rd2rk

The controller doesn't need to be "in sync" for Snapshot changes. The programmed switch will always send what it's programmed to send. If you're depending on an LED on the controller to tell you what Snapshot you're in, then I suppose that would matter, but most controllers are one-way. The MIDI Input is there to allow "chaining" MIDI devices - received messages are passed thru to the OUT/THRU port. It's possible that SOME controllers MIGHT be capable of being controlled by other devices, but the one I suggested does not have that ability. You would need to read the manual for any device that you are considering to verify that it has that feature. I had a quick look at a couple of the Morningstar manuals - MC6 MKII and MC6 PRO - and neither of those allow what you want, MIDI IN is only for passing messages to MIDI OUT/THRU.

I've no clue why you'd want to do that. New to MIDI? The main reason to use the "Snapshot CC Send" would be to record a MIDI track to automate a show. When you use a MIDI Controller for LIVE performance you just setup the controller to send the CC#69+Value (SS# - 0-3). Unless you switch banks on the controller, those switches will always send the proper message. Here's the MIDI controller I use. Cheap, powerful, easy to set up (mobile app), rugged, dependable, and widely available: Amazon.com: HOTONE MIDI Foot Controller Pedal Bluetooth Ampero Control 4 Button : Musical Instruments You didn't say WHERE in Europe, so...

The manual for your XYZ amp specifies that the FX Loop is PQR level. MOST modern amps' FX Loops are made for use with stompboxes/consumer grade devices. If the PQR level is -10db then that's what you've got. SOME amps' FX Loops were once upon a time made for studio/rack FX. If the PQR level is +4db, there ya go. If the amp has a -10db/+4db switch then set it appropriately for your gear.

Yes, that's what I described using the Snapshot Edits setting.

In Global Settings>Preferences>Snapshot Edits can be set to RECALL or DISCARD. RECALL retains the Snapshot changes until you switch Presets. DISCARD reloads the SAVED Snapshot configuration when you switch Snapshots. In either case, the changes are only permanent if you SAVE the Preset.

This is a user-to-user support forum. Nobody from Line6 hangs out here. There was once something called Ideascale where you could post "wish lists", but that no longer exists. Your best bet if you want someone from Line6 to see your request is to go on The Gear Page>Digital and Modeling Forum and start a thread in which you tag @Rikman. He is the Project lead for the Powercab CL.

This is not something that I do (use MIDI Clock), so the following could be all wrong, but until someone else chimes in it's something to try. I don't think that you can CHAIN tempo between devices, and I think the sync would drift over time even if you could. If your DAW is the MASTER CLOCK, I THINK that it needs to control each device directly. Try connecting all three to your computer via USB. Configure them all to receive MIDI Clock over USB and configure Logic to send MIDI Clock to all three. I also do not use Logic so I've no idea how to setup Logic to do that. In Reaper the setting is right there on each MIDI Device Output.

You are not switching analog channels, you are switching PRESETS. It takes time to UNLOAD one preset from memory, then load another into memory. It's nowhere near 1/2 second, MAYBE 100ms, if that. More like 30-50ms. You either learn to work around it (don't switch presets with a note ringing, pause for a dotted 1/8th) or find a product that doesn't do that. I don't know of one. The BOSS GT1000 is designed to work like that, but the Katana does not (AFAIK) contain that technology.

Looks like you did a cut/paste from an app with a black background - nothing is visible! The bit that was readable in the notification email said: Hi all. I have the itch to run stereo again and I’m hitting a mental block on designing a preset that can be stereo for both XLR and 1/4 (without IRs/. Create your split using a stereo block - I usually use a GAIN Block. The split itself results in two stereo paths. If you want IRs on the XLR path (usual practice for FOH sends), use the DUAL IR Block and Pan the IRs to taste. To complete the split to stereo 1/4" and Stereo XLR outputs, pull the MERGE Block down and select the output types on the Output Block. Refer to the manual if you don't know how to do these things.

I would go for Option 2. Things you may not have thought of: External drive pedals in the HXFX Loop could be too noisy (IMO) due to the small amount of noise added by the Loop. With "Lots of dirt pedals" you'll already have plenty of noise. YMMV depending upon the pedals, but that has been my experience. If you haven't spent much time with the HXFX dirt pedals, try them and compare. You might end up selling the physical pedals. Option 2 also allows you to put your MOD FX before the Opus and your time-based FX after, as would be done in a studio in post production. There's no reason to further complicate your signal chain with external MOD pedals, the ones in the HXFX are quite capable. Also, keep in mind that every time you go from analog to digital and back you're introducing latency which, even if not enough to throw off your timing, can still affect the "feel" of your rig. More external pedals=more cables=more potential failure points. "KISS" theory applies.

No. Nobody from Line6 hangs out here. Go to The Gear Page>Digital and Modeling forum and search for @Digital Igloo (Eric Klein). He knows. Trust him.

Just for my personal info, what method is the PV using that makes it incompatible? It appears to be simple TRS switching, as I don't see any voltage supply to the FS which uses a single TRS cable...

If you have the PV FS you can test what's happening with a multimeter or continuity tester. If you don't have that, I had a quick look at the manual and it appears to me that there's separate switch jacks for the Channel/Boost functions and the Reverb/Tremolo functions. Since the required PV FS has only ONE TRS jack, that would seem to indicate that you'd need TWO footswitches to control all functions, but that would be weird. If I were you, I would contact PV Support and see what they say about it. If you like the PV, the Marshall will be a very different sounding amp.

It's analog. All that's happening is that a circuit is either open or closed. IDK which is which on that amp, you'll need to experiment a bit, but let's say that TIP is CHANNEL, OPEN is CLEAN and CLOSED is DIRT. Set the IC on the CLEAN Snapshots to EXT AMP>NONE and the DIRT Snapshots to EXT AMP>TIP. The Channel on the Marshall will remain in the selected status (OPEN/CLEAN or CLOSED/DIRT) until you select a Snapshot that changes the OPEN/CLOSED status -NONE=OPEN/CLEAN or TIP=CLOSED/DIRT. Again, you'll need to experiment to figure it out. If you have the actual Marshall Pedal you can use a multimeter/continuity tester to see if the circuit is open or closed when the button is activated.

Each Snapshot has 8 "Instant Commands" (IC) which are sent on Preset load and Snapshot (SS) load. Assuming that your DEFAULT Snapshot (the one active when the Preset is SAVED) is SS1, when the Preset loads all configured Instant Commands in SS1 will be sent when the Preset loads. NOTE - All 8 Instant Commands BEING USED need to be configured in all Snapshots, else they will send the default value for that command type. IOW - If you are using IC1 to send command type EXT AMP, IC1 must be configured to send the desired EXT AMP command is each Snapshot. When you load a different Snapshot, any Instant Commands (IC) whose values have changed will be sent. If the values have NOT changed they won't be re-sent. You only need one IC for your EXT AMP command, so say you use IC1 for that in all Snapshots. If IC1 in SS1-SS4 is set to switch to the dirt channel and you are only switching between the dirty sounds, the EXT AMP command will only be sent once. If IC1 in SS5-SS8 is set to select the clean channel, that command will only be sent when you select one of those Snapshots and IC1 will not be sent again until you again select SS1-SS4. In this way you can toggle your amp's channels on a per Snapshot basis.

I thought that looked familiar.

Perform a Factory Reset. If that doesn't work, re-install the FW. Don't forget to do a backup before either operation. If neither works, contact support.

Your physical Helix has Auto Impedance, without which certain FX such as Fuzz could sound different. Latency could be an issue depending upon the power of your computer. Getting the Input Levels right could affect Native's sound. Quality of your interface could affect the sound. If you're using your Helix as interface, be aware that it's not the best quality interface out there. In the end, the final result depends upon the quality of your playback system. All else accounted for, Helix and Native should sound the same through the same playback system.

Just because you have a YT Channel doesn't mean that you know what you're doing. You can't actually HEAR a computer. You can't actually HEAR Helix Native or, for that matter Helix or any other modeler OR your "REAL" amplifier. The "chunkiness" of a "real" amp is largely a property of the amp's cabinet and speaker. Route your MOTU's output through the RETURN of a real guitar amp and speaker and it will suddenly be chunkier. EQ the crap out of the signal you send to your 3" studio monitors and it will sound exactly like your 3" studio monitors. Helix Native sounds just like Helix. Both sound like the playback system which provides the actual SOUND to your ears. My 6" Rokit studio monitors sound very different than my HR FRFR112 or my Catalyst100 or my Powercab PC212+. Death metal palm mutes sound very different when you play through a Fender Twin vs a Dual Rectifier and 4x12 cab. Finally - "TONE" is subjective. "Chunky" is a perception. 65db bedroom levels thru studio monitors will NEVER be as chunky as 105db thru the aforementioned Dual Rectifier and 4x12. Google "Fletcher-Munson". Have fun looking for a playback system that appeals to YOUR ears in YOUR playing environment! ;-)

The "ideal" setting is MAX***. At MAX (full clockwise) it is UNITY GAIN. That means 0db. Counterclockwise it attenuates the signal. The effect loop of most modern amps is either -10db (for STOMPBOXES) by default or there's a switch to allow for +4db (for RACK FX). If you're getting a huge volume jump when you switch amp channels (or switch from the amp's pre to a Helix amp) you can assign whatever switch you're using (or SNAPSHOTS) to send that message to also adjust the OUTPUT BLOCK LEVEL or the levels at the FX LOOP BLOCK. ***IIRC, the BIG KNOB adjusts the output level in the digital realm, it is NOT an analog control. That means that there is a possibility that it COULD affect TONE as a result of some complicated electronic mumbo-jumbo involving bits. There's an explanation I read on some high-end stereo site if you search for it. In reality, most humans can't hear it. Many people are quite happy using it as an easily reachable volume control. My thoughts are that at a real-world gig, if you're using a cable (vs wireless), the possibility of dragging the cable over the knob and thus changing the setting and pissing off the FOH tech or just causing your solo to disappear suddenly or blow everybody's head off is not worth the convenience of using the BIG KNOB. I disable it completely, which leaves it at MAX (UNITY/0db). At home I use it for a volume control.

I also have the Xvive U2. It sucks but does not have the problem you're describing. It is VERY sensitive to positioning in relation to anything else electronic that might be in the same room (or building or county). Also temperature, humidity, barometric pressure, phase of the moon... It doesn't work any better or worse when connected to my Helix than direct to the amp. I would look around for anything that might possibly be interfering with its signal (fish tank?). FWIW - I have another 2.4 unit, as well as a cheap UHF (excessive compression) and a 5.8. The 5.8 is the least troublesome and was the cheapest wireless on Amazon ($30 vs $150 for the Xvive 5.8). Go figure! The common problem with all of them is battery life. When fully charged they work OK but as soon as they get a bit discharged they start to have dropouts. I HATE grabbing my guitar in a fit of inspiration and having it ruined by dropouts!!! I'd rather trip on cables. Someday when I hit the lottery I'll buy a PRO system, but until then I'll stay wired.

If all you're doing is playing through headphones then the only setting that matters is "Guitar In Pad". If the signal from your guitar is too hot, turn it ON. If you're using external FX in the Loops then you probably want the SEND/RETURN you're using set to INSTRUMENT for Stompboxes and LINE for Rack Gear if there's no -10db/+4db switch on the device. -10db= stompbox/INSTRUMENT.

I didn't watch the video, but I see no advantage to putting the HXS in the HXFX Loop. I call clickbait. For that matter, I see little advantage to using HXS+HXFX vs Helix Floor/LT except a small size advantage and, if you're using an EXP Pedal, that's minimal. Helix Floor/LT - ONE (AD/DA(IN)-DA/AD(OUT)) conversion avoids all extra latency and potential tone loss. Also has more external routing options. Helix Floor/LT/HXS have auto impedance - necessary for authentic tone from certain FX, irrelevant if you're going wireless. HXS>HXFX = 2 AD/DA-DA/AD conversions vs 4 conversions using 4cm = less latency and less tone loss vs 4cm. FX Loop can add a small bit of noise. Saves a Block on the HXS vs 4cm. Less cables = less potential failure points, quicker setup at the gig. Finally, you don't gain anything in DSP vs using both paths connected in serial on Helix/LT. Get a Helix Floor/LT.

The first problem I see with this is that you're re-amping a track that was recorded off of your guitar amp's speaker. That's NOT actually a DI Track. You COULD use the HXS to add POST effects, such as would be done in a studio, but you're MUCH better off using some of the many plugins available to you either IN LOGIC or on the aftermarket. Your problem is that both your VOLT and HXS have very minimal I/O and you don't own HELIX NATIVE. If you bought your HXS NEW you might consider using your discount to buy NATIVE, which would allow you to do the re-amping completely within LOGIC and not have the hassle of re-amping through the HXS itself. But if you can't do that: I assume that you're using a MAC. Create an AGREGATE DEVICE: Create an Aggregate Device to combine multiple audio devices - Apple Support I am NOT a MAC person, but my understanding is that this will allow you to use all of the available I/O of both your VOLT and HXS at the same time. You should now be able to figure out the necessary routing to accomplish recording your mic'd amp (BLANK HXS preset with LEFT MAIN OUT>Volt Input 1) alongside a true DI track (HXS RIGHT MAIN OUT>Volt Input 2) and monitor the recorded project mix through the VOLT while re-amping the DI Track through the HXS. Hopefully, someone else who is actually a MAC person will chime in here to verify this method or have a better solution. Good Luck!

Not sure I understand. Using an IR loaded on the PC112+ it sounds great through the PC112+, but taking the XLR out to the PA causes the PC112+ to sound horrible?