I was trying to sort through your question earlier before having the benefit of manuals to search. What didn't resonate for me was your use of "dB tool". I wasn't sure if that was a Montage term or something "made up". It looks like the latter. So I'll now assume "dB tool" is referring to effects or settings that can affect the output characteristics of the connectors you are interested in. Although not stated, it is fairly clear that you are referring to the "Assign L&R" outputs when you write about the "additional outputs". This is easy to decode because other than digital outputs, these are the only outputs you can redirect PARTs out to and be "different" than Main L&R.

It's important to understand that Assign L&R outputs do not have the Master effects or the System Effects (meaning "Reverb" plus "Variation" effects) applied. To be clear:

Assign L&R outputs do not have Master Effects applied (Master effects are "turned off" for Assign L&R)
Assign L&R outputs do not have System Effects applied (Both Reverb and Variation effects are "turned off" for Assign L&R)

The only chance to apply effects to the Assign L&R outputs is in the Insertion Effects - which are not "stripped out" of Assign L&R like the Master and System effects are.

Knowing this, you may need to modify your paradigm for what you are trying to do.

Note that there is an Assign L&R Gain setting ("Audio Setting" - global/system) that can serve to trim the Assign L&R output. I would think this would not "undistort" a signal if there was clipping - but you can see if you gain any headroom by trimming the Assign L&R output. I see this more as a way to match an input buffer than a way to prevent the output from distorting (independent from an input).

Also, I know you are asking primarily about output distortion. However, you will be connecting these outputs (Assign L&R) to something downstream. So its input properties need to be considered as a signal that does not distort on output can cause an input buffer distort due to mismatch. This part of the signal flow makes a complete answer difficult without being able, necessarily, to know all the properties of what you are connecting to downstream.

One could probably start to answer how much headroom there is on the output buffers by connecting the output to something with "infinite" range (relative to the Assign L&R max drive characteristics) and start to feed a synthetic signal - such as a sine wave and superimpose multiple waves using FM-X or even the sampled sine waves. Turning the volume all the way up on all PARTs/elements/carriers - then look for clipping on output. But the determination is probably much easier on your end by using something non-synthetic like your actual sequenced sound and bumping up all the PART outputs associated with the sequenced sound to maximum (synthetic step) and listen, with your ears, for distortion. Then dial back the levels until distortion stops and keep going until you arrive at your desired level. You'll have a sense for margin once you reach hearing "no distortion" and how much level you have between that point and your desired output. If you have distortion at your desired final value - then you would have negative margin and something would need to be done.

I haven't heard distortion as a "normal occurrence" when making my own Performances -- so I would think, in general, you should have headroom with your sequenced PARTs routed to Assign L&R when using levels in a sensible way. If all levels are set to 127 - then this would be a fairly good clue that, by construction, something is "wrong" with the leveling.

Hey Jason,

Sorry if my terminology is not correct. I am new to the game, and I am basically repeating what someone else posted to me in another forum. I requested additional clarification and feedback, but no one answered the call in that forum, so I posted here instead.
Also note that I do not yet have my Montage, and I just recently decided (after 2 months of research) to go with the Montage8 over the Kronos 88. One of the key deciding factors was Polyphony.

My plans: When I gig, I plan to have several sequenced tracks running as a 'audition/song', such as Drums, Bass, Guitar sounds, strings, etc. (likely on parts 9-16 on the Performance). I also plan to have numerous scenes, whereby at times the keys that I manually play while the sequenced parts are playing, could be all 8 parts (1-8) of that same performance using splits & layering, such as piano with multiple layered strings in the middle section of the keyboard, some other sound in the lower notes split section, and a synth solo sound or multi layer vocoder sound in the higher split range.
So basically, I could have all 16 parts(or most of them) hitting notes at/near the same time in certain songs, therefore I need to keep Polyphony in mind for that, but also I need to keep Output in mind, utilizing the best settings to avoid clipping/distortion.

And that is kinda the basis for why I am asking about Outputs in the 1st question & possible limiters/etc in the 2nd question.

- So regarding best Output options - If in a Performance I use parts 1-8 as I mention above(to play live), and parts 9-16 via the sequencer, would I gain output volume (and possibly avoid clipping/distortion) by assigning parts 1-8 to the "Main L/R outputs" and parts 9-16 via the "Assign L&R" outputs? (As opposed to sending all 16 parts to just the "Main L/R outputs"..!?)
Or would splitting up the Outputs like that not gain me anything; whereby the clipping/distortion would still possibly occur the same whether I send it all (16 parts) to the "Main L/R outputs", or whether I send half to the "Main L/R outputs" & half via the "Assign L&R" outputs?

Does the total output that could start to distort/clip change if I were to use the "Assign L&R" outputs to redirect half the output volume on half of the parts, or does total output distort/clip potential happen before the output assignment stage? Basically, asked another way, does one affect the other, or are the "Main L/R outputs" & the "Assign L&R" outputs completely separate from each other in terms of output clipping/distortion?

- Regarding the second question about a Limiter/etc. - Is there such an effect that can be assigned to (at least) the "Main L/R outputs"?

In the event that I still (as you said)

have distortion at your desired final value - then you would have negative margin and something would need to be done.

, what would you suggest I do to fix it?

- Are their level/output 'meters' that I can monitor/view on the screen for both the "Main L/R outputs" and the "Assign L&R" outputs (or at least the total output!?), that will allow me to 'see' if it is clipping, so that I don't have to just rely on hearing/listening for distortion?

Thanks,
Darryl

Montage has meters for the sum of all PARTs and each individual.

I don't know what you really want to do with additional outputs. If it's just to control clipping - I'm not sure you need this. I also have no idea what you do. You have more options than what you've described including the sequenced PARTs mono to Main R and the "live" PARTs mono to Main L (if you still want separation). You may not need this at all - and I'm not advocating "killing" stereo - only that there are more options than described if separating PARTs into different channels will be your top goal.

Compression effects are available for Main L&R. Even Assign L&R has these - but you need to use Insertion (each PART).

If you're relying on internal ARPs - you'll find limits in using all 16 PARTs. If you are externally sequencing - then PARTs 9-16 would be used as targets for this and 1-8 for your "live" use. Only PARTs 1-8 can be simultaneously triggered by the keyboard keys. PARTs 9-16 cannot.

The effects architecture as explained before is one of those fundamentals. There are a few of these to learn - and it's best done when you get the keyboard given the "check is in the mail". I think you can go overboard with defining the twig/branches and leaves on the tree when this will turn more into a forest story (broader strokes) when you get the keyboard. Many of the finer details you'll be able to self-serve. The "gotcha" rules are good things to ask about once you've oriented yourself a bit more.

I think focus on distortion, at the moment, may be a distraction. Get the board and your Performances together and evaluate if you need to spend much time with the distortion question. Likely, energy will be better focused on general orientation items with the assumption that distortion is not likely to occur. No one uses the keyboard the same - so I'm not saying no-way-no-how. Just ... to cross that bridge (if ever needed) ...

Montage has meters for the sum of all PARTs and each individual.

Do the meters for the the sum of all PARTs, show all output levels of the Main L&R parts combined with the Assign L&R parts, whereby the sounds from all the 16 parts pass through this Output location of the Synth with potential to clip/distort here, before separating/re-routing parts 1-8 to the Main L&R output ports and parts 9-16 to the Assign L&R ports (as is what I considering doing)
Or are the Main L&R and Assign L&R outputs separated at/before this output metering location, and only sound from parts 1-8 show in the "meters for the sum of all PARTs"?

I don't know what you really want to do with additional outputs. If it's just to control clipping - I'm not sure you need this. I also have no idea what you do. You have more options than what you've described including the sequenced PARTs mono to Main R and the "live" PARTs mono to Main L (if you still want separation). You may not need this at all - and I'm not advocating "killing" stereo - only that there are more options than described if separating PARTs into different channels will be your top goal.

I am considering using the additional outputs for possibly 2 reasons:
Reason 1 ... just to control clipping on output (but only if the Assign L&R parts have completely separate output from the Main L&R parts output, and they don't all first pass through some internal output location on the Montage whereby all 16 parts of sound ("live" & sequenced) combine & could potentially clip/distort before being routed to the different external output ports).
And reason 2 ... I might want to eq/mix the sequenced parts (Drums, Bass, etc. - parts 9-16) differently than the sounds that I play "live" (parts 1-8). (Note - When I say eq/mix in this instance, I mean on a separate physical PA Mixing Board that has the two inputs for Main L&R and two inputs for the Assign L&R from the Montage output ports, as well as a Vocal mic that I will sing through.)
I definitely want to keep everything in stereo!

My theory was/is that if the Assign L&R output is completely separate from the Main L&R output, then that would offload having everything come through the Main L&R outputs, therefore giving those outputs more headroom before clipping, because none of the sequenced sounds would be added onto the sounds that I play "live" on the keyboard parts 1-8, as they would go out the Assign L&R outputs instead.
However, the question becomes: Do all 16 parts of sound first pass through some single internal output location on the Montage with all 16 combined/layered, whereby it could potentially clip/distort, prior to being routed to their different respective external output ports as I mentioned above (Main L&R and Assign L&R)? Or is it the case that when separating/routing the 16 parts into the 2 different respective external Main L&R and Assign L&R outputs, then the separated groups of parts(i.e. 1-8 & 9-16) would never pass through any single internal port/output/circuitry and remain completely separated, whereby the "meters for the sum of all PARTs" only shows the output of the Main L&R parts 1-8, and not parts 9-16..!?

I think focus on distortion, at the moment, may be a distraction

Right now I have some time before I get my hands on the Montage and focus on/dive into the main stuff (Programming & configuring all the parts & Performances, etc). While I'm waiting, I want to temporarily focus on & find out what my best options are (what might work and what might not) with certain design things, such as the Output configuration that might work best for what I am planning to do when gigging. I am taking just this one keyboard with me to gigs for everything (no other keyboards, laptop, nothing. Just the Montage & my vocal mic). All backing tracks (drums, bass, guitars, etc.) will run via the onboard sequencer (parts 9-16), and everything else I will play on the keyboard (parts 1-8). With some songs/auditions I may be hittig the Polyphony quite hard (and therefore the Output as well) ... So I want to figure out now, while I have some time to research/plan it a bit, what is the best potential design/setup for doing this..!?
I want to keep it basic/simple. For the most part, I likely won't use other types of ARPs, except just for having an audition/song play on parts 9-16 on the onboard sequencer and route them through the Assign L&R outputs. And play everything else "live" on the keyboard from parts 1-8 routing them through the Main L&R outputs. And keep everything in stereo.

So with this configuration, is there a way that I can avoid having all 16 sounds pass through the same internal output point/location on the internal circuitry of the Montage at any given point in time?

Short answer: If you assign some of the parts to the assignable outputs and some others to the main outputs they do not pass through a single "circuit" at any point.

Much longer answer: The name "circuit" might be pointing to some misunderstanding: there are not that many circuits in the sense of analog circuits in the Montage. Certainly none of the effects, EQ etc. are circuits in that sense. It's all software. From the generation of the waveforms through the amp envelope, through the filters through the effects, through the mixer, that's all just bytes passing through a computer. The level meters you see are just software as well. Only at the very last stage those bytes are converted to analog signals. The point where the signals are converted to sound are the D/A converters. Only after that point it is the analog circuitry. The main and assignable outputs need to have independent D/A converters and the analog circuitry after that should also be independent. I believe that getting those analog circuits to clip is probably not possible or at least extremely hard. It's easier to drive the mixer or amplifier behind the Montage into distortion and clipping. The clipping I have heard (and you almost need to produce it on purpose) was always on the digital side - evident because it was visible in the level meters. Or in the mixer behind the Montage...

So to make the first statement above more precise: The values going to main and assignable outputs are never added inside the computer and they do not go through the same D/A converters and analog circuitry, so they cannot produce any clipping together.

But:

You get clipping inside the Montage's computer if the computed values ever reach a level where they exceed the value which can be represented in the numerical representation Yamaha have chosen. I believe that's 24 bits, so that gives you a lot of headroom. But you can reach those limits. I have done that with a single element named "Case 78 hard" by setting its volume to 127 and setting the performance volume to 127. Then I can clearly see in the meter that it is in the clipping area. And I can hear it. That's unavoidable unless Yamaha would not use all those 24 bits and would limit the levels artificially. Which would not be good from my point of view because it limits the creative possibilities.

You can also get clipping if you boost the output levels by setting the output gains to higher values. You can also get clipping if you drive the computed values very high by using EQs and some of the effects which offer a gain. So you can do it with as few as one element. But it is relatively unlikely unless you do something very extreme. But I have never had the effect that multiple parts together would produce a clipping when the single parts where safely below the clipping levels. As Jason suggests, 127 for everything might not be a good idea.

So to avoid clipping I suggest keep the volume of the the parts and the performance normally at or below 100 and keep the gain at 0 db. I have never ever heard any clipping no matter how voices are playing at the same time if I stick to that. And I used a sequencer to drive all 16 channels at the same time...

In summary: don't worry. Even if you go through the same outputs with everything it is easy to avoid any clipping.

Short answer: If you assign some of the parts to the assignable outputs and some others to the main outputs they do not pass through a single "circuit" at any point.
So to avoid clipping I suggest keep the volume of the the parts and the performance normally at or below 100 and keep the gain at 0 db. I have never ever heard any clipping no matter how voices are playing at the same time if I stick to that. And I used a sequencer to drive all 16 channels at the same time...
In summary: don't worry. Even if you go through the same outputs with everything it is easy to avoid any clipping.

Sounds like good sound advice to me! Pun intended 🙂
I will definitely keep levels in check and keep them close to what is already there in the preset performances, as I suspect that Yamaha set them that way knowing the best volume levels that should be used on parts & performances, as well as the gain.

It's easier to drive the mixer or amplifier behind the Montage into distortion and clipping. The clipping I have heard (and you almost need to produce it on purpose) was always on the digital side - evident because it was visible in the level meters. Or in the mixer behind the Montage...

So basically if I understand you correctly, you can 'see' the clipping in the level meters &/or in the mixer, all on the software side of the Synth (as well as 'hear' it at the same time)..!?

Is there a separate Meter &/or Mixer for all parts that are routed through to the "Assign L&R" outputs?

You can also get clipping if you drive the computed values very high by using EQs and some of the effects which offer a gain.

I always tend not to drive effects gains higher. If there is a frequency that needs boosting, I will usually boost it very slightly and also lower all the frequencies around it using high pass filtering on the low end and low pass on the high end frequencies...

It would be awesome if Yamaha gave us a new software based Loudness Maximizer effect for the final mixer effects, like the Wave's L2 or L3 Maximizer for Protools (there is also a hardware version of the L2 that utilizes this software based tool). That would allow people to drive their software side of the Montage/MODX much harder than ever before, but without any clipping/distortion! I might just add that one to IdealScale 😉

The master level meter you see when you are in the mixing screen. If it is in the red then you get clipping. Similar for the individual parts which you also see in the mixing screen. I believe there is no meter for the assignable outputs. At least I did not find it.

I think the important point is that it is not necessary to drive the software side that hard. Question: Why do you drive the level in analog systems as high as you can, just staying below the level where you get distortion and clipping? I know of two reasons: Noise and interference from other signals. Because we are dealing just with numbers in a computer here, there cannot be interference from other signals (unless there is some bug in the program 🙂 ). The second one is the noise. Noise can be there through rounding errors when doing the computations. Because the accuracy of computations is limited, the results for example for a division or multiplication by a fractional number can be off by a bit. That will result in some noise. In older synthesizers the computation resolution was for example 12 bits, resulting in quite some noise. If it is in contrast 24 bit (which I believe is the case for the Montage), the rounding errors are much smaller and thus the noise is much lower. Which means that you can live with lower signal levels in the digital path. Cubase has 64 bits by now I believe which is even less noise... but 24 bits gives you 256 times the resolution of a CD player (note that the resolution of the D/A converters and the resolution of the calculation does not have to be the same, you might want more resolution for the computation to avoid rounding errors).

The master level meter you see when you are in the mixing screen. If it is in the red then you get clipping. Similar for the individual parts which you also see in the mixing screen. I believe there is no meter for the assignable outputs. At least I did not find it.

Yeah, based on what you and Jason have said, looks like I will have to check the "Assign L&R" output levels by ear to ensure no clipping. But if I keep all those parts relatively the same as the parts for the "Main L&R" output, and they are at or below 100, keeping the gain at 0 db, I should be good.

There's potential for another future enhancement to throw up on Ideascale: "Master level meter for the assignable outputs, as well as the ability to add effects there as well"

Separating your outputs between Assign L&R and Main L&R gives separation between the two sets of PARTs (with consequence for Assign L&R as described through loss of Main/System Effects). Assuming loss of effects on Assign L&R is OK - then you will maintain the additional "headroom" or lack of sonic clutter if Assign L&R has a clear signal path all the way from the outputs to its own set of speakers. The same for Main L&R.

If, at some point, you use a mixer to get the signals back together to drive out of two speakers - or the same sets of speakers - then you're back to having competition. Maybe the mixer has a bigger pipe - maybe not - but once running through the same sets of speakers - the result will be much the same as if you didn't split to begin with.

There may be other considerations as well such as phase cancellation depending on the sounds you have going through each pipe.

Much of this is better heard in practice than discussed in theory. As far as recommendations for your setup.

There are routing capabilities. Assign L&R is an output choice with the stated consequences on effect capabilities. Distortion is not generally a problem discussed. When discussed, it is usually a theoretical discussion and not a complaint.

Searching for montage + distortion on this site, I find:

Theoretical:
https://yamahasynth.com/ask-a-question/output-volume

This thread is not Montage - MODX. Which states that Montage does not distort where MODX does (for the same user Performance):
https://www.yamahasynth.com/ask-a-question/a-modx-companion-for-your-montage

Diagnosed as not a clipping issue:
https://www.yamahasynth.com/ask-a-question/montage-sound-clipping-issue

Related to matching Montage to an input:
https://www.yamahasynth.com/ask-a-question/monitor-speakers-master-volume

That's just about it. Lots of hits to sort through because "distortion guitar" comes up as a desired discussion about distortion so I filter those out.

Point reinforced is that from all the varied usage and experience in the forum - problems of output distortion are very difficult to find. The most common audio distortion topic is in properly matching the output signal chain (speakers, speaker amp) to Montage. Which always seems to be ultimately resolved and primarily from "don't turn it up to 11".

Separating your outputs between Assign L&R and Main L&R gives separation between the two sets of PARTs (with consequence for Assign L&R as described through loss of Main/System Effects). Assuming loss of effects on Assign L&R is OK - then you will maintain the additional "headroom" or lack of sonic clutter if Assign L&R has a clear signal path all the way from the outputs to its own set of speakers. The same for Main L&R.

If, at some point, you use a mixer to get the signals back together to drive out of two speakers - or the same sets of speakers - then you're back to having competition. Maybe the mixer has a bigger pipe - maybe not - but once running through the same sets of speakers - the result will be much the same as if you didn't split to begin with.

Point reinforced is that from all the varied usage and experience in the forum - problems of output distortion are very difficult to find. The most common audio distortion topic is in properly matching the output signal chain (speakers, speaker amp) to Montage. Which always seems to be ultimately resolved and primarily from "don't turn it up to 11".

Question answered and I like the answers/possibilities. It's what I hoped was the case!
I'm envisioning a small mixer board since I will only need 2 x Stereo inputs from the Montage, 1 input for a vocal mic, and possibly 1 stereo channel for playing songs in-between sets at a gig. Although, I have access to an iPad that I could just play songs off of from, connected to the Montage USB port..!? 😉

Thanks,
Darryl

iPad connected to Montage via USB using the camera kit would allow for your iPad to send audio through Montage's digital input out to either Assign L&R or Main L&R.

And while you are at it, maybe input the microphone to the Montage A/D inputs?!? Then you have nice effects and EQs with every performance just for the microphone. And if you can route everything through the main outs without getting distortion - which I am sure you can - you can get away completely without the mixer. Simplifies the setup...

iPad connected to Montage via USB using the camera kit would allow for your iPad to send audio through Montage's digital input out to either Assign L&R or Main L&R.

That's exactly what I'll do then ... Thanks!

And while you are at it, maybe input the microphone to the Montage A/D inputs?!? Then you have nice effects and EQs with every performance just for the microphone. And if you can route everything through the main outs without getting distortion - which I am sure you can - you can get away completely without the mixer. Simplifies the setup...

Wow! Food for thought...the thought of not having to take a mixing board and keep it even simpler is very appealing. I'll definitely be trying out that configuration!! Thanks!