# How to tame a huge dynamic range



## ptram (Dec 29, 2021)

Hi,

I'm working to an orchestral mockup, starting from a few violins in _ppp_, with a buildup leading to a climax in _fff_, played by a full string orchestra and timpani rolls.

Limiting the louder part is not the solution I would want to choose. I would like everything to still sound natural.

At the same time, lowering the volume of the whole piece would make the quieter parts too quiet.

How would you deal with this dynamic arc? Smoothly changing the volume while the music becomes progressively louder? I know this is done often, but would this result in the naturalness I would achieve?

Paolo


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## KEM (Dec 29, 2021)

Compression, soft clipping, limiting, etc.

You could do volume automation but I’d argue that would sound even less natural as there’s more room for error, not to mention it’d be a much bigger pain to have to do that to all of the tracks you have in the project

I know a lot of people on here have some strange fear of compression, clipping, and limiting but I can promise you they’re fine and they work perfectly well in an orchestral context


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## Trash Panda (Dec 29, 2021)

Parallel compression can do wonders for bringing up the volume of the soft parts while lessening the impact of the compressor on the loud parts.


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## Zanshin (Dec 29, 2021)

Use a reference track(s). Use tools like parallel compression suggested by Trash Panda, a good mastering limiter (like Elevate), etc, to get there.


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## ptram (Dec 30, 2021)

KEM said:


> Compression, soft clipping, limiting, etc.





Trash Panda said:


> Parallel compression can do wonders for bringing up the volume of the soft parts while lessening the impact of the compressor on the loud parts.





Zanshin said:


> Use a reference track(s). Use tools like parallel compression suggested by Trash Panda, a good mastering limiter (like Elevate), etc, to get there.


Thank you for your suggestions. Parallel compression is what I usually do to deal with pieces with a dynamic range going from pp to ff, and with short louder passages.

Another thing I usually do is to lower the background parts volume, and only let the foreground parts be really loud. This usually keeps the overall volume inside a reasonable range.

In this case the dynamic range is wider, and the extremes are not only in some peaks. It's a continual buildup. Compression is making the louder parts sound "squashed". Even parallel compression can't hide this.

Analyzing a reference track is something I’ve not yet done, and I guess it would be a more than needed thing to do.

I think I will have to post the problematic piece, maybe after having refined it a bit.

Paolo


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## MartinH. (Dec 30, 2021)

ptram said:


> Thank you for your suggestions. Parallel compression is what I usually do to deal with pieces with a dynamic range going from pp to ff, and with short louder passages.
> 
> Another thing I usually do is to lower the background parts volume, and only let the foreground parts be really loud. This usually keeps the overall volume inside a reasonable range.
> 
> ...



Have you tried upwards compression on the quiet parts? 









Expanding on Compression: 3 Overlooked Techniques for Improving Dynamic Range


In this blog, we take a look at dynamic processes often overlooked in mixing and production, with tips on how to employ them using compressor plug-ins.




www.izotope.com


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## ptram (Dec 30, 2021)

MartinH. said:


> Have you tried upwards compression on the quiet parts?


Not tried, yet. By reading the description, it sounds like it is more effective on percussive sounds than in sustaining ones. However, as the author states, it shouldn't be too different from parallel compression.

Something I should try is increasing the amount of compressed signal in the parallel mix. If the compressor is transparent enough, it should increase the quieter parts without damaging anything.

Paolo


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## chibear (Dec 30, 2021)

From a retired orchestra musician and DAW hobbyist. Sorry for the dense wordiness.

I have been struggling with this ever since I began working with DAWs. Up until now I have done my best for my SoundCloud and YouTube pages, but have never been really happy with any of it.

Pre-digital, back in the 1970’s, Dynamic Range Expanders by DBX and others were all the rage with audiophiles (for vinyl, of course). They expanded the dynamic range downwards so instructions suggested you drop your needle in the loudest section and set your volume accordingly. The multi-band ones worked quite well.

Digital recording and CDs were supposed to fix all that, but instead we got the loudness wars so that the bottom half of the available dynamic range is never used. My older audiophile friends have tried connecting their old DBX units to their digital sources with only limited success. Most are again concentrating on vinyl.

What I have been doing with these friends is providing them with a virtually unprocessed version of my tunes (basically only EQ, a little saturation, virtual soundstage, and reverb) with a timing for the loudest part of the track so they can set the volume. These are quite popular with my audiophile friends as, along with increased dynamic range, there is also enhanced spaciousness, sound stage, and tone colour changes.

I’ve used up my free time on SoundCloud, so, as I will now have infinite uploads (and pay), will begin offering, where applicable, 2 versions of future tunes, a “commercial” processed version and an “audiophile” unprocessed version with instructions for those interested (maybe different names for the versions). I’ll be interested in the results. Depending on your audience, you may wish to consider the same.


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## ptram (Dec 30, 2021)

MartinH. said:


> Have you tried upwards compression on the quiet parts?


Now, I have tried it. On this particular type of music it is doing something strange to the attack of the notes. At least, I've not found a way to avoid it.

Upward Compression Example

By comparing my mockup with some recordings of the same piece with various orchestras (CD and YT), I see the dynamic range is in any case very wide. But not as in my case. So, I will have to try something else.

At the moment, I've tried to balance the input and output levels in the compressor and limiter, and have the best mix I can between the dry and wet signal, so that it doesn't sound overcompressed, but it also doesn't disappear in the silence.

Bartók, Music for strings, percussion and celesta (VSL VI) - Mockup 04

Bartók, Music for strings, percussion and celesta (VSL VI) - Mockup 03

Bartók, Music for strings, percussion and celesta (VSL VI) - Mockup 02

Bartók, Music for strings, percussion and celesta (VSL VI) - Mockup 01

Paolo


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## KEM (Dec 30, 2021)

Is it a gainstaging issue then?


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## Buz (Dec 30, 2021)

With the caveat I don't know what I'm talking about, there seems to be a common sense solution. 
-Decide on an acceptable dynamic range and therefore determine how loud _ppp _must be.
-Turn up _ppp_ to fit these bounds and smoothly taper the fader through _pp, p_ to the point where the original loudness is already above the boosted level.

Now the desired dynamic range is achieved, the quietest sections remain the quietest, and everything loud is untouched.


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## Philip Vasta (Dec 30, 2021)

When I use upward compression, I tend to set the attack and release times to be pretty slow so that it’s just bringing up the body of the sound and not sounding like an effect. Maybe that could help?


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## onnomusic (Dec 30, 2021)

I'd say volume automation. you could also have a VERY slow compressor on your master chain, that simply gently start compressing when the louder bits come in.


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## Ivan M. (Dec 30, 2021)

Manual volume automation. It’s relatively quick, doesn’t have to be detailed at all, just bring up the quiet sections. I had to do it before on a piano and it worked great, saved both the transients and the quiet parts.

Edit: compressors will destroy the transients, details and that natural feel.


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## Nick Batzdorf (Dec 30, 2021)

ptram said:


> Now, I have tried it. On this particular type of music it is doing something strange to the attack of the notes. At least, I've not found a way to avoid it.



I haven't listened to your examples, but if it's doing anything audible to the attacks, dollars to donuts you just need a slower attack time and possibly ratio.

And I see that Philip Vasta already said that.

Also, some compressors (or compressor plug-ins, or settings within plug-ins) are optimized more for transparent compression like you want here.



Ivan M. said:


> Edit: compressors will destroy the transients, details and that natural feel.


They can, but they certainly don't have to! Most classical recordings use some compression, and generally it's not obvious to people who aren't engineers.


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## Nick Batzdorf (Dec 30, 2021)

Also also, given that a compressor is a very fast, automated volume control... volume automation is simply slow compression. 

Having said that, it's quite possible that raising the level is the appropriate policy response to this particular issue.


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## Trash Panda (Dec 30, 2021)

Ivan M. said:


> Edit: compressors will destroy the transients, details and that natural feel.


Not when used properly.

Remember: Only Siths deal in absolutes.


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## Ivan M. (Dec 30, 2021)

Nick Batzdorf said:


> Also also, given that a compressor is a very fast, automated volume control... volume automation is simply slow compression.


Compressor is triggered by transients/peaks. On the contrary, volume automation of the quiet parts avoids touching the transients completely. 

To do an equivalent effect with a compressor, my guess is slow attack, slow release and a look ahead plugin. But the difference is it still operates on loud parts (instead of quiet ones), and might mess the transients. 

The sound however might be nice and tamed transients pleasing. A matter of taste, I guess


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## Ivan M. (Dec 30, 2021)

Ivan M. said:


> equivalent effect with a compressor, my guess is slow attack, slow release and a look ahead plugin


<Ignore this post>
Actually, a slow attack means the waveform changes (slowly, but still changes). So it would change the loud parts. To mimic volume automation (which doesn't touch the loud parts), the effect would have to be instant, meaning instant volume reduction = minimum attack, with look-ahead, so that the waveform of the loud parts doesn't change over time. It should only change only when transitioning the dynamics, and not loud transients. But that minimum attack always gives artifacts, unless it's some perfectly transparent look-ahead compressor. Maybe I'm wrong...

edit: yeah, I was wrong, instant attack would create discontinuities, so a slow attack with look-ahead


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## Ivan M. (Dec 30, 2021)

Trash Panda said:


> Not when used properly.
> 
> Remember: Only Siths deal in absolutes.


Volume automation has the high ground here :D


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## ptram (Dec 30, 2021)

Nick Batzdorf said:


> Most classical recordings use some compression, and generally it's not obvious to people who aren't engineers.


Maybe compressors in classical music are just used for quick transients? I feel like my case is not one of transients, but of volume automation, as already suggested.

Now, I heard that volume automation is a capital sin, in purist classical recording, exactly as compression. One of those things that have to be done without saying it loud.

Paolo


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## Nick Batzdorf (Dec 30, 2021)

ptram said:


> Maybe compressors in classical music are just used for quick transients? I feel like my case is not one of transients, but of volume automation, as already suggested.
> 
> Now, I heard that volume automation is a capital sin, in purist classical recording, exactly as compression. One of those things that have to be done without saying it loud.
> 
> Paolo



Compressors in classical music are used to narrow the dynamic range so it fits in a recording. That's certainly less important now than in the days of vinyl, but still - even acoustic recordings are different from live performances.

(They could be used to limit quick transients too, of course.)


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## ptram (Dec 30, 2021)

KEM said:


> Is it a gainstaging issue then?


I don't think so. The chain is very simple (samples —> compressor —> limiter, all of the most transparent type).

But maybe in a wider sense it is:


Buz said:


> -Decide on an acceptable dynamic range and therefore determine how loud _ppp _must be.


Since this is a fake-studio recording, and not a fake-live recording, I guess limiting the playing dynamics could be acceptable. I remember some recordings led by Ormandy, where I noticed how pianissimos were more like a piano or a mezzo piano.

Paolo


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## Nick Batzdorf (Dec 30, 2021)

Ivan M. said:


> Compressor is triggered by transients/peaks. On the contrary, volume automation of the quiet parts avoids touching the transients completely


I mean, yes, the compressor does all kinds of things you can't do manually with the volume control.

But ultimately it just lowers the signal above the threshold (or raises it above the threshold if it's bottom-up compression). So how it reacts depends on how you have it set!

I think what you're saying is that the attack determines how slowly the compression "fades in," but there are parameters to make it all but inaudible - attack, knee, etc.


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## Scamper (Dec 30, 2021)

I know the issue. Wouldn't want to change the sound with a compressor and I'm too lazy to do manual volume automation for everything.

I think the most comfortable way is to setup your CC1 in a way, that it automatically scales CC11 when you use it - to either expand dynamic range or reduce it, when CC11 scales inverted to CC1. How easy this is to do probably depends on your DAW.

*Method 1 - Cubase MIDI Transformer*

Setup a MIDI Transformer, that takes the CC1 value and automatically adjusts CC11 using it. Takes a bit of math to get the scaling right. Here, I'm showing dynamic reduction: while CC1 moves from 0-127, CC11 moves from 127-42, and dynamic expansion: CC1 moves from 0-127 and CC11 from 64-127.
And if you have set it up once, you can copy the preset to other tracks easily.

View attachment Cubase Transformer.mp4


And looking at @Saxer's post, there is a plugin in Logic, that does the same.




__





To BBC or not to BBC


Thanks, great input everybody. I think my main concern is the dynamics. I would want to use it in pop/rock music. IMO, there are more appropriate libraries for that type of music. BBCSO is really more for classical stuff. JMO.




vi-control.net





*Method 2 - Kontakt MIDI Automation*

Within Kontakt, you can also scale CC11 automatically with CC1 and easily set the range. Problem is then, you have to do this for every single instance and instrument within Kontakt.

View attachment Kontakt Method.mp4


I think both work well enough and afterwards, you don't have to think about it anymore. No additional work.


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## re-peat (Dec 30, 2021)

You’ve gotten yourself into trouble right from the start by having the strings waaaaay too loud, Paolo. Which means that if you wanna do the crescendo, you’re hitting the ceiling waaaay too soon and there’s no headroom anymore for the cymbal crash (which sounds like a very weak ride cymbal hit in your version) let alone for the all-important timpani crescendo and climactic hit (which is also completely drowned out in your version).

Another problem that you have, is that with such a high level for the strings, there is no connection between their dynamic and their timbre. Half the time, the two contradict one another.

I don’t really understand why you’re so reluctant to start really pianissimo with the strings. If you start ppp (or whatever the dynamic marking is), everything that follows will fall more or less into place as well. It is, after all, intended to start very-very-very soft and mysterious. Ignore that, and you ignore an essential element of the composition.

Unrelated to the above but just as worth considering, I find, is the tempo. On which version is this modeled, if I may ask? Cause I find it excrutiatingly slow. All the versions I know are much faster. My version of choice — Mackerras with the Scottish Symphony, released on Linn Records — is more than two minutes shorter than yours.
I mention this because, if you take the tempo so slow and neglect to pay the utmost attention to the phrasing of every single note (and every transition from one note to the next), you end up with these long static stretches of sampled strings notes (many of which sound disconnected even when playing legato) which, to my ears, are totally incapable of communicating the music.

The image below compares the audio displays of your version, top, and the Mackerras version, bottom. Note not only the big tempo difference, but above all the huge difference in dynamics. In the Mackerras version, you can actually see very clearly where the cymbal and timpani hits occur.







(Even more unrelated, but perhaps you might want to take a look at it as well: your soft celli have no texture or definition whatsoever. They cloud, rather that participate in the counterpoint.)

If I were you, I’d mock this up as best as possible — preserving the natural dynamics of the piece and the instruments as faithfully as you can — without being in any way worried about the dynamic problem that you’ll have to deal with when you will be mixing the piece. And when that time comes, simply use a limiter to condense the dynamic range to a comfortable listening range, but never so much that you destroy the character of the piece. Make sure to use a very good limiter however, cause this is not a job for a mediocre tool.
And if you feel you have to use compression — I’m not so sure this piece needs it though (but that depends on your choice of limiter as well) —, use, as was already suggested in an earlier post, *very* slow settings for attack and release. Or, better still, use an opto or a vari-mu type of compressor.

_


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## ptram (Dec 30, 2021)

re-peat said:


> You’ve gotten yourself into trouble right from the start by having the strings waaaaay too loud, Paolo.


That's true. And there is no room for the percussion. I was trying to keep them up.



re-peat said:


> Unrelated to the above but just as worth considering, I find, is the tempo. On which version is this modeled, if I may ask?


Not modeled, but inspired to Bernstein's. His tempo varies much, but is always slower than the one written in the score. In this case, like in others, I find that this lack of literality lets Bernstein discover hidden things in the score. In Bartok, the theme is a modernized version of Bach. In Bernstein, it is a lament.

But yes, then the mock-up should have all the notes correctly sculpted (something that they are very far from being, at the moment).

Thank you for the precious hints! I'll see how I can implement them.

Paolo


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## fakemaxwell (Dec 30, 2021)

Taming dynamic range is best done with both compressors and volume automation, together. You can even go compressor->volume automation-> compressor, thickening things up as it gets louder or whatever you'd like.

But before any of that, for MIDI (and real instruments) it's better to start at the source. Without any extra effects the performance should be as close as you can get to what you're looking for, otherwise it's just extra work.


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## ptram (Dec 30, 2021)

In the meantime, I've tried to implement some of the hints given in this thread. Faster tempi, clearer and louder celli, more balanced contrabasses, louder percussion, some retouching to phrasing, filtered out sub-basses. The overall volume was kept as low as to allow the limiter to work only in the most extreme situations.

http://www.studio-magazine.com/music/musichealtri/bartok/Bartok-Music_for_Strings_Percussion_Celesta-VI-2021-12-30b.mp3 (Bartók, Music for strings, percussion and celesta (VSL VI) -)2nd try

Making a polished mockup will be a long work. This is an incredibly refined piece, pretending an incredibly refined performance. Any other hints on how to make St. Béla be less furious with me will be greatly appreciated.

And the volume riding part is still out…

Paolo


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## NoamL (Dec 30, 2021)

About Compression and NY Compression - they are two different techniques.

When you talk about compressing the music without affecting the loudest parts, that is clearly a case for NY Compression. However the way NY Compression works is kind of counter intuitive, people are also talking about a *slow attack time* and that is exactly wrong (as far as I know) but it is not easy to explain why.

I am not an audio engineer but this is how it was explained to me by a mentor.

*Any errors are mine & I welcome corrections so we all grow in understanding!!*

I'm gonna go @JohnG on this and write in chapters to help the clarity. 

*Part 1 - Making Music Louder*

Let's say this colored bar represents an orchestral recording. The bar represents the total dynamic range and I broke it into 4 chunks for illustration. When the music is very quiet it hits at -30 on the meters and is "living" at the bottom of the blue bar. The climax of the music (trumpets, timpani, piatti) hits at -6 on the meters, fills up all four bars, and peaks at the top of the orange bar. Like one of those test your strength carnival games.






There is 6 dB of unused headroom "above" the loudest part of the music. Adding 6dB of gain to your recording removes the headroom and transparently makes everything louder - awesome! But you can't add any more gain without clipping.






*Part 2- Normal Compression*

To add more volume, we need to create headroom.

Compressors can do this by setting a threshold and then, every time the music peaks above it, those peaks are compressed down.

The *lower the threshold*, the more of the overall music you are attacking. When the threshold is -6 the compressor is only squeezing the orange part of the music. When it's -12, it's squeezing the orange _and_ yellow.

The *higher the ratio*, the more you are squashing the music. With a 2:1 ratio, when the music would climb 6 dB above the threshold, the compressor only lets it climb 3 dB.

In the diagram below I illustrated various thresholds and ratios.






When you turn up the ratio *above 40:1* the compressor is starting to become a limiter. No matter how much added volume the music tries to add above the threshold, the compressor/limiter won't let anything through.

Compression does not make music louder. It only creates headroom. This headroom can be taken away as in the very first example, by adding gain until the music peaks at 0 and no further added gain is possible.

With the maximum possible makeup gain included for each recording, the result would look like this:






This is the view that shows how compressors make music "louder." At least, the top volume of the music is the same, but the quietest parts (blue and green) are now louder and easier to hear.

*Part 3 - Problems Of Compression*

Going back to the view of the music before makeup gain is added, some problems with compression become clear.






There are two things to notice. First, a compressor is not a fully transparent effect. It squeezes the music and changes the relationship between dynamics. This is a necessary compromise since it wasn't possible to make the music louder by just adding volume.

Perhaps even more importantly, only the orange and yellow regions are "attacked" by the compressor. You can see the green and blue regions remain unchanged in every case, because they are below the threshold.

Attacking the music "from the top" has two big problems. The orange and yellow represent the most exciting, climactic and dramatic part of the music so it really hurts to remove all the life from it. Additionally, this part of the music is very rich in transients, such as trumpet staccatos and crash cymbals. Our ears are well evolved to hear transients and can tell when they are decaying unnaturally thanks to this compressor squishing effect (or, for the same reason, large amounts of limiting).

It would be better to *attack the music "from the bottom"* and squeeze the blue and green bars. Almost by definition, anything music that consistently lives in the blue region (such as a quiet flute part with accompanying string tremolos) is going to lack strong and obvious transients. Therefore, compressing this part of the music is more "invisible" and less annoying.

The ideal end result would look something like this. But how can this be achieved?


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## NoamL (Dec 30, 2021)

*Part 4 - New York Compression*

The answer is to combine two signals. The first is the original recording, and the second is a copy which has been highly compressed.

Both signals need to have the *exact same latency to avoid phasing*. So the "dry" signal could have an instance of the same compressor on it, but with the threshold set to 0 so it's never doing anything.

The compressed signal has the threshold at -24 and a ratio of 10:1 or even more. The end result looks like this:






When is the compressor working? _All the time_, because _all of the music_ is louder than the threshold.

What does the compressed signal sound like? _It sounds super squashed and lifeless, _because the quietest and loudest parts of the music are almost the exact same volume.

How loud is the compressed signal? _It is very quiet_, exactly as quiet as the quietest parts of the music.

And the most interesting question:

*Part 5 - Sum Of Signals

What happens when the two signals are added back together?*

It may be easiest to reason about what happens when the music is very quiet and very loud.

When the music is very quiet, the signals have equal volume (look at the two blue bars). When you add two signals of equal volume, you get a 6dB boost.

When the music is very loud, the dry signal is much louder than the Compressed signal. (Look at the two orange bars). The result from adding the two together is only a tiny bump in volume.

This is shown by the two arrows.






In between, the amount of added gain will vary. The *louder* the music gets, the *LESS* gain is added.






If this does not make sense, think of someone *coughing at a concert.* When the music is very quiet, the cough is a very noticeable addition to the sound. But when the music is very loud, that same-volume cough is not making a noticeable difference.

The end result looks good, with the orange region hardly squashed at all, but the blue and green compressed quite a bit. The total sum is going to clip slightly so it's best to do this process on a signal that has headroom (such as the original recording) and then just apply slightly _less_ than 6dB makeup gain as needed.

*Part 6- Why The Right Settings Are All Weird And Screwy*

The settings to make a NY Compressor work are very strange but hopefully they are more clear now that you have this illustration.

In *normal* compression, you add makeup gain to the make the overall mix loud again.

But in NY Compression there should be no makeup gain. The compressed signal will remain very quiet.

In *normal* compression, when working on the music master you often set the ratio to something "gentle" like 1.5:1 to avoid being too noticeable, so that the compressor does just a _little_ gain reduction.

But in NY Compression the ratio should be HIGH, something like 10:1 or above, so that the compressed signal has very _consistent_ volume regardless of the musical dynamic - which means it is also doing _A TON_ of gain reduction. The only thing you want to watch out for is that the compressor does not cause ugly artifacts or errors, but apart from that, by all means squash it to hell.

In *normal* compression, you set the threshold to attack only the loudest part of the music.

But in NY Compression you set the threshold to attack all the music. The threshold must be _*as low as the quietest parts of the music*, _so that the compressor *works all the time*. If the threshold is too high, the material below won't be compressed (remember the whole point of NY Compression is that the quietest material is the "best" stuff to compress). If the threshold is too low, you will be compressing the sound of the empty room before and after the orchestra plays.

In *normal *compression you can set an "*attack time*" in milliseconds. If the compressor detects a peak, it will wait that many milliseconds before starting to squeeze the music. This is useful to let some transients through, especially on percussion. The idea is to get a compromise between a compressed mix, and unaffected transients that our ears are sensitive to.

But in NY Compression the ideal attack time is *ZERO MILLISECONDS.* The best compressor for this job is something with a super fast attack (<2 ms) and you can even put a safety limiter right after the compressor (before the signal is summed with the dry recording), set to for instance -20dB threshold in this case and no makeup gain. Why should the attack be as fast as possible? The reason is, if you let any transients through your NY Compressor without attacking them, they will sum with the transients of the dry signal and make your transients twice as loud, making your mix even more dynamically out of control. The right mindset is that _you already have your unaffected transients_, in the dry signal. The NY compressed signal needs to be SQUASHED and consistent.

In *normal *compression the settings are usually valiantly battling to stay "invisible" (because the compressor is attacking the most dynamic and transient rich parts of the music) and that's why people talk so much about *"gentle compression."* There is some idea that because NY Compression is so radical, and the compressed signal sounds so ugly, that you should *"mix it gently" *with the dry signal.

But in NY Compression the thing that makes it inherently "gentle" is that the compressed signal is QUIET. You can mix the dry and compressed signal at ratios of *at minimum 1:1 *and you can even experiment with 2:1 or 3:1 ratios - in favor of the _compressed signal_. I have seen some youtube videos where people add makeup gain to make the compressed signal loud, and then they "mix just a little of it" into the dry signal. This makes little sense, you're just adding 12 and then subtracting 12 again later.

In *normal *compression you have to carefully listen to the loudest & most transient parts of your mix to make sure you're not being too destructive and warping the feeling of the music.

But in NY Compression the parts you want to attend to carefully are the _quietest_ sections of the music, because those are the parts being compressed, and especially the transitions between quiet and loud sections.

*Part 7 - "I don't compress my orchestra" - some fellow on the forums named "rctec"*

After all of this, I still think the best compression is, no compression.

On drumsets, sure, and on cinematic percussion, sure, but I am so wary of compressing strings, winds and brass.

And on the full master mix like I've been "demoing" this whole time? Hell no!

As composers we should understand the concepts, but it's just not my job to do this, and probably make a damned hash of it far more than any mastering engineer would do.

The dynamic range of the orchestra is what it is, and I feel it's best to deliver natural, and great sounding music. If that creates problems down the road for our colleagues in film, they are the ones who have the best tools and best expertise.

You can get a very hyped sound with certain kinds of processing but I always wonder about the longevity of those sounds. The #1 thing that makes me skeptical of any kind of compression is the "taste to what you're adding to the cooking pot" maxim. If it doesn't taste good by itself, why would you add it to the stew even if it's "just a little"? The whole idea of NY Compression seems to violate this maxim. Thus, while I've heard other people get great results with it, I prefer to just deliver fully dynamic and uncompromised audio.


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## ptram (Dec 31, 2021)

Thank you Noam. Keep in mind that what is sometimes called "New York Compression" should be what we called, somewhere in this thread, "Parallel Compression". I don’t know if these are different nuances of the same thing.


NoamL said:


> The dynamic range of the orchestra is what it is, and I feel it's best to deliver natural, and great sounding music. If that creates problems down the road for our colleagues in film, they are the ones who have the best tools and best expertise.


When knowing that your music will be further edited, I guess there is no alternative to leaving it at full dynamic range. By doing your own processing, you damage the source material on which the editor will have to work.

But what about when you have to deliver the finished product? This can be your CD, a streaming album, a demo piece to be posted in a forum? Can we just ignore that the target listening media have a much narrower dynamic range, and by leaving our music at full range would either make the softer parts unheard, and the louder annoyingly loud?

Paolo


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## Snarf (Dec 31, 2021)

NoamL said:


> I am not an audio engineer but this is how it was explained to me by a mentor.
> 
> *Any errors are mine & I welcome corrections so we all grow in understanding!!*
> 
> I'm gonna go @JohnG on this and write in chapters to help the clarity.


Thank you for this write-up. In-depth posts like this are why I stay on this forum


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## Vik (Dec 31, 2021)

One thing to keep in mind when automating Volume or CC11 is that both these are altering the overall loudness of the players in a way which is very different from how the players alters loudness. If you alter CC7 or CC11 (or use a compressor), the color of the sound won't change – it will be quite...dense, in lack of a better word, if the samples are recorded playing mf, f or ff, but played back with a lower db-level. The end results will of course be very different from when player would play or sings softly, as illustrated eg. here. 

To deal with this in the most musical way, it's IMO best to avoid using CC7 (volume) or CC11 ('expression') and instead lower the volume with CC1 (dynamics) whenever relevant and possible. If not possible, but something with more dyn. layers next time. 

We'll probably be bombarded with orchestral libraries with 5 or more dynamic layers in the next few years, which makes dealing with natural sounding dynamics editing much easier, again, because the stuff that needs to be heard with a low volume also has the color of something that's played softly.


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## muk (Dec 31, 2021)

Vik said:


> To deal with this in the most musical way, it's IMO best to avoid using CC7 (volume) or CC11 ('expression') and instead lower the volume with CC1 (dynamics) whenever relevant and possible. If not possible, but something with more dyn. layers next time.
> 
> We'll probably be bombarded with orchestral libraries with 5 or more dynamic layers in the next few years, which makes dealing with natural sounding dynamics editing much easier, again, because the stuff that needs to be heard with a low volume also has the color of something that's played softly.


In my opinion you are conflating dynamic layers with dynamic range here. The issue in ptram's mockup is not missing dynamic layers, but an excessive dynamic range. If a strings library has a dynamic range from 40db (quietest signal) to 70db (loudest signal), its dynamic range is too limited. Whether it has 4 dynamic layers within its dynamic range of 30db or 8 is another question. The problem is that 30db are not enough dynamic range.

For ptram it's the opposite problem here. The strings have too much dynamic range. The solution is to limit their dynamic range. One way to do that is using compression (upward, downward, or parallel). Another solution is to limit the dynamic range of the strings within the library - if the library offers that option. The VSL player has a dynamic range slider, for instance. In that case, lower the dynamic range and the problem will be solved.

If the library doesn't have that option, using cc11 or cc7 can help reducing the dynamic range too. And it's not wrong to use that. Some libraries don't offer a to niente option, for instance. In these cases, using cc7 to fade to complete silence when cc1 is already at a value of 1 is quite helpful.


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## b_elliott (Dec 31, 2021)

re-peat said:


> The image below compares the audio displays of your version, top, and the Mackerras version, bottom. Note not only the big tempo difference, but above all the huge difference in dynamics. In the Mackerras version, you can actually see very clearly where the cymbal and timpani hits occur.


Bear with me as I am a newb but fascinated with this thread.

In addition to what re-peat points out I see two other things from this image:

1. The flat top of your peak dynamics = clipping for a stretch of time.

2. There appear to be only two spikes in the Mackarras recording (right channel); only one spike on left channel (I may have the L + R mixed up.) Not enough listens for me to say why that is.

When I next listened to version two from the OP, I hear balance issues in the basses (annoyingly too loud). Massive difference between their level on the Mackarras vs OP's mockup.

I have since created wav files of the two aforementioned songs as well as another reference which has similar properties (length, dynamic range, instrumentation, brilliant conductor). (For personal further study.)



My plan is to throw all three versions onto the ADPTR plugin (which I believe is what re-peat used above) then use its umpteen sound analysis tools to see if it can help me better spot the differences-problem area.

My suggestion would be for the OP to do like-wise. Perhaps then using Izotope's Visual Mixer one can nail the balances especially in low strings and match the crescendos per other means.
####################################

EDIT: after looking at OP's mix 2 and Mackerras on ADPTR:

1. No sub range for mix 2, seems to have a steep cut off around 40hz.
2. Mackerras lively sub bass vs none in OP vers 2.
3. In octave display it shows generally higher db ranges in OP from 63 hz on up compared to Mackerras.
4. A 12.7 dB gain required to match Mackerras peak to OP peak (loudest section).
5. LUFS-I compare: OP at -17.3 vs Mackerras at -32.2.
Re-peats of the world could better interpret that. Compression? 

FWIW: the two spikes shown in re-peat's image of Mackerras recording are first a cymbal crash; the second is a tymp hit.

Also noticed no celeste in version 2. Mackerras enters at 6 min 15 outro. Possible muted track in vers 2?

Wish I could say "twist this here knob, push that there setting, and yer done" but that's way over my current skill set.


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## vitocorleone123 (Dec 31, 2021)

onnomusic said:


> I'd say volume automation. you could also have a VERY slow compressor on your master chain, that simply gently start compressing when the louder bits come in.


This is what I was going to write: ride the faders or draw it in manually (but only a db or so). Also try multiple compressors that target different things (attack, release) but only compress .25db each or so.


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## Vik (Dec 31, 2021)

muk said:


> The VSL player has a dynamic range slider, for instance. In that case, lower the dynamic range and the problem will be solved.
> 
> If the library doesn't have that option, using cc11 or cc7 can help reducing the dynamic range too.


Sure, using CC7/11 is useful when one cannot do something with the dynamic range – I'll see if I can edit my post a little to make it clear that this wasn't a comment to this piece, but a side comment to something which is relevant when we work with VIs and want to 'fix it in the mix' eg. because the dynamic range because is too large. The reason I mentioned the stuff about 'when relevant/possible' was that very often, altering volume/gain/automation is the only choice we have when working with VI libraries.

IMO it's important to distinguish between whether we're fixing the mix of real instruments or virtual instruments. Ptram explained that the mockup started few violins in _ppp_, with a buildup leading to a climax in _fff_, and in almost all fix/mix situation it's relevant if the mockup is based on actual recordings of violins playing ppp or if it is a three layer library which neither contains ppp or fff recordings, because things will start to sound unnatural if the PPP violins weren't playing ppp, because then we're going to hear violins that played louder but which are manipulated only in terms of gain/volume/compressors: this is going to make the mix sound unnatural, especially if the fff stuff wasn't recorded fff but played more quitely but helped with some kind of volume control.

And even with libraries that offer a dynamic range slider, which a few libraries do, one needs to – if the goal is an end result sounding as natural as possible – relate to the fact that one can use use a dynamic range slider also with libraries with only three dyn. layers. This won't solve the special difficulties related to mixing/mastering mockups (as oppose to orchestral recordings).

This topic is very interesting – and also huge, and there are many threads on internet discussing both mixing and mastering orchestral music, but we're IMO doing a disservice to ourselves if we mix up tips about mixing or mastering actual orcehstras vs. mixing or mastering mockups. We also need to distinguish between mixing and mastering, and know the difference between compressing and limiting. Anyone who have watched classical concerts on TV may also pay a little attention to the fact that in the audience, that age group which is known to have far from perfect hearing is often the largest group. Even with a perfect mix, the quiet stuff will end up too quiet or totally lacking high frequencies before the signals reach their/our brains.

I'll go through all the suggestions in this thread, and especially Noahs detailed posts, but re. my statement about not ignoring that our mockups may sound unnatural already before we start to mix or master them is a statement I would stand by even if it would be the last words I said in this life.  Why? Because we can't expect automation/gain change to repair unnaturalness that is caused by the fact that we are dealing with main ingredients that are... well, fake. 

Example: It doesn't help to think that we deal with ppp or fff recordings if we don't, and the end result will suffer from that. Some of this can be dealt with by automating EQ, because this can be used to cover up for fake ppp 'recordings' sounding too bright, or fake ppp 'recordings' sounding too mellow, but the easiest trick in the book is IMOP to start with automating dynamics instead of CC1/CC7 _when relevant and possible._ 






How to mix orchestral music - what plugins?


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Best UAD plugins for mixing and mastering orchestral instruments?


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## NoamL (Dec 31, 2021)

ptram said:


> But what about when you have to deliver the finished product?


I would pitch the client on hiring a mastering engineer, score mixer, etc whatever is appropriate to do the final polish on the project.



muk said:


> Another solution is to limit the dynamic range of the strings within the library - if the library offers that option. The VSL player has a dynamic range slider, for instance. In that case, lower the dynamic range and the problem will be solved.



If the music has to mess with the dynamic range, I feel it should be across the entire mix or at least the entire orchestral submix so that it sounds consistent, the way a mixer would approach a live recording. Not instrument by instrument. Some libraries from 8dio let you link CC1 and volume, and VSL + CineSamples libraries have a dynamic range slider. In every case it's an invitation to create a complete mess. It's hard enough to balance an orchestral template when the dynamic range is _accurately_ represented by the VIs! I agree with you about using CC11 in some cases (string divisis, fading out pianissimo notes).


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## ptram (Dec 31, 2021)

Bill, great suggestions, thank you!



b_elliott said:


> 1. The flat top of your peak dynamics = clipping for a stretch of time.


Yes. An extended zone of clipping. I'm reducing it with subsequent tries, but still no way to avoid them without overcompressing.



b_elliott said:


> When I next listened to version two from the OP, I hear balance issues in the basses (annoyingly too loud). Massive difference between their level on the Mackarras vs OP's mockup.


Unfortunately, I don't have the Mackerras version, so I'm a bit playing blind. At the moment, I have the Fricsay version loaded in Metric AB for a comparison. So, I've for sure a very different point of reference.

Shockingly, my version has _less_ bass than the Fricsay's! 




b_elliott said:


> 1. No sub range for mix 2, seems to have a steep cut off around 40hz.


I confirm, I put a high-pass filter in the mix. Should this be avoided? I had fear of those powerful contrabasses!



b_elliott said:


> […]
> 5. LUFS-I compare: OP at -17.3 vs Mackerras at -32.2.
> Re-peats of the world could better interpret that. Compression?


Can't check it, but maybe Mackerras is also keeping the dynamics low? Compared with Fricsay, it seems there is more coherence between my mockup #2 and his performance.



b_elliott said:


> Also noticed no celeste in version 2. Mackerras enters at 6 min 15 outro. Possible muted track in vers 2?


Just checked the MP3 in my local drive, and the celesta is there. Maybe too soft?



b_elliott said:


> Wish I could say "twist this here knob, push that there setting, and yer done" but that's way over my current skill set.


So, I'm left to do all the work myself?!? 

Thank you very much! You guys are of incredible help!

Paolo


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## b_elliott (Dec 31, 2021)

BTW not suggesting anything illegal: one can listen to Mackerass on YouTube. 

Audacity (free) lets one record what is playing on the laptop. 

Somehow I got a .wav Mackerras to explore inside ADPTR (I paid $14 bucks this Xmas using PA coupons for that plugin; a 7 day demo is also available.)

Just sayin'. 

re: Mackerras' celeste is treated as a solo inst with string accompaniment between 6:14 and 6:34. Audio shows a bump in mids and highs to feature the celeste. Somehow I lose the celeste in your version.


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## ptram (Dec 31, 2021)

Reading in the manual of Metric AB: the ideal dynamic range for a mixdown is between 12-14 dB. And there is also a setting of 15 dB.

No idea if this is genre-agnostic.

Paolo


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## b_elliott (Jan 1, 2022)

ptram said:


> Reading in the manual of Metric AB: the ideal dynamic range for a mixdown is between 12-14 dB. And there is also a setting of 15 dB.
> 
> No idea if this is genre-agnostic.
> 
> Paolo


Yes I saw that. I also noticed a loudness preset in Metric A/B = Dynamic Pre-master -16 LUFS. 

My small experience with LUFS: Ayaic's Mix Monolith has those target ranges as well in its Mastering presets. After some testing on my music (non-classical), my ears preferred -18 LUFS over other settings. 

For fun this morning, I put your version 2 mix through two different mastering plugins on my master channel:

1. bx_Masterdesk Classic's preset 12 CD Remastering Start (no tweaks applied)

2. Ayaic Mix Monolith preset Master/MainFader(-18 LUFS)

I then ran the Ayaic MM in "learn mode" playing through the entire vers 2 track. With nothing else on the master I got the following results:

MM applied a -2.96 gain to vers 2 mix to achieve -18Lufs.
The clipping at 5min still registered in the red.

I then lowered the MM Fine Trim (-3dB) which still gave momentary and short term LUFS in the red but achieved -19LUFS integrated.

So without fixing the clipping, this showed the two plugins do the AI-work to hit target loudness levels.

Lessons I learned from all this:
1. The vers2 clipping in 5min range needs fixing. I'd place Stealth Limiter (set to -0.1 max) to handle the problem tracks. Likely a dozen other/better solutions.
2. Get the celeste volume at outro increased to match Mackerras' levels. Also liking how prominent Mackerras has his cymbals' and timpani strikes = another thing to adjust during mix stage.
3. Let Mix Monolith master to my desired LUFS.

As stated earlier I am new to this, but thought it worthwhile from a beginner's perspective since there is so much to digest tech-wise. 

Hope it helps.


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## b_elliott (Jan 2, 2022)

After some digging on this Bartok work and on mixing, I uncovered new-to-me data which is relevant to the OP's mix/mastering process (yes, broader than DR). 

This video highlights Bartók:_ Music for Strings, Percussion and Celesta_ recordings. The whole thing is worth a watch; however, a few take-aways relevant for the mixer, mastering eng.:

- Bartok (similar to Varese) wrote detailed instructions on how the 2 orchestras were to be set-up with placement of the basses in the center back. 
- several prominent conductors made multiple recording attempts to get this work right
- two high-quality reference recordings: 
1) Neville Mariner's Decca recording
2) Zoltan Korcis with the Hungarian Symphony Orch. (No link, but since Bartok was Hungarian, their performance is likely as legit as can be.)

- Bartok's writing spans from glacial cold emotions to extreme expressionistic passion. (A point for the mixer to address).

Another mixing mastering reference pointed to HZ's _Inception (Time)_ as the best reference for orchestral. Specific mention is made on increasing EQ brightness as the song progresses, then use of the entire frequency spectrum during the peak of Inception.

Now I am really curious about the final mix of OP's mock-up.


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## ptram (Jan 2, 2022)

b_elliott said:


> Now I am really curious about the final mix of OP's mock-up.


Not yet final, I fear, but a third try.

Bartók, Music for strings, percussion and celesta (VSL VI) - Mockup 03

What I did:

- Review the dynamic curve in Dorico. I set it so that the central dynamics are smoother (value 1.9).

- A finer balance of dynamics (via CC11 and some more adjustments on the instruments' volume).

- Some finer adjustment of the EQ, compressor and limiter in Ozone.

The result sounds smoother to me. I've yet to do a comparison in Metric AB to see how it is against the reference track.

Clipping remains at the big timpani hit (that I anticipated a bit, to avoid overlapping with the string's _fff_ chords). If I remove it, I also remove the efficacy of the hit. Or I end up overcompressing and squashing it. I know there must be a way to have both, but I've yet to find it.



b_elliott said:


> Bartok (similar to Varese) wrote detailed instructions on how the 2 orchestras were to be set-up with placement of the basses in the center back.


There is a little thing Bartók didn't maybe consider: the stereo effect of suspended cymbals. There are two in the score, but he only writes about a single position for cymbals. Assuming there are enough players (you would need three percussionists), having them on the opposite sides of the stage should make a very interesting effect.

(How beautiful is this piece, if it can stand all the mistreatments of my mixes?)

Paolo


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## b_elliott (Jan 2, 2022)

I'm liking this Paolo. I agree it is quite the work.

1. Cymbals seem perfect. I downloaded the IMSPL pdf score and noticed it is written as a cymbal roll then crash. In the Mackerras recording I could not hear the cymbal roll -- yours I do.

2. The peak @ 5 minutes still sounds clipped. When I look at that the FFF section (4:57 - 5:05) in Metric A/B it registers -7 LUFS-I (with short term peaks at -6 == squashed per the dynamics meter). On my headphones it sounds distorted. Unable to play on speakers due to time of day (people sleeping).

Side note: I noticed the HZ _Inception Time_ reference mix similarly ranges loud to squashed at its 3 minute climax; however, the difference is its LUFS-I for the loudest part is -9.5 versus your -7.6. Metric AB classifies your loudest section as "a really loud master" vs HZ's _Time_ as only "a loud master".

3. When I first listened to your mix I thought the intro instrument was a french horn. That made me download the score. I see it is a viola. Wondering if it is the wetness of the sample that is making it sound horn-like? Not sure if it is EQ or reverb issue. No other samples were an issue.

4. Celesta. Now I hear it. FWIW it is more prominent compared to the strings in Mackerras' recording as well as this live Radio France performance. Your decision to leave it as is or feature it more.

Please take my points with a grain of salt, considering my lack of experience. But I am certainly a fan of this work and what you are mocking up. As I listen, I imagined Bach would be intrigued by Bartok's fugue. Lord knows what he'd make of it with a pipe organ interpretation. Scary....

Cheers, Bill


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## Aldunate (Jan 3, 2022)

Sounds great. What are you using to make it sound older?

A low ratio (i.e. 1.2), low threshold (almost always working), high knee (approx. 12 dB) should work as a clean leveller, reducing the dynamic range without squashing it and keeping the "grain" of the compression.
Using a second one to compress or limit further would then sound more transparent.


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## ptram (Jan 9, 2022)

In the end, the only thing that seemed to work was 'riding the fader'. I smoothly lowered the volume of the Timpani part when approaching the big hit, and smoothly raised it again immediately after that.

The signal is still clipping for a few samples, but I think this is inside the range of acceptable clipping. Or should any clipping be absolutely avoided, even in cases like this? I'm disturbed by it being there, but I don't know if I should learn to live with it.

Bartók, Music for strings, percussion and celesta (VSL VI) - Mockup 04



Aldunate said:


> Sounds great. What are you using to make it sound older?


I don't know. In the first tries, I used a vintage console strip adding some 'grain' to the sound. But then I removed it. Yet, the sound is indeed, while transparent, somewhat vintage. Maybe it's in the samples, or in the orchestration…

EDIT: I see I used the analog version of Ozone's Dynamic EQ. It seems it is giving to the final sound some slight patina of 'vintage'.

EDIT2: Pulling our my head from the audio mixing sands: maybe it's the muted strings, to be giving an impression of 'aged'?

Paolo


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## labornvain (Jan 9, 2022)

Waves MV2. Waves' most essential plugin-- or at least the most irreplaceable. It compresses from the bottom up ( upward compression) and the top down while remaining entirely transparent.

It also uses some fancy algorithm to keep the noise floor down while you're cranking up all of lower volume content.

Don't be fooled by its simplicity. This is one of the greatest breakthroughs in compression technology in ages. As Jack Joseph Puig said, "if I didn't have MV2 I would quit mixing."


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## ptram (Jan 9, 2022)

labornvain said:


> Waves MV2. Waves' most essential plugin-- or at least the most irreplaceable. It compresses from the bottom up ( upward compression) and the top down while remaining entirely transparent.


I've a different problem than keeping the dynamics homogeneous. I have to preserve the full dynamic range, while avoiding a peak to distort.

Upward compression would make the ppp passages unnaturally strong. Downward compression is squashing the full orchestra peaks, again sounding in an unnatural way.

Maybe I'm doing something wrong, but everything is leading me to believe that only local manual control of the volume on some instruments can do the trick.

Paolo


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## b_elliott (Jan 9, 2022)

ptram said:


> ---
> 
> The signal is still clipping for a few samples, but I think this is inside the range of acceptable clipping. Or should any clipping be absolutely avoided, even in cases like this? I'm disturbed by it being there, but I don't know if I should learn to live with it.


Not sure which Pensado _Into the Lair _episode I watched but he had his VU meters hitting the red but had no issues with it. He basically said his ears were saying he is OK, just don't do this at home.

Your mix sounded fine. No distortion when I listened with headphones. 
Cheers, Bill


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## ptram (Jan 10, 2022)

b_elliott said:


> Not sure which Pensado _Into the Lair _episode I watched but he had his VU meters hitting the red but had no issues with it.


Thank you very much for listening again, Bill.

Paolo


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## bobby b (Jan 13, 2022)

ptram said:


> Clipping remains at the big timpani hit (that I anticipated a bit, to avoid overlapping with the string's _fff_ chords). If I remove it, I also remove the efficacy of the hit. Or I end up overcompressing and squashing it. I know there must be a way to have both, but I've yet to find it.


Since you can't use compressing no more, try control volume fader manually? If you haven't already. With broad strokes across seconds.


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## Emanuel Fróes (Feb 27, 2022)

ptram said:


> Hi,
> 
> I'm working to an orchestral mockup, starting from a few violins in _ppp_, with a buildup leading to a climax in _fff_, played by a full string orchestra and timpani rolls.
> 
> ...


This was and is my issue many times. Basically you do it by automation of the master or mix at the end, or you use something like Waves Mv2, to expand the lows. I use it as a correction after the Ozone9. Another wild option is to download the Davinci Resolve for free and do it manually, where you can see the wave form changing.

The most correct way is that you mix already from the velocity and midi parameters only on and on, to be sure the next mixing elements are really external to the core of the composition . Since there is no maestro this can still be very hard to put togheter. There comes a master fader or something like this…


Since most of music for media is very homogeneous, it is natural that composers use this "flat" and weak behavior of dynamics to their advantage, because the comissions may ask for something homogenous . They use highs and lows, but not in the traditional way. To achieve the "traditional" contrasts and still mix properly is an art

But there is a lot to this! People who can teach this well normally never talk about this, but about flattening the curve…

But the best technique I used, i reserve for a video of mine.

Here in this music I dealed with this problem of wide dynamics vs mastering  . But far from my vision of orchestral dynamics and how a DAW can express it quite so good as in concert hall, if we have skills for a fine tune of this hard parameter to master, and all illusions that it has.


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