This has nothing to do with logarithmic scaling, so humor me and keep that out of discussion here for a minute.

I think you are trying to find a way to use VSL's CC7 default values since that appears to be the way VSL is level-balancing Synchron instruments against each other.  Yes so far?

Defaults

So the defaults are to have CC11 at Max....  and then you can adjust CC11 from 0-127 which will essentially attenuate the signal even further as you desire over the range set by CC7.    Note that in a way CC7 provides the total dynamic range...and CC11 adjusts the level linearly within that range.   When CC1 and CC11 are set to max 127 you can think of that as the max possible volume of the instrument, which VSL is level balancing against other instruments and their max levels...by using CC7.

Breathing Room

Now I think you are saying that's all fine and good, but you want some breathing room because if you use that default setup and you start drawing in a bunch of expression curves to your composition and then suddenly you find out that its not quite loud enough even when both CC1 and CC11 are set to max.  And you need to be able to bump it up a little more.  Fair enough.   Also level balanced sample orchestras are probably never perfectly balanced.  So you need some room to fudge it.  Fair enough.

That means you didn't probably have the max volume level set right to begin with via CC7.  So assuming that you could bump up CC7 a little bit to get the max level where you want it...but then all the curves you already drew for CC11 would all be too loud.  So I don't think that's a good solution really, you probably should leave CC7 alone once you start recording tracks and programming CC11 curves.

So you suggest that you could have CC11=110 set as the home base "max" position, and that way if you get halfway into your composing you realize that you need more max volume then you originally thought, then you can just pump up cc11 past 110...take it to "Eleven" so to speak...  

Reduced level

So assuming you set CC11=110 and generally treat that as your max instrument level....and only rarely go to Eleven...I don't think there is anything else you need to do to compensate anything.  Just be consistent with all your tracks, always use the same approach.  Me personally I still prefer generally to use CC1 (velocity XF) for each instrument dynamics anyway...and I would generally leave CC11 set to its home base position (whether that is 110 or 127), and only rarely use CC11 to change dynamics slightly louder then max perhaps.  I don't see the point of using CC11 instead of velocity XF.  But no matter, some other sample libraries don't provide velocity XF, so that would be the case when you would use CC11 and just always treat the max as 110 except for rare situations where you have to go to Eleven as mentioned above.

In the above, yes the overall instrument output will be lower then factory, since you'll be using CC11=110 as the max in order to have room to go to Eleven.  That is an attenuated signal by a factor of 13%.  That doesn't really matter as long as all your instruments are following the same rule, then their output will all by attenuated by 13% and the level-balancing will be preserved...just everyone at a lower level.

MirPro

As a side note, MIRPro does essentially exactly this.  When you use MirPro instrument presets, they are all dropped way down in level in order to preserve level balancing against some other really loud instruments such as Timpani.  It means that other instruments output from MirPro and very low levels, way way way below -18db, which many people including myself have struggled with a little bit, but it doesn't matter, its all relative.  

Trim Plugin

If you want you can simply add a trim plugin to the master bus to bring them all back up again evenly by the same amount.  How much gain to add would be 13%...but of course gain plugins often use use DB units to specify the gain,, so you'd have to figure out how many DB's of gain to add, one setting for the entire orchestra.  One master bus with just one trim plugin using some amount of gain that brings them all back up, relative to each other, by a factor of 13%.  

That's all you really need to do.

Adjusting CC7 back up

I think it would be very difficult to try to adjust all the CC7 values up from the factory setting in order to compensate for the 13% reduction you are making with CC11=110 because of the math I mentioned where its a percentage of a percentage.  I am not sure you could simply raise CC7 by 17 (which is 13% of 127) and have the level balancing preserved.   It needs to be bumped up by 13% from the current cc7 range.

If you want to try the math, you need to use percentages, maybe you were already suggesting this but I could not understand what you wrote, but simply take the following math, assuming VSL's cc7 val and changing CC11 from 127 to 110.

@Another User said:

new_cc7_val =  oldCC7val / (110 / 127)

Then change CC7 to the above and use 110 for CC11 and it should be the same output.

Also, some of the instruments may not have room to go up enough to compensate, before hitting the ceiling of 127 after running this calculation.

Therefore its going to be easier to compensate that 13% signal loss from CC11 by simply adding trim plugins after Synchron in the plugin chain, you can either add one trim plugin on every track, and make sure they are all set the same...or just send them all to some common bus and put one trim plugin there that raises the level ~13%.

I am essentially doing exactly that with MirPro now when I have ensembles and stuff that will never have loud sounds playing....

I think involving DB calculations just complicates the issue because of the logarithmic nature of DB as a measurement.  You just need 13% of gain added back. whatever DB that turns out to be will be different depending on the loudness of the instruments.

Dewdman, it's not just a matter of getting louder. The Expression control can be used when you don't need a change in timbre, therefore prefer to leave the VelXF control unchanged, and change the relative volume with Expression. Sometimes, you want both controls to act together, to give more emphasis to the change.

Having Expression already at maximum would prevent any change up. I don't know how others do in this case, but I find much easier to start with some headroom.

Your handy formula to calculate the volume compensation is very useful. Funny how it give a similar result to the one I found, showing how little the difference can be with these small change. So, maybe it's not all that difficult to preserve the Natural balance set by VSL for their presets.

Paolo

@Another User said:

Having Expression already at maximum would prevent any change up. I don't know how others do in this case, but I find much easier to start with some headroom. Your handy formula to calculate the volume compensation is very useful. Funny how it give a similar result to the one I found, showing how little the difference can be with these small change. So, maybe it's not all that difficult to preserve the Natural balance set by VSL for their presets. Paolo

Andreas, I'm not exactly sure what you meant in you post above by "not identical" and "results in slightly different curves". Am I to understand that the individual Master Volume fader and the individual Expression fader each have a unique 'transfer function' or 'calibration' - call it what you will - in terms of attenuation dB versus CCvalue? I think that's what you've asserted but would you kindly clarify?

I promise you I'm not asking merely pedantically; I do have a real and practical need to know.

[UPDATE]

Well I've just done many more test measurements. And now my question is different. Now the issue is quantitative rather than qualitative.

Yes I have now discovered what a few spot checks didn't reveal previously: there are indeed numerous small differences in calibration between the two faders. I was caught napping by this unexpected circumstance.

And yet strangely, many CCvals give exactly the same attenuation dB level in both faders - within my meter's resolution of 0.01dB at least. The differences seem to be just peppered around - I haven't yet come up with any plausible purpose..

I've found that only 4 of these differences come anywhere near to being possibly significant (0,58, 0.61 0.70, and 0.76 dB, all at below -30dB of attenuation). The other differences are less than 0.4 dB, many of these being so small that they're impossible for me to distinguish from experimental error. I don't imagine that any of them are likely to have any noticeable impact on gain-staging and mixing in actual practice. It's said that most people can't distinguish level changes of less than about 1 dB. All of these differences are less than 1 dB; all but a few being far less.

So my question, Andreas, is changed.

Why is there this strange peppering of insignificantly small 'calibration' or 'transfer function' differences between these two main faders? And why your caution? Why do we need to be mindful of which way around they are connected? Cannot their respective attenuation dB levels be simply summed in order to know their combined attenuation?

I don't actually mind the presence of these small differences because I don't see how they can significantly affect any of our usual, practical level-adjustment activities. But is there perhaps some other factor, maybe some hidden conditional attenuation mechanism operating between these 2 faders that prevents their combined use from being straightforward and virtually 'ambidextrous'? 

[Deleted due to trolling]

(moved to other thread)

@PaoloT

I just made some measurements because I'm curious...in case this is of any help here is what I got.

I used Synchronized Dim Brass Trumpets XF

The default from factory is CC7=90, CC1=127, CC11=127. Max peak = -7.32 dBFS, RMS=20.84 dBFS

Moving CC7 to full max, I get Max Peak = -1.29 dBFS, RMS=-15.10 dBFS

So the first observation is that when all controls are turned all the way up, Synchronized Dim Brass Trumpet ALL XF gets very very close to clipping. Dangerously close actually, but it only clipped for me if and when I played a chord, which its not designed to be used that way anyway. So the point is, it looks to me that VSL endeavored to keep the signal coming out of Synchron in this instrument as close to 0 dBFS as they felt was possible.

It also shows that their default setting for cc7=90, which lowers peak to -7.32 dBFS, which presumably is how much they feel Trumpets need to be lower to balance realistically with other Synchron instruments.

Trumpet is a fairly loud instrument in the orch anyway, and at -7 dBFS its not at all a problematic level in terms of headroom IMHO.

Change CC11=110 and we get Max Peak = -9.90 dBFS, RMS=-23.34 dBFS

Now in my view that is still totally usable, but its also a little bit low, because of the CC11 home base point set at 87% of full unity.

In order to use a trim plugin to bring all the instruments back up together...I believe a corresponding gain plugin bump would be: +2.5db, which resulted in following measurements: Max Peak= -7.36 dBFS, RMS=-20.85 dBFS

That is close enough for government work, as we say....

I do want to point out one more thing though, if you bump the level back up using either a trim plugin or CC7, then later end up taking CC11 to "Eleven" (CC11=127), then you hit MaxPeak = -4.9 dBFS and RMS=-18.3 dBFS.

That's still not clipping range, so may be ok...but if working with something like percussion or something that has capability to get really loud, there is always a remote possibility of clipping that instrument if and when you push it to "Eleven" with the CC's and the Trim plugin adding +2.5db on top of that. So watch out for that. For example, with this instrument, if I move both CC7 and CC11 to max (while sending through the trim plugin at +2.5db), it definitely clipped.

One advantage of using CC7 to compensate, rather than a trim plugin is that you wouldn't have to worry about clipping like that (as long as you don't play chords in this case). Because VSL gain staged this instrument to peak out at -1.29db when all CC's are at full max and playing monophonic lines.

The disadvantage of compensating with CC7 is that some instruments may not have enough room at the top to fully compensate them, depending on what VSL's default CC7 setting was. Also, you have to make this adjustment in every single instrument preset as custom presets (which you're doing anyway when you set CC11 to 110).. whereas with the trim plugin you can send them all through a bus and compensate them all with a single trim plugin.

Hope that makes sense.

All of the above makes me ponder my MirPro setup.. MirPro presets impose quite a bit of attenuation also for the sake of level balancing. Actually significantly more than the Synchron CC7 approach. I will have to do some similar testing both inside and out of VePro to see what results I come to. But I think in the case of using MirPro, I would set CC7=127 for all instruments and then set CC11=110, as @PaoloT is doing, for the wiggle room, but with out the CC7 compensation (since I will use MirPro's natural volume slider for that.

its not clear to me yet whether MirPro presets are calibrated already to take into account that Synchron instruments have CC7 set lower or not and then perhaps adjusts them a little more, but what I can say is that by default the MirPro attenuation with Natural volume is rather significant and usually feels like too much, so I will almost certainly be using some kind of trim plugin to bring it back up, but it also means that if and when I go to "Eleven" with CC11, there is more possibility of clipping something if I'm not careful. So that will be something to watch out for.

UPDATE: I measured the results of running this default Synchronized Trumpets through MirPro, and yes its calibrated to depend on the factory default of CC7=90 in this case. When running this Synchron instrument through MirPro instead of using the IR's in the instrument, with CC7=90, CC11=127, and MirPro using Dim Trumpet instrument preset as well as Mirx mode for character....the resulting level measured very very close to the above scenarios without MirPro...so conclusion is: they have set these MirPro preset's natural volume level to expect the level coming from the VSL CC7 defaults, as far as I can see, at least with this instrument. That's a good thing...but it means basically I could pontentially change all my Synchron presets to use CC11=110 as the home base, and either compensate with cc7 bumped up a little bit for each instrument or a trim plugin with +2.5db on the whole bus...and they should all work with the MirPro presets and the default Natural Volume levels. and those Synchron presets would still workout without MirPro in that case.

The only thing I think I don't like about VSL's approach of using CC7 for the level-balancing, is that a CC7 fader is often exposed in numerous places on most DAW midi tracks, making it all too easy to accidentally bump this fader and lose the carefully planned orchestra balance.

I think I would rather reprogram my presets and swap those around, while setting the 110 thing, so that CC7 is set to 110 and CC11 gets whatever value that VSL put into CC7.

That way, the CC11 value will represent the orchestra balancing...(instead of expression) and CC7 would be used for expression...which makes it very convenient to access from the midi fader in most DAW's without upsetting the intended orchestra balance.

So in summary... This is what I will do:

1. Reprogram Synchron presets to assign VSL's cc7 value to the CC11 slider
   
   
2. set cc7 to 110 as home position
   
   
3. add trim plugin to my main orch bus with +2.5db of gain
   
   
4. use either with or without MirPro in this configuration
   
   
5. Use CC7 for expression above and beyond CC1 XF

@Dewdman42 said:

All of the above makes me ponder my MirPro setup..  MirPro presets impose quite a bit of attenuation also for the sake of level balancing.  Actually significantly more than the Synchron CC7 approach.  I will have to do some similar testing both inside and out of VePro to see what results I come to.  But I think in the case of using MirPro, I would set CC7=127 for all instruments and then set CC11=110, as @PaoloT is doing, for the wiggle room, but with out the CC7 compensation (since I will use MirPro's natural volume slider for that.

I asked Dietz about this a while back.  Specifically: what the CC7 should be set to when using something (either VI or SYN-zed with the SYN-zed impulse turned off) in Mir Pro.  He said that as long as CC7 is set to the same value across the board for all instruments, it shouldn't matter *what* it's set to as the natural volume values (which are just adjustments in gain, not CC7 values) will then all be based off the same CC7 "foundation".  He did mention that Herb tweaked the natural volume settings with CC7 set to 100 (as he recalled), and I decided just to go with that since CC11 increases in granularity the lower CC7 is from its max value.

What I mean by that is that Expression has iterations from 0-127, but within the volume range defined by CC7.  If CC7 is set to 127, Expression and CC7 are basically the same thing.  But if CC7 is set to, say, 100, then Expression gives you 128 iterations *within* the 0-100 window defined by CC7.  So, each step or iteration of CC11 then results in a finer volume change, which I enjoy as "smoothing" on top of VelXF (CC1 in my case).

@Another User said:

and I decided just to go with that since CC11 increases in granularity the lower CC7 is from its max value.

What I mean by that is that Expression has iterations from 0-127, but within the volume range defined by CC7.  If CC7 is set to 127, Expression and CC7 are basically the same thing.  But if CC7 is set to, say, 100, then Expression gives you 128 iterations *within* the 0-100 window defined by CC7.  So, each step or iteration of CC11 then results in a finer volume change, which I enjoy as "smoothing" on top of VelXF (CC1 in my case).

I understand you perfectly here.  The interesting thing here is that the desired level of the instrument needs to be pretty much where it needs to end up for that increase granularity to hold..  If you were to later add some audio gain to the signal (after reducing the gain with lower CC7 setting), then the increments would simply be expanded into the same granularity as before... if you see what I mean..  But yea if lowering the CC7 instead of lowering audio, gets you the correct final desired level for that instrument, then that will result in finer granularity while using CC11

@Dewdman42 said:

I understand you perfectly here.  The interesting thing here is that the desired level of the instrument needs to be pretty much where it needs to end up for that increase granularity to hold..  If you were to later add some audio gain to the signal (after reducing the gain with lower CC7 setting), then the increments would simply be expanded into the same granularity as before... if you see what I mean.

I do see what you mean, never considered that.  My approach (thus far) has been to set all the CC7 and Gain levels (natural volume +/- my own preferences, etc.) and then let VelXF and CC11 do everything regarding volume and dynamics.  I'm not sure if this is the best way, but so far it's working for me for anything needing to sound more-or-less like orchestral whatever.  However, given what you said, I can see how that "fixed" balance could be thrown off via something like stereo bus gain automation after the fact.  One solution (if needed) that comes to mind would be to render/freeze the audio before automating gain - that way, the balance is retained because gain automation is raising the level of a simple audio file, not one of three (four?) different sliders...

@Dewdman42 said:

In some ways it could be better, to use mixer automation instead of cc11 for that particular aspect of dynamic expression above and beyond velXF.

That's a good point!  I'm going to mess around with that a bit...

Something to consider, when setting the starting Volume in Dorico, is that it seems Dorico sends an initialization value to its mixer and to the Master Volume faders of the VSL players. One can change this value in the Init event of the expression map for that instrument. Or can insert a different CC7 value at the beginning of the score.

https://www.vsl.co.at/community/posts/t57399-Dorico--VSL-Synchron-ized--volume-difference-between-instruments

Paolo

@Macker said:

Paolo, I've been measuring what Synchron Player's Main Volume and Expression faders do, but haven't finished that yet - in every measurement it can take ten minutes or more until the meter readout is settled to 2 decimal places. (I'm using BlueCat's DP Pro Meter plugin, which does lots of very useful stuff, including RMS level averaging in real time over however long I want.)

A few highlights of my findings so far:-

   Both faders have the same logarithmic transfer function, as one would expect.

   -1.0 dB is at MIDI value 120; (all dB values here are relative to 0 dB attenuation at MIDI val 127.

   -2.5 dB at 110;

   -4.1 dB at 100;

   -6 dB at 90

   -8 dB at 80;

   -10.3 dB at 70

   Going all the way down to -83 dB at 1, then complete cutoff at 0.

Just saw this thread. FWIW this is to confirm that I got exactly the same results as Macker’s table using a lazier method.  In my DAW (Studio One), dB change is the default unit for changes in Expression. 

So I put in different values of dB change for an Expression automation point in S1, and noted how the Expression MIDI values changed in the Synchron Player for that instance.  I’m glad that the equivalence table I’ve been using from doing that has been confirmed by someone who knows far more than I do.

This and the related threads (even with the twists and turns of the discussion process) have provided me as a newbie with lots of insight into questions I’m trying to figure out.  My thanks to Macker and the other participants.

and I also measure the same results when I have done so, but its still important to keep in mind that the results you measure with an external meter do not tell you what the actual faders are doing.  They report resulting change in DB level as measured by the meter plugin.  The DB concept is a logarithmic concept in the way it is measured and reported.

A totally linear fader behavior will translate into a DB measurement that is logarithmic in nature.  That is because of the way DB's are measured as a ratio, based on a reference level.  

The VSL devs have already clarified that the actual faders in Synchron are using a simple multiplier using from 0.0-1.0, which is essentially saying a percentage of the amplitude.  Nothing about DB's are taken into consideration while processing the CC faders.  They simply apply a percentage of the amplitude and that's it.  That is a linear response, in terms of the wave amplitude.  However, a DB measuring device will report that linear amplitude response on a logarithmic scale. 

We are talking about two faders here, they are as multipliers, which means the net effect of the two faders together is as follows:

@Another User said:

dBFS = 20 x log10( multiplier)

However keep in mind that the above is not taking into account the reference level.  

There is a lot of confusing information that is being circulated now on this forum about DB's in general, and the poster keeps moving his post to a new thread when I have tried to clarify these facts, I'm not really sure why other then it seems he really wishes VSL would add DB notches on the GUI next to the CC7 and CC11 faders...which is not in itself a terrible idea, but without having the notches there, the above comments will tell you what you need to know about these two faders.  They are a applying a simple linear percentage to the amplitude.  Linear Amplitude can be converted to logarithmic DB measurement using the above formula.

You can either compute both faders to a DB value and add them together, and also take into consideration the reference level..to arrive at the total final DB value...  or you can multiple them together (as the first quote above), resulting in a final net multiplier, which you can convert to DB once from that, but also you need to take into consideration the reference level if you really want to do that right.

Its a lot of math juggling though.  For me its quite enough to just think of the CC faders as simple multipliers as stated by VSL...they do that job perfectly.  But I can see if someone is really used to keeping track of how many DB they are attenuating things or re-gaining back later, they want to know how much attenuation they are applying in terms of DB.  It would be nice if VSL added DB notches to the GUI to show those cc fader changes as -dBFS attenuation also for that reason.  so a person could set one fader to X number of -dBFS and fader2 to Y number of -dBFS and then set an external gain plugin to X+Y +dBFS to make up the exact same amount of gain..  theoretically.  I get the point of that, but I also haven't found it that necessary actually know that or use it that way.

Welltempered, thanks for your kind words, and for your info about Studio One's automation dB calibration in connection with Synchron Player's Vol and Expr faders.

Indeed without this valuable help from DAW automation's indication of dB, I suspect it can at times be a bit of a struggle for some users when trying to use either or both of these Sy Player faders for mixing purposes - especially given the numerous technical myths and plain incorrect 'facts' that have been posted about these two faders.

I'm just doing what I can to pop these myths and debunk the worst of the specious 'factoids'. (However, that doesn't mean I'm like Doc Holliday looking for a gunfight at the O.K. Corral, lolol. It doesn't serve excellence in any field to dignify the efforts of any laughably cocky champions of their own incompetence; there's no honour in taking them down; I'm content to let them disappear up their own rear ends, cuz, y'know, entertainment. lol.)

As an aside, I've been intending - If I find time - to investigate the accuracy of automation's indicated dB values for the AU Synchron Player in Logic and the VST version in Cubase, versus measured end-to-end automation of the two Sy Player faders by these DAWs. I've already noticed some small numerical discrepancies when using Logic, but thus far, glad to say, nothing to write home about.

The vast majority of members here I'm sure don't need reminding of the differences between gain, attenuation and level, when dealing with audio; nor how these things have usually been dealt with for many many decades.

To confuse these things would be like, when dealing with money for a purchase, confusing added tax or discount deduction with the total amount to be paid. Pretty basic stuff, uh? lolol.

But ah well, there's always a few. I guess it's just another of the hazards these days when so many depend on Googling to cover for lack of basic knowledge, understanding, experience and skill in this or that field of interest.