A friend of mine does this .I usually just use a compressor and up the attack .The ssl Duende stand alone comp is one of my favs for smoothing transients. There is something special about that compressor i find
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the loudness is measured in approximately half second intervals and the dynamic range changes over that half second. If you have a transient that goes over the trigger threshhold within that interval, then that sample is considered “loud” by streaming platforms. The more ‘such samples’ you have, the louder your song will be considered and turned down by streaming platforms. A way to avoid being squashed too much by streaming platforms is to ‘pump’ the loudness in 1-2 second intervals to avoid the half second detection over prolonged periods, hit hard and then soft (going over the threshhold and then under it). This can be done to high mids and drums.
punchiness is relative to the listener. A listener will consider a song punchy if they have some non punchy parts in the same song to compare and contrast.
Well isn’t there a lot of other methods to measure loudness? Average, short term, momentary, integrated, loudness range, true peak… What you are describing is the momentary loudness (measured over 400 milliseconds) right? This is one of the loudness measurement stages as recommended by the International Telecommunication Union (ITU-R BS.1770-4) but other recommendations exist (like EBU in Europe, ATSC in the US, etc) and they all add their own tweaks and specificities to this standard. As far as I know, Spotify and other streaming platforms don’t care about these norms. Some use LUFS (as described by ITU but not with the -24 LUKS target), some use ReplayGain (Spotify actually just switched from the latter to the former a month ago), some turn music down but not up (Youtube, Amazon…), and some don’t normalize at all (Bandcamp, SoundCloud…). As of December 2020 these are the main platforms algorithms targets (source):
| ### Platform | ### Peak | ### Loudness | ### Dynamic Range |
|---|---|---|---|
| Spotify | -1.0 dBTP | -13 to -15 LUFS | >9DR |
| Apple Music | -1.0 dBTP | -16 LUFS(±1.0 LU) | >9DR |
| Apple Podcasts | -1.0 dBTP | -16 LUFS(±1.0 LU) | >9DR |
| Amazon Music | -2.0dBTP | -9 to -13 LUFS | >9DR |
| Spotify Loud | -2.0 dBTP | -11 LUFS | >9DR |
| Youtube | -1.0 dBTP | -13 to -15 LUFS | >9DR |
| Deezer | -1.0 dBTP | -14 to -16 LUFS | >9DR |
| CD | -0.1 dBTP | > -9 LUFS | >9DR |
| Club Play | -0.1 dBTP | -6 to -9 LUFS | >8DR |
| Soundcloud | -1.0 dBTP | -8 to -13 LUFS | >9DR |
So all this leads to my question for you: why focus on the momentary loudness only? I am genuinely interested in the loudness topic so if you have reliable source material about this I’d love to see it.
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momentary lufs is a big factor in determining loudness by some streaming platforms. It is a factor we can control quite easily and it is a factor that also gets overlooked quite easily without taking a huge dip in overall loudness. Which is why my focus is more on momentary loudness. Maintaining good healthy momentary loudness levels is still very critical to what is perceived as punch. A listener wont really perceive deviations from the average loudness over a long interval ( for example a casual human listener isnt going to worry about parts louder than the parts 30 seconds ago in the song, unless ofc its obscenely loud suddenly) so DR becomes less of an issue. Our goal is to impress the the human not the machine … atleast not yet lol. I dont want to derail this thread into loudness war direction 
but it does have some common elements to the topic.
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Well this is a topic about loudness and in a world where the majority of streamed music is normalized, I feel that this is an important one to understand for anyone who releases music digitally. We don’t have to make it about loudness wars though, personally I am not at war with anyone (but myself maybe!)
In 2017 only 17% of the music that was streamed was not normalized. I don’t have the recent figures but it is obviously less by now, the trend clearly is to have everything normalized. So IMO the loudness war is not even a relevant topic now. But when you’re involved or interested in mastering, you need to have a good understanding of the normalization algorithms in order to ensure the best possible playback for the end user.
I always read that PLR and PSR are the key values that streaming platform algorithms rely on to normalize, this is why I’d like to know where you got your info about momentary loudness being a big factor. I’m not trying to contradict, just seeking to learn and understand. It is quite a big deal because in the domain of short-term loudness we are looking at 3 second periods of time so we’re no longer in the transient realm.
Here are some sources quoting PLR/PSR being key factors (and not mentioning momentary loudness):
https://www.soundonsound.com/techniques/mastering-streaming-services
https://www.producenewmedia.com/podcast-dynamics-loudness-range-vs-psr-plr/
momentary loudness is factored into the lufs calculation overall. I wrote a blip about streaming here on a fine day.
I mostly mix for streaming platform and while some info in that thread maybe old, there are still some streaming platforms that use lufs as a main factor.
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Yes, and I also found out that the “Heavy” Saturation mode is a clipper! It doesn’t seem to spell this out in the manual, but the Schepinator said it himself in his “In-Depth” video for Waves on the plugin. I don’t know if it’s hard or soft clipping, though maybe that can be controlled with the “Saturation Drive” (?) control … the big round knob that goes from 0-100. Something to play with. So while this plugin is the Swiss Army Knife of audio, it looks like it also covers your criteria for either limiting or clipping, or both.
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That’s cool to know… that Andrew S is a smart fella!
Ok then, I just finished to read the whole thread and I wish to thank you Andrew for sharing this. I’ll try this to see how it works soon. And I hope I don’t mess everything up as I’m used to…
But the initial paragraph that started this topic was
Which means that you mixed one way before then change something and now you’re happy with this thing.
Controlling peaks isn’t the main topic as I understood, or I missed something 
The manual says:
Saturation level
Adjusts the amount of harmonic distortion added to the original signal.
Range: 0 to 100%
Saturation type
• Selects between Odd and Even harmonics. The impact of odd vs. even harmonics on a signal is very
content-dependent.
• In contrast to Even and Odd, Heavy is less about adding harmonics and is more of a traditional clipper. It has
a custom response to give you a different sound than most clippers, allowing you to shape the sound in ways a
simple clipper can’t.
So it looks like it’s kind of going from soft to hard clipping as you turn the knob, but it also says that it’s not about adding harmonics, yet I always thought soft clipping was adding even order harmonics and hard clipping was adding odd order harmonics so it’s a bit confusing. Fortunately, we have our ears to tell us which one sounds good when we use it 
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I was with you all the way Andrew, until…
Clippers - OK, as long as it’s inaudible, then clipping is worth looking at. But surely if you use limiters all the way through the chain you’re committing the same crime as slapping a limiter on the 2-bus, just in a different place?
Personally I don’t think there is anything better than manual automation but if you can’t be bothered (like me ) I suggest something like ERA Vocal Leveler. I use Drum Leveler on pretty much everything that needs leveling out. Not only does it reduce the transients but it boosts the low level stuff.
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No, Andrew is discussing the finer points of leveling out inherently spiky signals such as snare, vocals etc. In other words, he is suggesting alternatives to compression.
I would like to add Torque to the list of piling up punch related plugins already. Torque works really well on isolating the energy of the punch, and pitch and can work on a variety of tracks.
Thanks for posting the text quote, my PDF manual doesn’t seem to have any of that language, or at least I can’t find it through a text search. And it looks like they haven’t updated the manual since I downloaded it (41 pages).
Yes, I got that. I would still call it gain staging though, since it presumably relates to “headroom”. As I understand it, since the analog days the definition of gain staging had to do with 1. Noise Floor (or signal-to-noise ratio), and 2. Headroom. With analog they had to work harder at these things. With digital, a lot of the noise floor problem was solved (unless you do something wrong), and the headroom was potentially significantly increased. So presumably you don’t have those headaches any more. Then the Loudness Wars happened (sorry Andrew for bringing it up 
). Everybody got creative with digital techniques and plugins to push the envelope. I understand and like the idea Andrew has described, I’m just framing it in the wider context of “mixing theory” or what have you. As long as it sounds good, and achieved your goal, there are “no rules” as they say. “If it sounds good, it is good.” As you pointed out, does shifting the load from the master buss limiting to tracks sound better or work better? Andrew seems to think so. We just have to experiment ourselves, if we want to.
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As I was watching that Scheps Omni Channel " In-Depth" video, at around 21:23 (see quote below), he describes inserting an additional plugin. The SOC will let you insert one Waves plugin in the chain. He demonstrates with the Torque plugin. So both to your point, and potentially using that technique (along with Andrew’s) in SOC, if anyone is interested in it. 
Regardless, if you use SOC then this video is extremely helpful. And if you open it in YouTube and look in the video Description, there is a helpful time-stamp index to all the topics!
.
21:23 – How to insert additional plugins into the Scheps Omni Channel
.
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Aha! I’m glad you asked this, because this is at the heart of why this method works so well… No it’s definitely not the same… and here is why…
Take a snare drum - you can clip a snare drum much deeper than you can, say a vocal, or say some cymbals, or some other other musical instrument like a piano. By clipping or limiting the individual sources, you are tailoring the type and the amount of clipping and or limiting to suit. You have control over it individually.
When you slam all your stuff into a limiter on the 2 buss, it gets limited all the same amount. Some of that will be favourable to some the of programme material, while at the same time being devastating to other sources fed into it. The only option in that case is to reduce the amount of limiting until the devastation is minimised. Taking care of it most impactful sources “upstream” minimises the what you have to do at the end, and the net result is that it can sound better.
It’s the same principle as to why we use multiple stages of compression. Each earlier stage lessens the work needed by the processors in the next stage. So if you goal is to not “hear” compression working, then condition the material going into the compressor.
There is also the fact that transients in heavily rhythmic music compound, because they tend to all happen at once. If you tame each individual transient at the source the compounding effect of them is lessened.
committing the same crime
A “crime” is good analogy that actually illustrates the point:
Who is likely to be the more successful, wealthy and uncaptured criminal? The guy who walks in the front door of the bank with a balaclava and a sawn-off shotgun demanding money; or the quiet, nerdy computer geek who comes up with a way to silently skim a few dollars each from millions of different bank accounts?
Little gains add up to a lot, and can be pretty much penalty-free.
…I dunno, Adrian, I could talk all day trying to prove to you why it works. Ultimately, that is meaningless. If you care, try it; If you don’t, ignore it and carry on as you were.
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I’m glad you explained this, based on Adrian’s postulation. It has always concerned me that with a Master buss limiter (or compressor, for that matter) you are having to use one attack and release setting to cover all the frequency range, as well as your point about the limiting being being the same amount (or at least generic). Different frequency ranges respond differently to attack and release (owing to longer or shorter wavelengths), or certainly can in many cases, so treating things more individually could potential give you more control with attack and release on a specific sound.
Certain side-chain filters have been a workaround for getting frequencies to actually be attenuated the same amount in compressors and limiters, including the Master buss. Otherwise they could be imbalanced. It seems to me you could somewhat solve or avoid that compromise with your technique? At least with the Master buss.
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But why would you use a master limiter when mixing? I personally think that if you feel the need to use anything that deals with dynamics or frequencies on the mix bus, you have done something wrong in the mix and you should find the cause and fix it at the source rather than patching it at the end of the chain.
If the mix is done right, mastering can be done with harmless limiting, compression and EQ on the whole mix. It doesn’t have to be detrimental to the sound because it can be done in light touches. You shouldn’t have to make upfront moves like fast release times, >2 dBs of boost/cut on an EQ, narrow Q etc while mastering (unless you’re salvaging a bad mix).
It’s a good point, and I think it just became a trend and a workflow in the modern era. Many point to the invention of the brickwall digital limiter in 1999 (IIRC), which changed the way mixing and mastering was done, and brought about that famous audio war which shall remain unnamed. Efforts in recent years to undo that damage have had some success, and then I guess we can ask “where do we go from here?”
Your approach seems like a very classic one, which is refreshing. It does cause me to reflect on the reasons behind things. The quest for loudness, at least commercially, isn’t that new though. The famous mastering engineer Bob Ludwig got his big break when he gambled on a brand new and expensive Neumann cutting lathe (for vinyl) that could print the audio 6dB louder than before. He took all the business from everyone else for awhile (1960’s or 1970’s I would guess).
Yes, I would say that’s the classic standard. Though things have changed over time. When cutting vinyl, they had to do some pretty drastic EQ to the low end for the vinyl standard IIRC. And cutting the disc was part of the mastering process back then. Digital changed everything (or many things), and opened up a whole new world of possibilities. Things got wild and crazy, and probably out of control just a bit. With a renewed focus on recordings with actual “dynamics”, it may get back to that standard. I would remark though, that the explosion in plugin designers and technology probably doesn’t help. They have to create a need to sell their merchandise, and they’re not required to attach warning labels for its use. 
That’s very true, the offer is incredibly extensive and often quite cheap. It’s very easy to buy a plugin because of a trend, and designers have understood and taken advantage of the fact that we’re attracted to shiny things. Scrolling through the Facebook groups about audio engineering is quite scary in this regard, there are thousands of “producers” and “mixers” who think the quality of their work depends primarily on their DAW and plugins…
I have been a victim of this shiny objects syndrome myself when I started out. Fortunately I’ve matured and now I only buy something when I need it, or when something new brings a significant improvement to something similar I already have. And the biggest mindset change I made when I started doing this for a living is thinking a lot more in terms of ergonomics and time efficiency.
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