waymda

This path is likely your problem. I do this as Guitar >> Helix >> Mixer (XLR) Laptop (with audio) >> Mixer (USB) Mixer >> Heaphones Set the mixer as the USB playback device in Windows, and trim inputs from the Laptop and use the faders to set the over relative volume - then use Patch changes to adjust the Helix output levels. I normally do this with both my mixer and Helix connected via USB but the Helix USB only being used for edit. This approach allows both the channel input levels to be monitored and the relativities of patches to pre-rec music - noting that pre-rec audio may vary in loudness as well.

I'm not near my gear atm, and haven't run speaker models for quite a while. However, when I did I ran through the multi output. Worth a try?

Does the label say dryclean only? But, no I don't know for sure and would be concerned about damaging my washing machine with its size and external zips, along with how the internal padding would come out the other side once it got wet.

If I read this correctly, you're expecting the Helix to change to the USB input mode, AND retain the settings for the Multi/Helix input mode for when you switch back. Whilst I understand why you want to do this I can't think of a way the Helix would achieve this, short of retaining parameter settings independent of selected input choice, which would require some sort of input mode related memory provisioning and management. I suspect its dropping all parameters not relevant to the selected mode (ie Variax model settings) as it no longer 'needs' or can use those parameters. I would also expect that the output block behaves the same way with any patch specific settings for PowerCab and DT amps when USB is selected as well as the Multi block output levels. This would explain why the other snapshot parameters remain in place - nothing has changed to wipe them. Think of these blocks as behaving like any other block and the input/output sources as being a new model dropped into the block. To do what you want, Multi input would accept any of the USB inputs (not just 1/2) as well as guitar/variax, and Multi output would allow selection of USB 7/8 within the DAW, but it doesn't look like its engineered that way. This might be worth putting onto IdeaScale, after checking it hasn't been done already.

Well yes and no, as you'll need to consider stage real estate, not getting in peoples way etc. Having undertaken a long journey of using IEMs and other monitoring since using a Pod HD500x, my experience is: all monitoring is a compromise adding ambient stage sound to direct IEM mix is much harder to control/manage than you expect - and can have perverse results (for example being flooded by bass through conduction) that are really hard to track down. Its just not as obvious as you might think when you're sound checking etc I used stereo condensers for a while, and franky they made it more problematic than not having them Good IEMs make all the difference, and the amount of bleed you want/need varies from stage to stage (event) I have good molded 64 ears and the Sensaphonics and each is different (not better just different) and I vary what I use depending on the event controlling my IEM mix has proven more important that adding ambience to get the sweet spot I want/need But finding what works for you is the real challenge.

That sounds like a recipe for a feedback loop. Bass >> HX >> mains to Desk >> Desk FB to HX in >> HX mains..... you can see where that's going. To achieve HX to desk without a loop would require an output pre the return of mix from desk that is routed to your IEMs. I have a Helix but I'm sure there's a way of sorting that routing for the HX at the expense of blocks. TBH - even with the full Helix I run the IEMs from the desk back to a dedicated headphone amp with my guit in the mix. I have a strong preference for separation of hardware functions reducing decencies and complexity, and allowing fallback solutions. But YMMV.

I'm not sure it is. The kicker will be the low range frequencies from the Nord and the bass. Getting them loud and clean takes a lot and most FRFRs and not designed for truly full range. I'd be guessing that's what craps out the QSC10.2. A single 10inch speaker (with horn) can't cover that frequency range AND volume for a sustained period. Guitar targeting FRFRs will likely be worse as they are designed for a narrower band of frequencies. A reasonable compact powered sub with appropriate crossover could give you that range with the addition of one box for keys and/or bass gigs. Alternatively a compact bass combo sitting under the QSC.

Get a set of these and hear the stage sound properly as you move around. https://www.sensaphonics.com/products/3dme-custom-tour-gen2-music-enhancement-iem-system Easily supplements a proper IEM mix (which your suggested solution will not obviate the need for).

Now I'm gonna have to test that. For a while I was having issues with patches before realising I'd dropped mono gain blocks at the end instead of stereo (I run dual amp setups). I don't do it anymore, but now you have me wondering if they were all turned on when they caused the issue.

Following - can you please confirm if you fixed this. I'm still on 3.70, but if what your describing is a bug in 3.71 it would break me (well lots of my patches).

So DC plays back your entitled BS line that demonstrates that you don't read the manual and you want to have a shot. There's no exchange of views to be had on this topic, it's a technical answer to a technical issue on some highly technical hardware. You made your view very clear, it's the manufacturer's problem that you don't understand the product. My view, I wish to exchange, is RTFM so you have at least a minimal understanding of what you're talking about. A quick look at your previous posts suggests doing your own homework is an anathema to you, and you think an end-user forum should be your personalised help desk. Best of luck with that.

An alternate approach to the routing is us a send block before the cab block as a tap for the poweramp/cab. Set it with 100% passthrough so signal still goes the cab block. You could assign a pedal to turn the send off for recording, stopping any signal to the power amp with a single click. It does mean you need to use effects you want in you amp/cab pre-speaker block, but it can also provide a really versatile solution for post-cab-block effects being recorded and not impacting on the amp/cab, which could then be mic'd up and added to the recording/mix. For example room reverb and/or compression. Another suggestion is as you get your routing as you want it, save as a template so you just change component blocks for future patches. Your approach may change over time, but many templates is possible. My standard routing has amps and effects that run before amps on path A, effects that are post amp and/or cabs on path B (A routes to B). This seems to give me the best balance of CPU load across the 2 CPU and allows for complex patches, ymmv.

A thought - if you can establish the optimal input gain for Native you could normalise the recorded dry tracks in the DAW to that level. I'm not Reaper user, but my DAW allows selection of individual tracks and normalisation to predetermined levels.

Maybe - Unity means exactly what you've said no gain added (or removed) the maybe is because different elements of DAWs may add or remove gain at what appears to be a 0 setting. They shouldn't, but....... This might be a useful reference for recording and levels https://mojosarmy.medium.com/recording-and-mixing-levels-demystified-151ec65705fa

Some observations: the I/O with Native vs the Helix as described is fundamentally different so I would expect different sounds the Helix is being fed an unamplified instrument level guitar signal, your DAW is recording a line level signal which is then being 'reamped' this is not the same as recording the dry output from the Helix via USB - those levels are very low and often a cause of consternation for new users hence, the input levels for the hardware and software versions is not the same - at the very least you'd need to match the input levels you mention setting parameters and gain leveling to 0DB, but this has nothing to do with the recorded level - you should be aiming for -13dbu (if not less, many aim for -18dbu@peak) for any digital recording to ensure no digital clipping in the recorded signal - the digital signal can be increased without increasing the noise floor so no need to record hot there are both input and output gain settings on native that are independent of the patches created on the hardware - these will be your friend is there a single simple rule/approach to use across all DAWs and hardware/DA scenarios? No, you're adding complexity/hardware/driver/software layers so the solution will be complex and bespoke.