DunedinDragon

That really opens up more questions than anything. Going into the main front input on your Marshall you were getting quite a bit of help from the Marshall pre amp and amp tone stack, so it really begs the question of whether you used the Helix pre-amp block or the full amp block. It also would likely require you to change the way the Helix amp model was dialed in. I would think you'd need to dramatically change the setup (tone knobs, gain, master volume, channel volume, etc) of the Helix amp block once you go into the effects return given you're no longer getting the benefit of the Marshall pre amp and tone stack and most likely you'd be better off with the Helix amp block than the Helix pre amp block. Since you're now going through only the clean Marshall amplifier circuit, you'd also likely carry the Marshall volume louder and simply control that overall volume using the Helix master volume control.

Just to clarify, if you go into the effects return on the Marshall you would be bypassing the tone stack of the Marshall so your Marshall tone knobs wouldn't do anything . You'd be controlling the modeled amp on the Helix with it's tone knobs. I believe all codamedia is referring to is to make sure your mid tone knob and treble tone knobs are fairly close rather than have your mid tone knob turned down too much and get too much high frequency feedback.

Ideally you should be plugging the Helix output into the effects return of the Marshall which will then use only the power amp portion of the amp allowing you to get the clean and accurate modeling output of the Fender Twin through the Marshall power amp and cabinet. Because you're using a physical cabinet you wouldn't typically use the modeled cabs or any IRs in your signal chain.

I can only go on what I was objectively measuring on the signal meters of the mixing board in that regard which measured a much greater range of response in the peaks above the average signal at the lower channel volume/higher Helix volume than at the higher channel volume level/lower Helix volume. In some cases I had the compressor after the amp which registered smaller differences (as would be expected) than patches with the compressor at the beginning of the signal chain. However, even with the compressor after the amp I needed to significantly increase the peak reduction on the LA compressor to match the response I was getting at the higher channel volume/lower Helix volume with the same patch layout. I'm not sure how else to read those results unless it relates to the behaviors of the higher Helix volume...which could also be possible.

There's no doubt that could be true. However both the drummer and one of the singers in the band asked me if I'd changed something this weekend when we played. They couldn't exactly verbalize it to say what they heard differently, but they noticed something...so it's not just me.

Yep...but as I said before, I really haven't needed that much more range in my on stage volume levels in any of the situations I've been in, so I feel pretty comfortable with it. Since my output to the board has always been at full volume from the Helix due to the fact that I've disengaged the Helix volume from the XLR outputs, that hasn't changed at all.

I always place mine in more of a traditional backline arrangement on a half-height pole behind me (roughly chest high) and a bit to my right or left and pointed in such a way as to get good coverage across the stage for the other members of the band. This is the same whether the instruments go through the PA or not. The only thing that changes if the instruments aren't going through the PA is my Helix volume gets turned up a bit to get the sound out into the audience.

It depends on the amp model as they're all different. For some like the Fender Twin model they channel volume was up around 9 or even 9.3 and went down to around 5.5 or 6. For the Placater it went down from around 7 to 7.5 to between 4 or 5. All of them seemed to respond in a more linear fashion relative to the increase in the Helix volume.

There's really no need to do that. By turning up the Helix volume and lowering the amp model channel volume I end up at the same output signal level as I had before, so there's no change necessary on the speaker volume.

As a quick follow-up on this effort I wanted to mention something that I've seen much more consistently ever since I increased the Helix volume level. I previously mentioned there appeared to be more dynamics in play within the converted patches and now that I've done a few more it's very clear that's the case. Because I measure the signal levels coming out of the 1/4" output as well as through the XLR outputs it's very clear to me that the range of touch dynamics is MUCH more prominent than what I originally thought. It's most evident with cleaner patches, and particularly prominent when finger picking on my hollow body gretsch. But it's also quite a bit more prominent even on heavier rock style patches with a pick on a Les Paul, both in terms of how the pick is used, but also where on the strings the picking is done (toward the neck or toward the bridge). I'm not sure I can explain it but the measurements I've taken are very much different between my old patches at the lower Helix volume and the newer patches at the higher volume, as well as subjectively to my ears. I can only guess that it may have to do with the way the amp circuit is modeled to respond at higher channel volumes versus lower channel volumes, but who knows? It could also have to do with the bit depth as has been postulated by some. The amps I've seen this on are the Placater, the Fender Deluxe, the Marshall JTM-45, and the Vox AC-30. For myself this is a big benefit as I do a lot with dynamics in my playing, but YMMV....

Probably you're simply going to have to put up with some of the cable switching for the time being given what you're describing. You might be able to save a little bit of complexity by just connecting the Helix direct to the PC so you can play tracks from any source whether it's Cakewalk or YouTube. It does get a little more hairy if you're wanting to record the jam session in which case you're probably going to have to do some cable switching and use the 2i2, or you might just be better of in that case using the 8 track recorder to capture the session rather than Cakewalk. That's just going to be the situation until you can get over to a better system. Be patient...we've all been there at one time or another. If you think this is bad, you should see some of the crap we used to kludge together back in the 60's.

Yeah, that's much more appropriate for what you're wanting to accomplish. I've owned a couple of Zoom products over the years and they're pretty decent. Another consideration you might want to look at in that same (or even less) price range that's one of the most popular solutions for what you want to do are the Behringer X Air series mixers if you don't mind using a tablet or pc as your physical mixing board rather than a hardware mixing board. In live performance this allows you to leave the main wireless unit on stage and then manage your mix wirelessly from a tablet either on stage or out in the audience. These are probably the most popular and widely selling units in this space. The XR18 which is an 18 channel unit which is the one reviewed in the attached link, prices in at 499.99. A little less for the units with fewer channels. Of course you'd need a tablet (android or ios) or Mac or PC if you don't have one. Behringer XR18 Product Walkthrough Video Personally, if I were in your situation I'd probably opt for standard PA type speakers maybe like the QSC CP12 or Yamaha DBR12's simply because they'd work better with a jam session than would studio monitors, plus be a viable solution if you wanted to play live.

I think you're letting your gear drive your approach to a problem rather than your problem defining the solution you need. There is no simpler way in the world nowadays to accomplish what you're wanting to do as far as jamming and capturing the jam session with the technology that's available now. But if you try and kludge a bunch of stuff together you'll have a constant battle of getting it to do what you want it to do. The simple approach is get a digital mixer which will have all the capabilities you need to capture the performance in a multitrack session on a simple digital disk drive. Import that into your Cakewalk and mix and polish it to your heart's delight. Run your backing track through the stereo channels of the digital mixer and jam with them, capture the jam session and import it. All you need is a couple of monitors and you're good to go. If you want two guitars going through the Helix, follow the instructions laid out. If you want to add drums, plug them into the mixer. Same with bass, same with keyboard. The session will be captured on the disk drive. It's even possible with some digital mixers to direct the output directly into your Cakewalk. The answer you want isn't in the Helix because that's not what the Helix was designed to do. It's a guitar modeler and processor. It's a good piece of the puzzle, but it's not the whole solution. But the simple solution is out there and it's not complex at all. It's a turnkey system nowadays.

As cruisinon2 says, sell the interface, the mixer, the USB hub, and the katana and buy a good pair of studio monitors. Hook the Helix up to the PC direct into one USB port and connect everything through the Helix through the different inputs or use the returns for things like the drums. Create different patches for each instrument. When you want to record an instrument, select the patch for it, enable the track in Cakewalk, and press record. That's pretty much exactly how I have my Helix set up in my studio.

It may have but he really didn't go into the setup of the output with the Fractal i.e. into 4cm, into FRFR speaker, though IEMs or stage monitors. How exactly did he "match" that sound exactly? Nor was he very specific about how he got the sound. We just have to take his word for it and bow in homage to his expertise. You have to excuse me but I've been around these kind of guys since the '70s so I tend to be a bit skeptical about their broad "expert" statements and their overrating of their ability to distinguish differences due to their "golden ears" that us common folk just simply don't have. The most telling statement was when he gushed all over about the sound of a good tube amp over any modeler as if that's the sound he's presenting to the audience. When most of us know what he's presenting to the audience is a mic capture of that sound, not the tube amp sound, but he assumes we're all to stupid to understand that. And if he loses the modeled sound in the mix there's a big problem with either his skills at the board or his skills at the modeler..or both. My opinion is that since the late 90's and especially over the past few years technology leaps not only in modeling but in many other instruments and technical leaps in sound reinforcement gear have made the "magic" and value they used to provide accessible to everyday musicians, so they're desperate to prove they've still got some value to offer. Case in point, these guys absolutely despise some of the new digital mixers that have advanced features like auto feedback detection and correction, auto room acoustic measurement and EQ setting, auto gain staging, and auto conference volume management. Things that they could only do are now done automatically for anyone that can push a button. Their response is always, "it's not as good as when it's done by trained ear, even if you poor schmucks can't hear the difference".

Quite honestly I've never felt my stage sound was any different from the FOH sound even when my Helix volume was at 11 o'clock. And I'm pretty familiar with it since I'm out front when we do our sound checks helping to gain stage and set mixes for the band using my wireless tablet typically from different areas of the venue. I figure if I'm not going to hear any qualitative difference between my sound through the DXR12 with my Helix volume at 11 o'clock and the sound coming through full signal through the XLR into two stacks of QSC KLA12 line arrays, I doubt anyone will ever hear the difference. However setting my Helix volume knob at 3 o'clock had nothing to do with that. It had to do with retaining the flexibility to adjust stage levels easily with the Helix volume knob while getting a better alignment of my signal level from the Helix with all the other instrument and voice levels of the other channels coming into the mixer...nothing at all to do with tone.

Actually it's no different than what I was doing before. In practice I still have plenty of room to adjust my on stage volume level up or down using the Helix volume knob. I was keeping it at 11 o'clock by the way. Even then I've never encountered any situation that required me to turn the volume knob higher than 12 nor lower than around 9:30. So I still have ample room either way using the Helix volume knob and none of it will affect my signal to the board. Of course I'll have to adjust the individual levels in the patches to match the new settings, but that's not really that big of a deal.

Nor any in depth experience with how to use a modeler.....

My experience after 3 1/2 years of using my Helix both in the studio and live performance is exactly the opposite. I use a Yamaha DXR12 for live performances and Yamaha HS7 studio monitors. I design all of my patches using the DXR12, and occasionally bring those same patches over into the studio. There may be a few very minor tweaks, but generally they all stay intact going from one environment to the other. In fact we've make several multichannel recordings direct from the mixing board in live performances that needed no adjustment at all. What this tells me is that you may simply be experiencing the differences between PA speakers and studio monitors which are very different by design. I've never used an L3M but I do know there are different DSP settings that can be used depending on how you're using it or have it placed. Some of this could be not having the L3M settings correct for how your using it. However I suspect it has more to do with the way one listens to studio monitors and the way one listens to live music. I doubt very seriously you've ever listened to a live performance positioned right in front of a PA speaker like you would a studio monitor. The sound you hear in close proximity to a PA speaker is likely to be much more harsh than if you back away from it and give it some room. That's normal because PA speakers are built for projection, which studio monitors are not. When I'm setting up my patches for live use with my DXR12, I generally play them with my DXR12 at about torso height and I listen to them from different points in the room maybe 5 or 8 feet away, sometimes even turned to the side or with my back turned in order to get a feel for how they're going to sound on stage. When I listen to those same patches through my studio monitors I'm sitting in an equilateral triangle of around 3 feet from each speaker with each of those speakers at roughly just around head height when I sitting down. I think it's a huge error to make the determination that FRFR type speakers won't sound as good live as an amp given that an amp will most likely be miced and will eventually go through FRFR speakers. In fact, I suspect you've never heard an amp in a live concert of any size. You've heard a mic'd amp sent through a FRFR type speaker.

I apologize in advance for the long post, but hopefully there's a few tidbits of info that may be useful to some members of the forum in here. We've seen a number of discussions recently regarding where to set your Helix master volume setting, the pros and cons, and the speculations. I've mentioned several times that I normally have mine set at 11 o'clock when I'm working on and normalizing the volume of my patches, and have had it there with no problems for the last 3 1/2 years. However I have had some concerns about that that particular setting that really doesn't relate to better or worse quality of tone, but revolves more around having a better setup that's more easily managed. This stems from the fact that my rig is setup to use an on stage Yamaha DXR12 as my stage speaker which I position behind me on a half height stand in a normal backline situation, and a separate XLR output that goes direct to the mixing board using an inline phantom power blocker. I've disengaged my Helix master volume from controlling my XLR outputs and have set those XLR outputs to Mic level with my 1/4" output going to the DXR12 at line level and being controlled by the Helix master volume knob. This isolates any on stage volume changes made on the Helix volume knob from impacting the XLR signal level output to the mixer. And although this has worked perfectly in a wide range of settings with different PA's or with just using the DXR12 for instrument and vocals only through a PA, it has concerned me at a couple of levels for a while that made me start considering a change in the Helix volume knob setting. As far as background info, as you may know, when you disengage the Helix volume knob from affecting an XLR or 1/4" out, the Helix defaults to sending a signal that's equivalent to having your Helix volume knob maxed out on that output line. Even sending that XLR signal out at Mic level (which is quite a bit lower than Line level), there's still a considerable difference in the signal strength that goes into the mixing board, causing it to need the gain setting on the channel to be cut down considerably due to the volume levels within my patches which are higher due to having my Helix volume set at 11 o'clock. This isn't a huge issue as it can easily be gain staged at the board to unity, but it could present problems on certain mixing boards that might not have very capable channel preamp setups on their mixer. So by raising the Helix master volume up to 3 o'clock and adjusting my internal preset volumes lower, I can minimize this major disparity between the two signals that could potentially be a problem. Although I might lose some of the overall headroom for making volume adjustments to my on stage setup, I've never had to raise my Helix volume above 12 o'clock. So setting it at 3 o'clock gives me ample room for adjustment, and actually more headroom within my patches for adjusting master and channel volumes on the various amp models. I did this adjustment on six of my presets this morning and all went quickly and very well, although there were a few things I learned in doing it that may be something people need to consider. Although it's generally held that making volume adjustments in your preset using the channel volume of the amp won't affect the overall tone of your patch, that's true except for patches that contain effects after the amp since they're receiving a much lower input signal. Most of my post amp effects tend to be EQ's, compressors, reverbs, delays, twin harmony, and an occasional chorus or phaser, but that's pretty rare. First I didn't notice any kind of quality level difference in the sound by having higher Helix volume setting once I adjusted my amp levels. What was most noticeable were the level of dynamics in picking or plucking the strings harder. Not a huge difference, but noticeable. I suspect this may be attributed to the circuitry modeled in the amps that may act as a limiter if the channel volume is very high. What was most affected was my post amp compressor which is typically the LA Compressor. With the lower amp channel volumes I noticed I needed to raise my Peak Reduction or amount of compression from my normal settings of around 5.5 to up around 6 or 6.5 and that I needed to also raise my compressed/dry mix from around 80 to up around 90. I suspect these are artifacts from the greater dynamic range I got from the amp models when I lowered the channel volumes. Other than that I can't say I ran into any problems with the reverbs/delays or EQs. I wasn't working with any patches that used the Twin Harmony, but that will be the next thing I concentrate on. In the end I was able to accomplish what I set out to accomplish with my disengaged XLR output line signal now falling in line with the rest of the normal XLR inputs from other sources which is exactly what I was after. Hopefully some of this info will be useful to others.

Interestingly enough, I've been experimenting today with adjusting my Helix master volume up from my normal setting of 11 o'clock to 3 o'clock. There's several reasons for doing this which I won't go into now, but it does entail me making fairly significant adjustments to my amp channel volumes to retain the same overall output level. The only impact I've seen on after amp effects seems to be on mostly around the LA Compressor which kind of makes sense in terms of when the compressor will kick in and how it affects my mix between uncompressed and compressed signals. But that was all relatively minor. I saw no effect at all on reverbs, delays or EQ's. I wasn't working on any patches that used the twin harmony which I normally place after the amp, but I may mess with one of those later. All said, there wasn't anything that was really problematic or couldn't be adjusted pretty easily. One thing I did notice is the signal can tend to peak higher in the case of more dynamic playing (harder picking attack) than it would at a higher channel volume levels. Probably due to the modeled circuits in the amp that likely protect or limit such spikes at higher channel volumes. Just interesting points to share I thought. I'll probably end up posting a more extensive stand alone post on some of the reasons why I did this and what lessons I've learned from doing it as I think it provides some good insights into Helix behaviors given all the recent discussions about Helix master volume settings.

Why not just take the time and build your own? It falls under the category of "teach a man to fish...". You could probably do that quicker than trying to find one.

There are several places you can potentially adjust the volume of a patch without affecting the tone. The most common one is the channel volume on the amp model. You could also use the level on the output block.

I think what I might try first would be to place a Parametric EQ at the end of my signal chain that way you can use it both to add volume to the signal as well as tailor your frequencies to dial in the tone. Alternatively you could also use the output block on the signal chain to increase the volume output of the signal chain without adding a block. The gain wizard is different from the feedback wizard as it's used to help in gain staging all the signals to the appropriate levels whereas the feedback wizard is used to help dial out frequencies that are prone to feedback. I'm not sure what you're referring to as the autotone reducer though. I've never heard of it. You're correct that all of these tools should be used at the gig as they're general sound performance related tools applied for gigs.

The key thing to remember with dialing in a preset at low volume is that mids will sound more prominent and lows and highs will be less emphasized. As the volume goes up, the lows and highs become more prominent which can make the mids seem less prominent and things may sound more boomy and harsh. It has less to do with the speaker than it does the simple characteristics of human hearing and is referred to as the "Fletcher Munson effect". Where the speaker comes into play is based on how the speaker is placed. When a speaker is positioned as a floor monitor bass frequencies tend to accumulate due to it's interaction with the floor. This is known as "bass coupling". This is the reason some speakers have a DSP contouring setting that allows the speaker to compensate for such things. Unfortunately the Headrush doesn't have this type of setting. It has two contour settings. One setting accentuates lows and highs, while the other one sets the speaker flat. This is one of the reasons why many Headrush users place the speaker on a stand behind them like a normal backline setup to avoid the bass coupling. The value in the Headrush contour setting is that you can ideally use the setting that accentuates the lows and highs at low volume to give you a better representation of what it will sound like at a higher volume with the contour button dis-engaged. Bottom line, this is one of the reasons you'll hear advice about building your presets at performance level volumes so you're not surprised when you turn up the volume. I don't personally own a Headrush so I can't speak to how accurate the contour button is in helping to compensate for the Fletcher Munson effect. I personally build my presets at performance volume levels for a couple of reasons. One being the Fletcher Munson effect, but the other being that I need to equalize or normalize the different volume levels between all my patches so that they're consistent. One other thing that most of us do is to set our FRFR speakers volume knob at unity (12 o'clock position) so that we can control our volume levels from the Helix master volume knob rather than having to reach behind the speaker to change it's volume. Because Fletcher Munson is an artifact of human hearing, it makes no difference whether the volume is controlled on the speaker or on the Helix, it's just more convenient that way.