.
Good idea is too use combination of these techniques with low amount of reduction each.


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## KEnK (Jan 30, 2017)

Sonorityscape said:


> Isn't the first thing you mentioned more or less parallel compression where the dry signal gives you the transients and the over compressed signal gives you more of the body? Wouldn't your method just be equal to turning the dry knob down slightly?


Although you can route the tandem compressors in parallel, that's not not what I was talking about there.
I use the above method for 'transparent' peak lowering- especially useful on acoustic guitars.
Just a few dB, nothing drastic.

Another useful tool for this purpose is the spl transient designer


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## Joram (Jan 30, 2017)

Sonorityscape said:


> ...I noticed after the fact when comparing my mix to another that my percussion transients were quite dramatic and significantly reduced the amount of headroom I had to work with.


Can you give an example? It would make the problem a lot clearer for me.

btw: It is quite possible that compressors cause stronger transients. Many mix engineers use compressors not to get rid of dynamics but to shape the attack of sounds and make transients clearer and louder.

(nice tunes on your soundcloud, Daniel!)


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## Daniel Petras (Jan 30, 2017)

Joram said:


> Can you give an example? It would make the problem a lot clearer for me.
> 
> btw: It is quite possible that compressors cause stronger transients. Many mix engineers use compressors not to get rid of dynamics but to shape the attack of sounds and make transients clearer and louder.
> 
> (nice tunes on your soundcloud, Daniel!)


Thanks Joram. I was actually referring to this piece of music when I initially made the post:



Now looking back on it, I think I probably could've lowered the percussion volume a few db in most places. Especially in the initial stages where there's an interesting cello line that is getting buried. However, at the time I liked the power of the drums, but they were peaking too high on the meter and taking up a lot of head room, so I was looking for away to transparently reduce the peaks.

I've since tried to create a template that will be more conducive to better bus control of instruments and hopefully therefore better dynamic control.


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## Oliver_Codd (Jan 30, 2017)

A couple of options not mentioned already:

- If using samples, ADSR can work really well
- Surgical automation if the transients are not consistent
- EQ can help a lot, especially for high frequency stuff. Low Pass filters sometimes. 
- Adding reverb or delay will lessen the intensity of the transients

And of course, compression, transient designers, saturation, and balance. Also, monitoring very quietly can make it a lot easier to hear transient information.


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## pixel (Jan 30, 2017)

Joram said:


> btw: It is quite possible that compressors cause stronger transients. Many mix engineers use compressors not to get rid of dynamics but to shape the attack of sounds and make transients clearer and louder.



Exactly!


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## Joram (Jan 31, 2017)

Sonorityscape said:


> Now looking back on it, I think I probably could've lowered the percussion volume a few db in most places. Especially in the initial stages where there's an interesting cello line that is getting buried.


My thought exactly when I heard Overworld Action


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## Rob Elliott (Jan 31, 2017)

FF-L is exceptional. Also the transient designer IN Neutron (isotopes) is quite serviceable.


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## MartinAlexander (Mar 27, 2017)

Why sacrificing a transparent, dynamic and open sound just for loudness at all ?


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## charlieclouser (Mar 28, 2017)

MartinAlexander said:


> Why sacrificing a transparent, dynamic and open sound just for loudness at all ?



Because sometimes, some of us need to do music that's going to be behind a scene with machine guns, cars exploding, people screaming as their arms are getting torn off, and a giant machine that grinds up human bodies (not kidding). A "transparent, dynamic, and open sound" in the score will wind up sounding like cute little plinks and plonks, off in the distance somewhere, when it needs to sound like a massive, girthy slab of dark apocalyptic mayhem that's right up front and in your face - a churning, turbo-charged, music-grinding machine. 

Tonight's cue has distorted industrial programmed drum machines, epic war drums, clanking scrap-metal percussion, evil hammering synth bass patterns, death-metal guitars drop-tuned to C and run through bit-crushers, grinding little sounds from circuit-bent toy keyboards, deadly ominous brass swells, chugging low spiccato strings, insane swirling orchestral chaos effects, massive glissando risers, ultra-doom slams - all of these, all at the same time, with weapons set to "kill", not "stun". This cue has more than 70 stereo tracks.... of drums (!). With so many little molecules of sound flying around like a hailstorm, having limiters on each stem is how get them to "glue" together. You hear so much talk in record-mixing forums about people trying to find that "glue" compressor; that's why the old SSL 4k's "center compressor" is still used and emulated, both in hardware and software. But I've long moved past what the SSL can do - I need multi-band, look-ahead, brick wall, auto-makeup-gain limiters deployed at every point where signals are summed together or else everything just sounds like a bunch of tiny samples plinking away on the grid. With the amount and type of compression and limiting I use, sounds "push each other out of the way" as they battle for space in the mix bus, but there's no distortion or "analog mojo", just full-fidelity fist-fights at the summing point. Brutal beatdowns where only the strongest signals survive. It's fun to watch them fight!

I use limiting to have the effect of pushing up the quiet stuff so it can be heard over all that chaos, and to clamp down on "pointy" sounds during these super-dense passages. Even on quiet, floaty, ambient cues, this kind of limiting is very helpful, as it has the effect of "floating" the quiet stuff *upwards* so it can be audible and sound thick, and the effect of quiet sounds pushing each other around, gain-wise, creates all sorts of cool movement without manually automating things on the atomic level. A lot of stuff I do is "quiet sounds played loud", and massive compression and limiting is how I get there. I very rarely need to use automation for volume. About the only thing I actually use automation for is dub-delay feedback or synth parameters like filter sweeps. 

On huge beat-down cues the limiting is doing kind of the opposite - pushing down the loud stuff so it blends downwards and helps form that churning, rumbling murk with sharp transients that don't peak too far above the sludge. Multi-band limiters are critical for this, since whenever a sub-bass boom hits, that frequency range has its own band in the limiter and won't affect the other frequency ranges, like the "pitched" area of bass where cellos and stuff like that need to live. I think of the five bands in the multi-band limiter as, from low to high: "subs, bass, honk, clank, and fizz", and when the bands don't mess with each other I can sculpt things with much more precision than any full-range dynamics unit would let me. My band-split points are 80hz, 350, 1.25kHz, and 5kHz. It's fun to see the lowest band reducing by 12db when the sub booms hit and see no gain reduction at all in the next band up, so the smooth cellos can pass by unaffected. Apply this process across each stem and the composite mix and you can really apply serious pressure without ruining things (any more than my music is already ruined right from the start anyway!).

Even on individual tracks, I hit 'em hard with compression, and this can turn an electronic kick drum sample from a weak little "boof" into a juicy sub-bass monster. I audition and select sounds, compose, and mix with all dynamics processing turned on - I never "start the mix from zero" by pulling down the faders and turning off all the plugins and starting from scratch. All of my processing is so integral to the sound that I save channel strip settings for every sampler instrument with all the effects fully applied, so that it comes up already sounding the way I like it. 

Pressure - that's how I like to think. I apply pressure to my sounds. They're all guilty and need to be punished - none shall escape unscathed!

A "normal" score mixing engineer would raise eyebrows at how much compression and limiting I'm spraying across individual tracks as well as on each stem sub-master - but on the dub stage the music mixers always ask how I get things to sound so thick and turbo-charged, and I tell 'em I'm seeing 12db or more of gain reduction at multiple places in the signal path, and they're like, "WTF?!?! Well, whatever, man... it sounds great." On every project, I ask if I'm stepping on it too hard, is there too much reverb, is there too much sub-bass, do they want un-compressed print-outs of the stems.... and the answer has always been, "Nah, man... sounds fine... Stems in a straight line, faders grouped... It mixes itself." Either the mixers don't give a crap because they just want to drag this pig across the finish line, or.... maybe it actually does sound okay? I dunno... anyway, I like it. But my background is in making super-heavy-sounding records, not simulating lifelike, dynamic orchestral recordings - so there's that.

Sure, if you're doing Alexandre Desplat's score to "The Ghost Writer" or "Syriana" (both of which I love and envy) maybe you don't need to bring the hammer down so hard (or at all). But I don't get called for those kind of scores (sad!). I'm doing movies where too much is never enough. So it's a case of horses for courses.


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## Daniel Petras (Mar 28, 2017)

charlieclouser said:


> Because sometimes, some of us need to do music that's going to be behind a scene with machine guns, cars exploding, people screaming as their arms are getting torn off, and a giant machine that grinds up human bodies (not kidding). A "transparent, dynamic, and open sound" in the score will wind up sounding like cute little plinks and plonks, off in the distance somewhere, when it needs to sound like a massive, girthy slab of dark apocalyptic mayhem that's right up front and in your face - a churning, turbo-charged, music-grinding machine.
> 
> Tonight's cue has distorted industrial programmed drum machines, epic war drums, clanking scrap-metal percussion, evil hammering synth bass patterns, death-metal guitars drop-tuned to C and run through bit-crushers, grinding little sounds from circuit-bent toy keyboards, deadly ominous brass swells, chugging low spiccato strings, insane swirling orchestral chaos effects, massive glissando risers, ultra-doom slams - all of these, all at the same time, with weapons set to "kill", not "stun". This cue has more than 70 stereo tracks.... of drums (!). With so many little molecules of sound flying around like a hailstorm, having limiters on each stem is how get them to "glue" together. You hear so much talk in record-mixing forums about people trying to find that "glue" compressor; that's why the old SSL 4k's "center compressor" is still used and emulated, both in hardware and software. But I've long moved past what the SSL can do - I need multi-band, look-ahead, brick wall, auto-makeup-gain limiters deployed at every point where signals are summed together or else everything just sounds like a bunch of tiny samples plinking away on the grid. With the amount and type of compression and limiting I use, sounds "push each other out of the way" as they battle for space in the mix bus, but there's no distortion or "analog mojo", just full-fidelity fist-fights at the summing point. Brutal beatdowns where only the strongest signals survive. It's fun to watch them fight!
> 
> ...



Do you have any tracks available to listen to that show your approach explained here? I would be interested to hear something.


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## willbedford (Mar 28, 2017)

charlieclouser said:


> Perhaps someone else can suggest other, cheaper alternatives to the Waves L3 series?


LoudMax is good. Shaves off transients quite transparently. http://loudmax.blogspot.co.uk/


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## charlieclouser (Mar 28, 2017)

Sonorityscape said:


> Do you have any tracks available to listen to that show your approach explained here? I would be interested to hear something.



I don't have a website, but I have a few cues that I had to put on SoundCloud for various promo uses. Here's a few of the "squishy" ambient tracks that were still hitting a few db of limiting on every stem, and then again on the final composite mix. Some of these used my old favorite TC MasterX5 as the multi-band limiter, but now I'm using Waves L3-LL MultiMaximizer since MasterX5 is a dead product and only works on my old Snow Leopard / PowerCore rig:

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And I found a bunch of cues that people ripped and uploaded to YouTube - but who knows what the audio quality is like. Beware of the various "remixes" that people like to post; chances are I had nothing to do with them!

Here's an industrial cue from a movie called "The Collection" with some of that "hailstorm" approach:



Here's a quiet cue from The Collection that has a lot of compression just to get the quiet stuff to "speak":

https://www.youtube.com/watch?v=UanFF_NbSBo

Here's the main titles from a movie I did maybe ten years ago? Wide variety of sounds, big sections vs quiet sections, and mixed as just three stems in Logic with all stock plugins except for MasterX5 on each stem:

https://www.youtube.com/watch?v=UI2WuKFX7u0&list=RDUI2WuKFX7u0#t=63


This last one is a link to someone's YouTube playlist with a bunch of my tracks interspersed with all sorts of other junk, but there's a few heavy ones in there. The one that plays from this link is an ending theme montage from one of the SAW movies and has a lot of the elements I'm talking about - crazy metal percussion, big drums, digital-sounding guitars, and strings and stuff all through.

https://www.youtube.com/watch?v=rF_4uEtFYAs&list=PL94C19A8D9DE4703B&index=7

Here's a massive playlist with lots of crushed cues from the SAW movies:

https://www.youtube.com/watch?v=YixC8bASrcY&list=RDYixC8bASrcY#t=72

And here's a playlist with a bunch of cues from a movie called "Death Sentence" that's pretty stepped on in terms of compression:

https://www.youtube.com/watch?v=VemEQ3_9tYw&list=RDVemEQ3_9tYw#t=32

Just search Charlie Clouser on YouTube and you can find dozens of cues that people steal and upload - if it's not some fan-made "remix" or cover version, then I mixed it using the techniques I described. Also if you go to my agent's licensing company scorerevolution.com you can find tons of cues of mine from various horror movies - but weirdly they don't let you search by composer.


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## charlieclouser (Mar 28, 2017)

willbedford said:


> LoudMax is good. Shaves off transients quite transparently. http://loudmax.blogspot.co.uk/



From time to time AudioDeluxe.com has L3 on sale - I found it there once for $69. I think I've seen it on sale for similar prices on other "plugin deals" websites and even on Waves' site - they push me ads every weekend with special prices on their plugins, but I already have the Plutonium bundle or whatever it's called. Still, keep you eye out and you can probably avoid paying $349 for it.


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## Daniel Petras (Mar 28, 2017)

charlieclouser said:


> I don't have a website, but I have a few cues that I had to put on SoundCloud for various promo uses. Here's a few of the "squishy" ambient tracks that were still hitting a few db of limiting on every stem, and then again on the final composite mix. Some of these used my old favorite TC MasterX5 as the multi-band limiter, but now I'm using Waves L3-LL MultiMaximizer since MasterX5 is a dead product and only works on my old Snow Leopard / PowerCore rig:
> 
> <iframe width="100%" height="450" scrolling="no" frameborder="no" src=""></iframe>
> 
> ...



Wow, thanks for sharing! I love your stuff.


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## charlieclouser (Mar 28, 2017)

Sonorityscape said:


> Wow, thanks for sharing! I love your stuff.



Thanks man! Everything I've done before Childhood's End (last year) was printed as only three stems, all mixed inside Logic using stock eq (the normal one, not the linear phase one) and stock compressor and just the "Piano Hall 2.3s" preset in Space Designer for reverb and a stereo ping-pong delay on dotted eighth notes - with a few exceptions. I used that damn reverb preset for ten years! Every individual track or instrument is definitely hitting its own instance of the Logic stock compressor at 2:1 with zero attack time, platinum algorithm (no mojo), and at least 3-6db of gain reduction right from the start. So everything's getting bent a little bit before it even gets to the mix bus. That way I can play a single low note on a string section patch and it will be basically the same apparent volume as when I play a two-handed chord later in the cue, and I don't need to automate volume to make that happen. I'm lazy that way - plus I just like how it sounds.

Stems on most of those cues were each being crushed by TC MasterX5 and probably another dose of that on the composite mix as well. Often there's 6-12db (or more!) of gain reduction happening in one or more frequency bands at the stem sub master - but sometimes it's only one frequency band that's hitting that hard while the others pass through with much less crush. Sort of like a dynamic eq I guess? Any dub-delay-feedback effects are either Logic's Tape Delay or sometimes the green Line6 delay pedal. 

Ninety-nine-percent of the sounds are being played from Logic's EXS-24, and most of the synths are just ES-2. A lot of the low, softly pulsing synths are actually a string section sample in EXS played an octave below its normal pitch and with some LFO>Filter modulation to get the pulsing effect. That's a good example of why I love EXS - any patch has full access to the filter, LFO's, etc. right on the front panel - unlike Kontakt, where you've got to open the hood on the instrument (if it's even unlocked) and deal with dozens / hundreds of sample groups, even if all you want is to add a lowpass filter to an orchestral sound. Lots of (most) Kontakt libraries don't give you that stuff in their fancy user interface, and if you do go under the hood and make those tweaks, you've got to save the edited version of the Kontakt instrument. With EXS, all front panel tweaks are recalled with the song, so you don't run the risk of saving the actual instrument and thereby messing up other songs that use that same EXS instrument. For the way I like to work this is ideal, and it's why I spend way too much time and effort converting any libraries that I love to EXS format.

On the Childhood's End stuff I moved from three surround stems to seven when I got my MADI setup going. Before that I was using 24 channels of LightPipe to get from the Logic machine to the ProTools print rig, so three stems plus a composite mix was the maximum. With MADI I can pass 64 channels across, so I use 48 - that's seven 5.1 stems plus a 5.1 composite mix. 

Also on Childhood's End, and on a couple of the Wayward Pines cues there is some legato solo cello - that's not EXS24, that would be the CineSamples Tina Guo legato patch, probably layered with Spitfire solo cello and / or Emotional Cello from Best Service. The mournful solo female vocals are from Heavyocity's Gravity expansion pack Vocalise - another late-night impulse purchase that saved my butt on those emotional cues, but I think I used up every decent sample in there! The atonal and aleatoric orchestral and choir effects come from every which where - some of those "Ligeti" choir fx, like the "dissolving descending wide vibrato" go all the way back to the ProSonus / audio sampling CD era, and some of the swirling chaos strings came from a Synclavier library back in the 1980's. I'm a fiend for that stuff and collect every single library that might have two or three useful samples.

There's such a mixture of sample sources in those cues - tiny little 100 millisecond bursts of guitar pitched down, little blasts of harsh audio I sampled from old Japanese cartoons back in the 1980's, fancy expensive orchestral libraries, old favorites like Kirk Hunter's ancient string libraries for the Akai S-1000, even bits of string section samples from my old Ensoniq Mirage. I'm pretty meticulous when I audition, name, map, and organize my samples, and I never throw anything away!


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## FredW (Mar 28, 2017)

charlieclouser said:


> I don't have a website, but I have a few cues that I had to put on SoundCloud for various promo uses.


Nice share Charlie! I found some Hi-Res versions of some of your scores over at 7digital the other day when I was looking for some lossless reference tracks. These are all flac 16 bit/44.1 khz, but I assume you produce in 24 bit/48khz for the dub stage? Are you using L3-LL differently in a case like this when the music is going to be played on two different platforms?


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## charlieclouser (Mar 28, 2017)

FredW said:


> Nice share Charlie! I found some Hi-Res versions of some of your scores over at 7digital the other day when I was looking for some lossless reference tracks. These are all flac 16 bit/44.1 khz, but I assume you produce in 24 bit/48khz for the dub stage? Are you using L3-LL differently in a case like this when the music is going to be played on two different platforms?



What the heck is 7digital? Never heard of that site. I wonder if I'm getting paid for those? Ehh... who cares.

Anyway.... I don't make any concessions for various destination formats. I just do my thing at 48k / 24 bit and then when the time comes to make a stereo mix for CD or whatever, I stay at 48k while I bring in my original composite mixes that were printed along side the stems. These are just a straight sum of the original compressed stems. I then make a new Logic session to edit and tweak, and I do any edits, like shortening cues or combining multiple cues together. Then when everything is edited, I export those versions as-is, so if they're going to be mastered by a proper mastering engineer they've got those just in case. I then take a pass at my home-brewed mastering, which is basically just automating the compression threshold in Ozone or MasterX5. In quiet passages I'll lower the threshold to dig into the mix more and bring up the level of those passages, and on louder passages I'll raise the threshold so it doesn't bite down too hard. Since I'm usually using some sort of auto-makeup-gain thing like Ozone's Maximizer module, I always wind up right at -1db. Then I export those versions to a 48k / 24 bit folder. Then I convert sample rate down to 44k and bit depth down to 16 bit if needed, right from within Logic.

I don't use dither. Never have. Don't like the idea of adding noise to my signal for any reason, even if that supposedly will make reverb tails smoother at they pass below -72db or whatever. I know lots of plugins like Ozone and L3 have dithering options, noise shaping settings, desired output bit depth, blah blah blah - I ignore it all. Bypassed. I don't use fancy third-party sample rate conversion algorithms or anything, I just use the "Copy/Convert" function inside Logic's audio pool window. Simple. Sounds fine. I also don't buy into the argument that you need to print your mixes at -9db or whatever so the mastering engineer "has some headroom to work with". Uhhh....hello? I think the mastering engineer probably has a way to reduce the level of the track on the way into whatever processor they're using that might wind up boosting the level. One more digital gain change ain't gonna hurt anything, not on my mixes anyway. I do use iZotope's "Insight" metering plugin (or just the meters within Ozone) to check for any inter-sample peaks, since I'm so close to full-scale signal, but it's never found any. Whatever limiter I'm using - MasterX5, L3, Ozone - is smart enough to catch all that stuff. I've been printing everything at within 1db of full-scale for decades and no problems yet. Back in the album years, when I printed mixes from analog consoles to DAT tapes on the Panasonic SV-3700, I used the little Apogee A>D unit with SoftLimit™ and those LEDs were definitely lighting up.

If the tracks for a CD or download / streaming release are *not* going out for mastering, then we use my home-mastered versions that were sample-rate and bit-depth converted from within Logic. If they *are* going out to mastering and QC then I give them all four versions - dry and home-mastered, in both 24/48 and 16/44 versions. Let them sort it out. Can't really remember which they wind up using, but it always sounds okay, I guess, so.... job done. On to the next.


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## zvenx (Mar 28, 2017)

Just want to once again say thanks Charlie. Reading your posts are not only very informative and educational for me, but strikes me how willing you are to share your vast knowledge and experience.
Huge Kudos for that.
thanks
rsp


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## charlieclouser (Mar 28, 2017)

zvenx said:


> Just want to once again say thanks Charlie. Reading your posts are not only very informative and educational for me, but strikes me how willing you are to share your vast knowledge and experience.
> Huge Kudos for that.
> thanks
> rsp



Hey, you're welcome to it. Beware though - I make no pretensions that my opinions are correct, they're just that - opinions. I may be completely wrong on dither, noise shaping, what sample rate converter to use, how much compression is "okay" etc. - but for my music, it sounds fine, and these are the techniques I use to get wherever the heck I get to. But my music represents a tiny slice of the broad spectrum of sounds that people might want to achieve. I seriously doubt that anyone could just do everything I talk about and "boom" - magic is gonna happen.

But I do see lots of forum posts from folks who seem to be about to get really worried about stuff that is going to take 30% more effort and expense to get 2% better results. I've gone down a few of those dead-end trails over the years - like fancy word clock generators - and for sure some are worth pursuing, like nice mic preamps and DI boxes, etc. But only if you can actually hear the difference and the extra effort is worth it to you.


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## FredW (Mar 29, 2017)

charlieclouser said:


> What the heck is 7digital?


Heh, yeah I found lots of these Hi-Res distributors that I never heard of, but I have to admit that I don't hear much difference in quality when I stream the same thing from Spotify.



charlieclouser said:


> I don't use dither. Never have.


Very interesting read, I have mostly been using Logic's multipressor or just the adaptive limiter on my stems until now. Got the L3 during one of those crazy deals some months ago and realized after reading your post that dither seem to be active on default. Will flip that off.


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## charlieclouser (Mar 29, 2017)

FredW said:


> Got the L3 during one of those crazy deals some months ago and realized after reading your post that dither seem to be active on default. Will flip that off.



Hey, you might want to read up on dither and noise shaping before turning it off! I did a little reading, got confused, did about five minutes of A/B testing, couldn't tell if I was really hearing any difference, and said, "screw this" and turned it all off.

But depending on your program material and delivery formats you may find a real benefit from these features. 

I think it's okay for me to not use that stuff because my music is most often a big solid brick of sound and very rarely am I dealing with a mix that has smooth reverb tails that gently decay down to low levels. Perhaps someone who actually understands dither and noise shaping will chime in and educate me on that stuff?


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## dgburns (Mar 29, 2017)

charlieclouser said:


> Hey, you might want to read up on dither and noise shaping before turning it off! I did a little reading, got confused, did about five minutes of A/B testing, couldn't tell if I was really hearing any difference, and said, "screw this" and turned it all off.
> 
> But depending on your program material and delivery formats you may find a real benefit from these features.
> 
> I think it's okay for me to not use that stuff because my music is most often a big solid brick of sound and very rarely am I dealing with a mix that has smooth reverb tails that gently decay down to low levels. Perhaps someone who actually understands dither and noise shaping will chime in and educate me on that stuff?



You only want to dither when reducing bit depth, so going from 24 bit to 16 bit and only once, so at the very end. What you're doing is trading noise for distortion (trying to get rid of truncation distortion). You never want to dither inside a multitrack session. There is nothing to gain from dithering a file that stays at the same bit depth.
The noise shaping component of dither algo's is done in order to try and move the noise out of the range that we are better at hearing, so it makes it harder to hear the noise the dither is applying.

I don't know for sure, but any daw that has a float engine and records pcm files might in fact be applying dither in the conversion stage when writing the files to disk during the recording, as the floats are at a higher bit depth.


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## charlieclouser (Mar 29, 2017)

Okay, that makes sense - question is: this truncation distortion that dither is trying to reduce - is it present across the whole dynamic range of a signal, or only on low-level stuff? In other words, will correctly-applied dither increase fidelity even on loud signals, or is it primarily meant for situations when signals get quiet (like those reverbs decaying to silence I was talking about earlier.)?

Obviously, when signals are super-low-level in a 16-bit depth file, they'd effectively be "3 bits" or something, whereas in a 24-bit file they'd be "7 bits" or something (my math is not correct!). Is this the only time when dither helps, or does it help even when stuff is within a few db of full-scale?


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## dgburns (Mar 29, 2017)

Some, like Paul Frindle of Sony Oxford and SSL fame, claim that yes quantization distortion is apparent at normal listening levels. From what I gather, if you run at 32 bit float (like inside a daw), and all your plugins do the same, and properly (which actually may be not true), then there is no need for dither. Also, from what I read, triangular noise is likely the best to use as it's possible to dither at a few points in the production process without worrying to much. For example, if you bounce down to 24 bit pcm, you could use triangular dither there, and once again if you went down to 16 bit pcm.
I've not really heard a huge change at 24 bit myself, but when bouncing to 16, boy does the dither you use change stuff, and the powr dithers are very different, I need to be careful using anything but powr 1, cause the other two change the sound too much. Ironically, it's more apparent to me on loud music then very dynamic music.

Here's a link to some of Paul's thoughts-

http://productionadvice.co.uk/when-to-dither/

Hey, just want to add,if you have recordings made in the real world, ie stuff you recorded and there's a baseline noise floor, it can act like a form of natural dither.


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## charlieclouser (Mar 30, 2017)

Thanks for that info dgburns - that, and the article from Paul Frindle, makes things a little more clear.


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## Daniel Petras (Apr 13, 2017)

charlieclouser said:


> It's fun to see the lowest band reducing by 12db when the sub booms hit and see no gain reduction at all in the next band up, so the smooth cellos can pass by unaffected


Do you mean that you have the booms sharing the same buss as the cellos or how do you decide what shares a buss?


charlieclouser said:


> I audition and select sounds, compose, and mix with all dynamics processing turned on - I never "start the mix from zero" by pulling down the faders and turning off all the plugins and starting from scratch. All of my processing is so integral to the sound that I save channel strip settings for every sampler instrument with all the effects fully applied, so that it comes up already sounding the way I like it.


This makes so much more sense than what I've been doing. Or at least addressing the major dynamic processing in the beginning of the mix rather than at the end which it seems is totally going to mess everything up. Gonna really give this method a shot since I have not been happy with my end result loudness.


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## charlieclouser (Apr 13, 2017)

Sonorityscape said:


> Do you mean that you have the booms sharing the same buss as the cellos or how do you decide what shares a buss?
> 
> This makes so much more sense than what I've been doing. Or at least addressing the major dynamic processing in the beginning of the mix rather than at the end which it seems is totally going to mess everything up. Gonna really give this method a shot since I have not been happy with my end result loudness.



In some cases, like when delivering tv scores with only three stems or even just stereo mixes, or preparing stereo mixes for CD releases or whatever, then, yes - the sub booms will be hitting the same limiters as cellos or other tonal bass instruments. Happens all the time. For the film I'm doing now I'm delivering seven stems, and I kind of prefer doing three stems since the more stems I give, the more things sound "un-glued" and separate. I prefer when everything is all mushed together and you can't slide a piece of paper in between the bricks. That makes for more work on my end to get it to sound that way, but less hassle in terms of re-routing everything to separate stems. When it's just three stems the mix just flies along - drums, keys, orchs = done. 

Breaking every element out into a separate stem makes more work for the mixers, although it does give them more control, but it's also more hassle in some ways on my end too. Every time I want to move a track from one stem to another, it takes five click-n-drag operations - one for re-assigning the track's destination, and four to re-assign the sends to the appropriate front/back reverb+delay that go to the desired stem. Hassle. If I stopped using send-based reverb and delay this would only be one click, but then I'd have hundreds of reverb and delay instances eating up CPU.

I decide what elements go to what stem on a cue-by-cue basis, since most of my music is not a standard set of sounds but rather a wild, lopsided combination of whatever I felt sounded good that day. In theory, the names of my stems are:

A - drums
B - percussion
C - metals
D - keys
E - strings
F - brass
G - orch fx

But a quiet emotional cue might be laid out like this:

A - soft sub booms
B - three little bowed metal squeaks
C - low grinding guitar tremolo
D - low piano notes played by finger tapping on strings
E - main piano with black hole reverb
F - low meaty cellos+basses
G - high sul pont tremolo strings

While another cue might be laid out like this:

A - full-on drum mayhem with kicks, subs, war toms, etc., with subs and kicks going to LFE.
B - top kit percussion with metals, hi-hat-like things, backwards cymbals, etc. 
C - bangs, slams, sucks, and doom hits, some going to LFE.
D - bowed metals, spooky guitar effects, weird ambiences, basically anything scary and not percussive.
E - synth bass and super-heavy guitars in different parts of the cue.
F - all strings and brass that are not effects, so highs and lows and brass swells - all of it.
G - all the chaotic orchestral effects - dissonant sustains, atonal shrieks, psycho string fx, symphobia slams, etc.

Sometimes the content of the cue lines up with the names of the stems, sometimes it doesn't. That's why I stopped naming the stems that I deliver to the stage with "drums, perc" etc. - now I just call them "A stem, B stem, C stem" etc. and I include a ReadMe file telling the music editor what the hell is going on with each cue. But on some cues there's just no avoiding having big subs on the same stems as pitched, tonal bass elements - so that's what I was talking about with the five-band limiters. I try to separate things into stems in a way that will give them the most control on the dub stage, even if that means disobeying my own "rules" for stem names. If a cue has just some low strings notes holding the bottom down, with some high sul pont tremolo and a few brass swells here and there, then I can lump it all together and free up a stem for more separation of other elements. So I decide on the fly most of the time.

I just don't like seeing cues get printed with three stems full of stuff and four stems mostly empty, so I move stuff around to put something on each stem.


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## Vischebaste (Apr 14, 2017)

Hi Charlie, 

Your posts in this thread have been amazingly open and detailed about your approach and refreshingly unapologetic in the way you've developed your own unorthodox working methods that yield great results to give your sound its own signature. Some of the most inspiring posts I've read on this forum (and pretty unexpectedly, given that the topic was only about reducing transients). Thanks!


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## Daniel Petras (Apr 14, 2017)

Great video on transparency and how to gain more headroom. The big tools used here are saturation, a lot of parallel processing and limiting:



For me the visual part of analyzing the waveform in order to properly mix it is important. For the past little while I've been using Pro-L for the visual representation of the wave since it doesn't require any rendering to actually see changes to the waveform. Does anyone else have a tool they use for this purpose?


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## AdamAlake (Apr 14, 2017)

Parallel compression.


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## macmac (Apr 14, 2017)

Charlie, do you have plugins on your master buss? If so, how do you handle printing the stems so that the combined stems sound like the full mix (e.g. if the editor wants to re-combine or make edits), and not over-processed by having each stem run through the same processing? Thanks.


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## givemenoughrope (Apr 14, 2017)

^I thought he had the multiband limiters on each stem....? I wonder how well the Waves limiter does as far as eating cpu and latency.


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## FredW (Apr 14, 2017)

givemenoughrope said:


> I wonder how well the Waves limiter does as far as eating cpu and latency.


L3-LL has a latency of 64 samples (48khz) = 1,3 ms - So extremely low latency and on my old 2010 iMac I don't notice any difference in CPU load when I bypass four instances of the L3-LL that I use in one of my templates.


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## charlieclouser (Apr 14, 2017)

macmac said:


> Charlie, do you have plugins on your master buss? If so, how do you handle printing the stems so that the combined stems sound like the full mix (e.g. if the editor wants to re-combine or make edits), and not over-processed by having each stem run through the same processing? Thanks.



I only have plugins on my individual stem sub masters - and they all have identical settings most of the time. The final composite mix has no further processing. 

Typical settings for a cue that is absolutely slamming on all stems - threshold between -9 and -12, output ceiling at -9 to -12. That way the apparent loudness of stuff isn't changing by a huge amount, but that top 9 to 12 db is getting utterly clamped down. When seven stems like this combine, the final composite mix (hopefully) won't clip. If it does, I just back down the output ceiling on all stems by the same amount until there's no more clipping.


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## dgburns (Apr 15, 2017)

charlieclouser said:


> I only have plugins on my individual stem sub masters - and they all have identical settings most of the time. The final composite mix has no further processing.
> 
> Typical settings for a cue that is absolutely slamming on all stems - threshold between -9 and -12, output ceiling at -9 to -12. That way the apparent loudness of stuff isn't changing by a huge amount, but that top 9 to 12 db is getting utterly clamped down. When seven stems like this combine, the final composite mix (hopefully) won't clip. If it does, I just back down the output ceiling on all stems by the same amount until there's no more clipping.



I hit the individual tracks harder if need be. It's cumulative, and I think you can hit the individual things harder and get away with it rather then slamming the stems in the hopes of staying within bounds. Also, remember that the cues should marry up with dialog at some point, so I tend to monitor with dialog up as I write. It avoids alot of issues later (no that slamming hit doesn't fly when the ppl are whispering)
I also pissed around with sending the full mix to a bus that then fed each individual compressor on the stems, so kinda sidechaining the comps to react to the full mix. Surprisingly, I felt luke warm about the result, even though it did contain things well, like you were doing a single stem. The problem with this is the individual stems can get squished because of the other stems, and when/if the other stems are pulled from the scene, the score kinda sounds funny in a bad way, especially high pedal strings getting sucked down by some phantom drum track.
I would say that if your individual stems are not clipping, you're good, because the mixer can pull down the stems and put some clamping limiter on the full mix anyway. Like I said, it all starts with the individual tracks first.
my two cents


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## charlieclouser (Apr 15, 2017)

Yeah, I tried the trick of side-chaining each stem's compressor from the composite mix - and like you, I think it can create more problems than it solves. Maybe it would be more appropriate on a big orchestral score where you know they're not really going to be chopping the heck out the stems on the stage, but on most of my stuff they're likely to mute a stem or three, or grab one piece of a stem from one cue and slide it in somewhere else, etc. 

I mean, technically, it "works" but like you said, then the individual stems have some dynamic control going on that's not directly related to their own content, and I think that's problematic.

I do hit the individual elements pretty hard for sure, but that's more of a way to get each one to speak the way I want as opposed to a method of controlling overall mix dynamics. Usually zero attack time, 2:1 or 4:1 ratio, and as quick a release as I can use without chattering on low frequencies. The only time I use a non-zero attack time is when I'm trying to do some sort of "transient shaper" effect and get some pop on the front of a drum or whatever, but as often as not I just use an actual transient shaper plugin (if it's an audio recording) or just the envelopes on the sampler.

I still get those magic moments where huge distorted guitars, a live bass, a synth bass, and some synthetic blast all hit the limiter just right and combine into the meaty chunk that makes me think, "okay, this sounds okay" - when five seconds ago they just sounded like four separate elements. The sweet spot in situations like this is pretty narrow though.

The main reason I cap all my stems at -9 to -12 is so that I can make a non-clipping composite mix right then and there, without having to trim everything down in level somewhere else in the chain. I don't think the music mixers mind that the score stems are each 12db down as opposed to right up there at the ceiling - they're trimming the levels down on their end anyway.


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## brett (Apr 15, 2017)

@charlieclouser - you mentioned that you automate the threshold of your limiter / compressor for CD releases, but do you do the same for film/tv cues?

In other words, if you have a dynamic cue with a quieter suspense textural first half leading into a full on action chase second half do you ensure both halves of the cue get their share of gain reduction in certain frequency bands? Or do you set and forget across the whole cue such that either the quieter first half misses out, or the loud second half gets ridiculously slammed? What about changing or automating threshold between different cues?

I've always used a bus compressor and automate the threshold so that even in the quieter passages I can add a couple of db compression if I wish - it helps glue the whole cue together rather than just the louder passages. But I've been reading your posts with interest and am interested in multiband compression / limiting so that the low boomy stuff doesn't clamp down on the rest. Also, while I've considered moving my bus compressor to individual stems, automating threshold on every stem compressor within and across multiple cues just feels like is be making work for myself

Interested to hear how you approach this, and apologies if you've already canvassed this above or elsewhere

Cheers


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## brett (Apr 17, 2017)

@charlieclouser bump


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## charlieclouser (Apr 17, 2017)

brett said:


> @charlieclouser - you mentioned that you automate the threshold of your limiter / compressor for CD releases, but do you do the same for film/tv cues?
> 
> In other words, if you have a dynamic cue with a quieter suspense textural first half leading into a full on action chase second half do you ensure both halves of the cue get their share of gain reduction in certain frequency bands? Or do you set and forget across the whole cue such that either the quieter first half misses out, or the loud second half gets ridiculously slammed? What about changing or automating threshold between different cues?
> 
> ...



Most of the time I'm set-and-forget on automating thresholds on stem sub masters on individual cues - it seems like it's more "okay" to leave some dynamics for the mixers to work with on the dub stage, plus when the cue is sitting in front of me and I have access to every track I can control stuff better than when dealing with a stereo mix after the fact. Plus the dynamic range you want for a CD that's made "just to listen to" versus a mix for the stage where there will be dialog and fx on top of the music is a different beast - so I can squash a CD mix even more than a stage mix in order to get those tiny moments to be legible, when on the stage they were only meant to be little tendrils of background bowed metals or whatever.

I also break up cues quite a bit. I just finished an eight minute monster that had two huge industrial sections with quiet ambient bits in between, so I broke it into five pieces and mixed them separately. I had the entire thing mostly programmed and mixed, and I just did a "save as" five times and dealt with each chunk separately. Not the same as automating the limiters, but a similar effect - the loud pieces had threshold at -9 and ceiling at -12, and the quiet pieces had threshold at -12 and ceiling at -9 - so that's sort of like automation I guess. The cues all overlap by maybe ten seconds or so, and then the loud piece can be just crushing it and when it rings out the following quiet bit can come in "fatter".

This also means the mixes are checkerboarded on the playback system on the stage, so they can use automation or clip gain in ProTools to make incoming / outgoing cues mesh how they want - and if I need to make changes to one section I don't have to re-print the whole eight-minute monster. 

But yeah, automating fourteen instances of L3-LL is too much like actual work. Ain't nobody got time for that!


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## JBW (Apr 21, 2017)

charlieclouser said:


> From time to time AudioDeluxe.com has L3 on sale - I found it there once for $69. I think I've seen it on sale for similar prices on other "plugin deals" websites and even on Waves' site - they push me ads every weekend with special prices on their plugins, but I already have the Plutonium bundle or whatever it's called. Still, keep you eye out and you can probably avoid paying $349 for it.



Just happened to see this thread... And it's on sale at audiodeluxe for less than $49 right now.


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## charlieclouser (Apr 21, 2017)

JBW said:


> Just happened to see this thread... And it's on sale at audiodeluxe for less than $49 right now.



Geez, for $49 it's a no-brainer to add L3 to your collection. Might not be to everyone's taste, but it's a powerful thing for sure.


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## Phillip (Apr 22, 2017)

Isotope Ozone Limiter


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## brett (Apr 22, 2017)

Waves sent me an email which asked me to do a survey for a $10 off code and so I picked up both C6 and L3 for $76.30

Not too shabby


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## Daniel Petras (Apr 24, 2017)

This track is insane and at one point it goes -5 RMS. How the hell do you do that?


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## Serg Halen (Apr 25, 2017)

Sonorityscape said:


> This track is insane and at one point it goes -5 RMS. How the hell do you do that?



Actually.. This is pretty ez, just put compressor on master and turn down threshold, then knob gain turn up. After compressor put limiter. Profit.
A little example


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## dgburns (Aug 30, 2017)

A little novelty trick of mine is those Prism Overkillers. But you have to go out the daw and back in.


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## Nick Batzdorf (Aug 30, 2017)

Sounds like some peeps here would really benefit from playing with a compressor/limiter on both mixes and individual instruments. Getting that into the ears is one of the most important parts of mixing.

Some of the applications:

Limiting (10:1 input:output ratio or higher) frees you from setting the level of the whole mix relative to the highest transients. You can also use a limiter to make percussive sounds crunchy, which works really well in a parallel compression setup as Kenk suggested - the original sound has the dynamics, but you mix in the hard attacks.

Bus compression - smoothes out the whole mix. Narrowing the dynamic range can also let you push up the bottom, which increases the density. Usually slow attack and release.

Individual instrument compression can get a smooth, even sound. It can let you shave off the attack, for example Paul McCartney's bass sound on a lot of songs. Or you can exaggerate the attack (fast release).

And then you get to multiband stuff... but the main point is that buying plug-ins is great, but it's only part of it.


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## aumordia (Aug 30, 2017)

Lessons from the "make everything loud as possible" branches of EDM would be:


Multiband compression, working both downwards AND upwards, especially on the low end.
Multiband soft-clipping, i.e., distorting each band separately in (you can usually crush the mids, low and highs need a lighter touch), then possibly the whole thing together.
Perhaps most importantly, ducking! AKA side-chain compression. Make the rest of the track duck behind the key percussion elements. Vary the degree of ducking based on the material -- e.g. duck bass and pads a ton, leads just a touch. Not only does this create tons of headroom, but it plays a psychoacoustic trick -- any drum sounds yuuuge when it causes everything else to temporarily get quiet to make room for it.


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## dgburns (Aug 30, 2017)

So the thread title is reducing transients transparently, and I have yet to find something like the overkillers in software save for a few clipper plugins. Those overkillers are a simple diode circuit but they do wonders on the right material.
Also don't know if it's been mentioned, but the excellent Drum Leveller by Sound Radix is really very good at allowing you to set each drum hit as well. It's on my "to buy" list for sure.

In the picture below, same song, same section, one with overkillers on the output of my Neve8816, one without the overkillers. There's a bit of volume loss, but the transients are contained and it doesn't sound "compressed" in any way. Maybe it's not your thing, but it's a useful tool in the arsenal. And one that wasn't mentioned here till now.


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## whoot whoot (Aug 30, 2017)

I'd suggest watching the FabFilter Videos on Compression Techniques, also the UAD Compression overview videos. There are standard ways of dealing with Transients - Transient shaper plugins - and then there are older more tried methods of dealing with the transients + opening moments of a sound source like the LA2A and 1176 emulations. 

There are lots of great videos talking about how to handle, cut and boost tranisents by using compressors on Dave Pensado's youtube channel and the above mentioned.

I guess what I am saying is: it is more of a technique thing to learn than to buy.


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## karelpsota (Aug 30, 2017)

Daniel Petras said:


> This track is insane and at one point it goes -5 RMS. How the hell do you do that?




This track is extremely well engineered. Attila spent almost a year on it. The production and sound selection is 90% of why it works.

Mastering wise, you need to control the low end very well if you want to achieve loudness. You also can't stack too many things. Less is more as usual.

Titan must fall has barely any low end. No bass line, just a glimpse of sub under the hits every bar. The hits sound huge due to harmonic exciters. All the energy comes from tasteful distorsion and good orchestration. The tonal anvil also help to enrich the hits, so it doens't sound like noise.

Try using pro-MB to control your lowend, softclip for a bit distorsion, then push as much as you can into a good limiter. (I believe Attila uses the AOM Limiter)


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## Nathanael Iversen (Aug 30, 2017)

Cubase has a look-ahead limiter built in - has a handy little graph that shows it pushing down the peaks. If you have Cubase, you don't even have to spend $$...


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## Daniel Petras (Aug 31, 2017)

karelpsota said:


> This track is extremely well engineered. Attila spent almost a year on it. The production and sound selection is 90% of why it works.
> 
> Mastering wise, you need to control the low end very well if you want to achieve loudness. You also can't stack too many things. Less is more as usual.
> 
> ...



Wow a year!?

Yeah, I've learned soo much about low end since I've made this post. And my mixes are constantly getting better and louder as well. A lot of my EQing now simply consists of cutting the low end where it is not needed.

One of the things I still find difficult is crafting really nice trailer hits. I've definitely started to do a lot more processing and parallel stuff, but it still requires really great sounds through the spectrum to get something hard hitting, but with good character.

Edit: Btw which clipper do you recommend? I started using this crispy present on FF Saturn which did really nice things for a hit I was working on. I tried it on this track I did and the hits sounded a lot snappier, though I think there are things I could do to make them bigger (if you have any suggestions).


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## Daniel Petras (Aug 31, 2017)

Nathanael Iversen said:


> Cubase has a look-ahead limiter built in - has a handy little graph that shows it pushing down the peaks. If you have Cubase, you don't even have to spend $$...



I just realized FF Pro - L has that and I've never used it. I'll mostly use a looked ahead if I'm side-chaining something and I want it to get out of the way quickly.


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## storyteller (Aug 31, 2017)

For anyone interested in @charlieclouser's technique, the seemingly-never-ending-series-of-waves-sales has L3 complete for under $23 with CK901 discount code (on the plugin sale sites). Supposedly the sale ends today... but, its waves, so who really knows.  Killer deal though for sure.


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## karelpsota (Aug 31, 2017)

Daniel Petras said:


> Edit: Btw which clipper do you recommend? I started using this crispy present on FF Saturn which did really nice things for a hit I was working on. I tried it on this track I did and the hits sounded a lot snappier, though I think there are things I could do to make them bigger (if you have any suggestions).




GClip is really good. I know Joshua Crispin uses it too.
http://www.gvst.co.uk/downloads.htm

Also its free 

try pushing single tracks, busses, and masters as much as you can before the distorsion is audible. You'll reduce the crest factor without any pumping or "compressed" effect.

Sometimes, if the master clipper is distorting. Leave the clipper ON and work in the mix to reduce clashing elements. (often comes from hit, bass conflict).


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## charlieclouser (Aug 31, 2017)

I have recently figured out some settings on Ozone's Harmonic Saturator module (or whatever it's called) that don't wreck things too much: I use the "tape" model with intensity set to about 5 out of 15 and mix at 50% on all four bands, tweaking band crossover points, mix, and intensity a little as needed. This puts back a bit of grind and sparkle that compensates a little for what is lost by heavy limiting. Many of the other saturation models seem to break up too easily, but I like what the tape model does.

Saturn is also a great processor, but has so many options that it takes a minute to get the hang of it. I've used it on the short strings stem to get a bit of bite and get the shorts to cut through the mix in dense passages.


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## Daniel Petras (Sep 1, 2017)

I've recently discovered that people will do 2 stages of mastering where they will have mastering on the on the mixbuss (which If I'm correct, is the final output for all tracks) and then will do mastering on the printed track. Is this something that people often do? I've tried to do my mastering on the mixbuss stage as the final master to save time, but for some reason it has a different effect from rending and then doing the mastering on the printed track. I have not really figured out why.

Here's a quote from what Mick Gordon does:

Mastering: it depends on what we sort out with the Audio Director. Each game is different and games are still trying to figure out their ideal loudness standard, so whatever I do usually has to line up with that. Some games, I've made the action music quite dynamic with a loudness range between 6-12db. Other games, I've brick-walled the music and the volume of the music changes according to the in-game mix engine (so, when SFX play the music gets ducked, etc). Depends on the game. I master it myself - often there's multiple versions that get changed right up until we release the game (and even sometimes afterwards...) so I can't keep sending it out to someone externally. Mastering is really basic. I have two stages: MixBuss: usually a buss compressor into an EQ. I usually use either the Slate Buss Compressors but I'm loving the UAD2500 now. That goes into the two UAD Pultec Plugins, adding 3-5db at 60hz, a broad shelf of 2-3db from 3k upwards. Then, I have the midrange EQ adding about 2 db of 1k, or 1.5k (depending on the track). That's just the overall curve I like for most stuff. Mastering: Depends on the track. It always starts with the Waves Low Band LinEQ shaving off everything below 32hz or so. That could go into any combination of things depending on the track (usually the Shadow Hills, maybe another EQ, etc). It always finishes with a limiter. I audition several limiters each time, trying to find a nice balance of control and dynamics. I don't like limiters that "pump". The DOOM stuff was mostly clipped through a converter for limiting, just cause the music is stupid and aggressive.


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## charlieclouser (Sep 1, 2017)

I do the "mastering twice" thing to some degree - stems delivered to the dub stage have per-stem limiting from Waves L3-LL Multi only, since the composite mix is not usually used on the dub stage, and there will probably be some additional dynamics processing on the final mix from the stage as well.

But when I prepare the composite mixes for release on streaming / CD / vinyl I take another pass with Ozone and other goodies - on my latest score (JIGSAW) this was a +3db hi shelf above 6k, an "infinite slope, zero resonance" hi-pass at 20hz, Ozone's Harmonic Saturator on the "tape model" setting with intensity at 5 out of 15 and mix at 50%, and up to about 3db of "peak shaving" from Ozone's Maximizer with a little transient recovery and the most advanced release algorithm (ARC IV in classic mode I think?). I set the Maximizer ceiling to -.5 db just in case of inter-sample peaks even though it's set to catch those. I automate the threshold of the Maximizer to "chase" the level of the mixes so I can bring up quiet passages and back off on loud stuff - the range might vary by up to 12db across a long cue.

My scores are always mastered by a real mastering engineer for release on CD / streaming / vinyl after I do this, and I always send my Ozone-ed versions as well as the dry mixes that are just a straight-fader composite of the already limited stems as delivered to the stage. On this last score, the mastering engineer really liked my Ozone-ed versions and used them instead of starting over with the dry versions, only adding "a little bit of presence" as he put it. 

So I guess home mastering with Ozone passes the smell test.


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## dgburns (Sep 1, 2017)

charlieclouser said:


> I do the "mastering twice" thing to some degree - stems delivered to the dub stage have per-stem limiting from Waves L3-LL Multi only, since the composite mix is not usually used on the dub stage, and there will probably be some additional dynamics processing on the final mix from the stage as well.
> 
> But when I prepare the composite mixes for release on streaming / CD / vinyl I take another pass with Ozone and other goodies - on my latest score (JIGSAW) this was a +3db hi shelf above 6k, an "infinite slope, zero resonance" hi-pass at 20hz, Ozone's Harmonic Saturator on the "tape model" setting with intensity at 5 out of 15 and mix at 50%, and up to about 3db of "peak shaving" from Ozone's Maximizer with a little transient recovery and the most advanced release algorithm (ARC IV in classic mode I think?). I set the Maximizer ceiling to -.5 db just in case of inter-sample peaks even though it's set to catch those. I automate the threshold of the Maximizer to "chase" the level of the mixes so I can bring up quiet passages and back off on loud stuff - the range might vary by up to 12db across a long cue.
> 
> ...



Logic's new loudness meter is awesome. You can now get to a definitive LUFS reading, say -14 (or whatever) without having to guess. Also the Level meter now has "True Peak" metering, so you don't have to guess on intersample peaks. The integrated let's you see infinite, or over the whole program. The Level meter comes up as "Peak" so you have to switch it over to "True Peak". (kinda dumb if you ask me)

I've found for hard hitting program material, it's been hard to stop intersample peaks if my brick wall limiters are anything over -1db ceiling. I don't know if -.5 will stop all peaks from happening. And apparently when it goes to mp3, it sounds better if there aren't any intersample peaks. Some mixes I've had to lower the ceiling even lower, like -2db or more even.

Course some like the sound of their converters clipping, so what the hell do I know??......


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## LaboratoryAudio (Sep 6, 2017)

Try using parallel compression. Compressor set to super fast attack and fast release. You need a compressor that can grab the initial transients like The Glue. The mix that in with the "dry" signal. You should be able to bring up the overall volume but with less transients. Hope that makes sense.


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