DunedinDragon

Modeling is modeling...it's math and calculations. Everything beyond that is simply subjective impressions and is based on belief and not fact. The truth is there's probably FAR more difference in the tone of either system based on the output sound device you connect it to. At this level of equipment either of them can do whatever you need it to do once you master it effectively. No one in your audience will ever know the difference between the two, and neither of them is going to turn you into a guitar god all by themselves, so go with whatever makes you most comfortable and get to work with it.

Well there's lots of things going into this performance other than the guitar. It's not a particularly original tone. In fact it's very reminiscent of the older George Benson sound. First things first, you'll need a very accurate playback setup with good studio monitors or a decent powered speaker. If you go cheap on those it will show up because of the accuracy and clean nature of the style of playing. Any of the clean amps would probably be okay. I might choose the Jazz Rivet since it's a very popular amp for this style, and a 1x12 Blue Bell for the same reason. I'd start with a R121 ribbon mic to ensure you get the darkness/fullness and add a 421 dynamic mic if necessary to get the crispness without any of the harshness some of the other dynamic mics would add. I can't really make out what his selection is for his pickups but I suspect it's a combination of both pickups. An important aspect of the tone he gets is his picking style as it is for most jazz guitar players. Look carefully at how he holds his pick. It's never flat against the strings because he holds it at an angle. The makes the picking action very smooth and reduces the pick noise. There's just the very slightest hint of a crunch when he digs in so that may take a while to find the settings for that. That's about everything that hits me when I watch it.

Wow...that's a lot of work. My approach was to buy a GoldTone Banjo. But to each their own.....

That won't surprise anyone as most anyone that's had a Helix for any length of time understands the DSP limitations. But DSP limitations is a natural byproduct of all modelers. That's why the system is designed with a LOT of presets so you can organize things and don't have everything and the kitchen sink crammed into a single preset. The advantage of a large physical pedalboard is that you can built it as big as you want, but you can't engage and adjust several effects with a single button action the same as a Helix snapshot which is always going to be more optimal than physically engaging multiple effect buttons. I'd suggest rethinking your paradigm more along the lines of now you have an almost unlimited number of pedalboards you can use and organize those pedalboards along the lines of what are the demands of the music you play. I'm at the other extreme end where I have a preset for each song, but many organize to have different presets for each TYPE of music they might play. Whatever strategy that best compliments your general type of performance.

I went through this when I got my Helix going back 4 years or so and ended up dumping my interface and just using the Helix as it's so much more functional as far as the workflow. I ended up buying a second Helix floor just for he studio so I could keep my gigging Helix separate and always have a backup if I need it. To me it totally simplified my workflow in the studio by allowing me to keep everything plugged in all the time to that Helix and simply switch to a different preset depending on the instrument/voice I'm using, arm and record. This goes into my main computer which, like yours, is the center of everything I do including networking with the other laptops and devices and so forth. I just have my studio Helix and the studio monitors controlled by a furman power conditioner, so if I'm not doing studio work they're turned off and I only turn them on when I need to use audio for some reason...which honestly outside of studio work is fairly limited.

I think this is an apples to oranges comparison. The samplers used on drums and keyboards is a thousand times more complex than an IR. Suffice it to say I think any of the major companies producing IRs will do a fine job of capturing the IR itself. What varies significantly is how broad their selection of cabinets, mics and mic combination/positions are. At this point Ownhammer seems to be the one that captures most of the attention in the marketplace as they have a fairly broad selection and are easily used, but there are others like Redwirez that are VERY deep in not only the type of speaker cabinets they offer, range of mics, mic positions, but also provide utilities that allow you to combine and build your own IRs by combining fundamental IRs into more complex IRs if you're really into doing that kind of stuff. In a way I guess Toontrack Superior Drummer would compare well with Ownhammer in it's mass appeal and ease of use whereas Redwirez would compare with something like Superior Drummer with it's extensive capabilities for modifying the details of your kit and extensive library of beats, but not as easy to use and deploy.

The easiest way is if you have your Helix connected via USB to you computer and the speakers connected to your Helix, just plug your android into any other USB port on the computer, make sure it's connected for file transfer, select the song frpm the file manager in Windows and play it. The Helix should already be selected as the sound device. Probably easiest to control the playback volume for the music from the android on the phone itself.

I think the verdict is in if you want an alternative based on everybody's feedback here. Control it with your Helix volume knob. That's what it's there for. I can guarantee you it's undetectable unless you have the ears of a bat..on steroids. There are plenty of people here that would bet a lot of money that you couldn't detect the difference in a double blind test, but I doubt you'd want to lose that much money.

I have my Helix setup so that I have the Helix volume knob disconnected from the Helix XLR output and set to Mic level output. This is used to send a full volume Helix signal to the mixing board and I generally set the peaks in my patch (using the channel volume in my patch) as measured by the signal strength on my channel at the mixing board with my gain on the mixing board channel set to 12 o'clock. My 1/4" output is connected to the Helix volume knob and sends a Line level signal to my stage monitor. I adjust that stage level via the Helix volume knob and it doesn't affect my XLR line at all. This completely isolates the actions needed to be taken with my signal at the board by the soundman from the actions I need to make for myself for my on stage sound. I use a Yamaha DXR12 as my stage monitor and my home monitor. My volume on my speakers is set to the normal convention of 12 o'clock and I never touch it. Based on this setup both at home or on stage I generally have my Helix volume knob set at around 11 to 11:30 both at home and on stage. Of course I can turn it down to anything I want but at a certain point Fletcher-Munson effects start making the sound a bit muffled. The net effect of this arrangement is I know my signal to the board is gain staged appropriately so that saves a lot of time in sound checks. My on stage mix is easily adjusted with the Helix volume knob. Full volume gain staged signal to FOH, adjustable signal on stage. Best of both worlds.

I truly wish the Variax could duplicate the real sound of different guitars because that would save me having to carry around different guitars for all my gigs, but it's just not there yet. It's impressive for what it is, but what it can never do is replace the feel of the guitar which really becomes important for certain types of styles. Especially when it comes to finger picking.

Well heck...by the time you've gone through researching how many things were dup'd you could have easily built your own preset....

Yeah...so does the Touchmix, but I just find it easier to capture and import so I don't have to worry about the DAW.

I'm not sure the XR has the same facility, but I'm pretty sure it does. On my QSC Touchmix I just record the whole band to a USB memory stick which captures all the instruments in individual WAV files which can then be individually imported in your DAW to create a standard multitrack recording so it can be mixed, edited and output to a stereo track (minus the guitar track if you choose) from your DAW. It's pretty simple process and I think I heard it's pretty much the same process on the XR. This allows you to capture autonomously from the DAW so it simplifies the capture process.

Chances are this is a result of a low memory condition in older machine with limited processing power. When you say you have no problems like this with anything else you run that doesn't surprise me at all. The VAST majority of end user software requires very little in terms of memory, especially things that are more internet based. It's not a particularly wild guess that the DAW is the most memory and processing intensive app you have running unless you do something like CAD, 3d animation, or video editing which would have similar memory constraints. What happens when you begin running low on memory the computer will swap real memory out to disk to provide more free memory when an app require it. This is referred to as virtual memory. Even if you're not running other end user apps (which helps) you still have background tasks the operating system requires that intermittently require processing and memory so that memory may get swapped out so your DAW has more free memory. But that memory will still have to be swapped in from disk when the background task needs to process something, and that swapped out memory gets swapped in while the DAW memory gets swapped out momentarily This is the reason you don't see the latency until it's been running for a while because the DAWs memory requirements have grown so the operating system is swapping memory in and out to accommodate all the applications (both foreground and service background apps) once you get going and begin using more memory. I would almost bet if you doubled your memory you'd never experience it again. Of course that all depends as well on the processing power of the microprocessor in the machine as well. Faster microprocessors with swap faster than slower ones.

Well the first thing I'd suggest is to read the release notes for the version you're installing because the manual, it appears, is way behind in the way updates are done if what you're saying is true. The process for updating has changed dramatically.

Yes...the main reason for the HiWatt's popularity back in the day is that it was specifically designed to work well with pedals.

You may also want to look in the global input parameters for changes as anything in Globals will be reset with the upgrade.

One of the key amp models I use in a lot of mid-crunch classic rock such as Rolling Stones, Who, Doobie Bros, Joe Walsh, Post Cream Eric Clapton, etc. is the WhoWatt or HiWatt which was a very popular amp during that era and was used at one time or another by most of the big bands. The particular value of the HiWatt was it gave a nice clean but could easily be dirtied up with with gain, but it also took pedals VERY well and worked well with compressors to get that "struggling" crunch often used by the Doobies, Stones, and the Who. Probably something for you to explore.

The general way people have been doing this in the past is to simply use 3 snapshots which make the appropriate changes to the controls on the amp model or engage/disengage some sort of drive in front of the amp, or even engage/disengage a different amp model. Pretty straightforward stuff. Plenty of examples of this over on Jason Sadites YouTube channel for the Helix. As far as classic rock it really depends on what kind of classic rock you're talking about. Led Zep classic rock is very different from Queen classic rock, Rolling Stones classic rock or Doobie Bros classic rock each of which would use different amp models for the right sound.

The Global Ins/Outs provides a way to set your output level to either mic or line on the XLR output and line or instrument on the 1/4" output. You'd need to check what level your Synergy expects. As far as clipping on the new firmware...I haven't had a problem. But then I'm not sure there's a whole lot of folks using a preamp on their outputs from the Helix. Of course anyone that comes out of the Helix direct into a mixing board has the preamp on their mixer channel, but that gets gain staged at the mixer to provide appropriate headroom without clipping.

I would also make sure you have everything set up correctly on your LSR305. If things are sounding weak/thin my first response is how do you have your speakers positioned. LSR305's are rear reflex speakers so they're highly dependent upon being positioned correctly with a rear surface to get the bass frequencies full and correct. Check the manual and see what those speakers expect in terms of signal levels on the 1/4" and XLR inputs and ensure you've matched those signal levels (Line level, Mic level, or Instrument level) on the Helix global ins/outs. Make sure your trims are set flat and your volume is around noon. That should give you a very flat response on your speakers.

I guess it's a cool feature, but I can't say it's a "game changer" as far as what I need out of my Helix.

I personally just use a G10T wireless. Unplug from one and plug into the next and switch patches. Takes less than 10 seconds.

I personally don't use the looper, but I'm pretty sure it's not going to meet your stated needs. From what I gather most people that use a looper use an external one with greater capabilities and is easier to use than what's on the Helix.

The place where most people run into gain staging problems is when they start messing around too much with volume levels in the various blocks. The generally accepted recommendation is to try as much as possible to have the same volume output with any given block turned on or off. Typically the only volume changes I make in a patch are on the amp models channel volume. That allows you to maintain a more consistent output level across all your patches and not have to figure out which of the many volume possibilities you changed in a given patch. For example I have my Master Volume disconnected from the XLR outputs, which is what I use to go to the mixing board. By doing that I'm sending a full strength signal to the board as if I had the Master Volume turned on full. However I also have my XLR outputs set to Mic signal level as that's what's most commonly used when sending a signal to a mixing board and generally the trim/gain on my mixing board channel is pretty consistently set at noon, or slightly less when I'm properly gain staged at the mixer. I send my 1/4" output to my stage monitor at Line signal level. My Master Volume (which does control the 1/4" output volume) is normally set at between 11 to 11:30 and goes through my Yamaha DXR12 with it's gain knob set at 12 o'clock, or halfway which provides ample stage volume in an 8 piece band. With that configuration all the various amp models I use in my patches tend to have their channel volumes set between 4.5 to around 7 at the highest.