DunedinDragon

Welcome to the awesome world of modeling. I think you'll like it here. One thing I would add to this discussion is a bit of context. Rather than regard your POD as a collection of amps, you're probably better off thinking of it as a virtual environment for creating a variety of different signal chains. Yes, amps play a part in that signal chain as do mixers, cabinets, microphones, effects, etc. In effect it's more similar to what a recording engineer does in a studio cabling together different components to capture a specific sound, but in this case you're going to create those configurations in a virtual world of the POD and save them as patches or presets so you can use them over and over again. I only bring this up because I think if you focus too much on the amps you might miss out on many of the creative possibilities of the POD. As a self-described "versatile guitarist" I would think understanding it from this perspective will open up the possibilities of exploring a wider range of sounds applicible to different music styles and artists. That's where the POD really shines!!!

The most straightforward way of doing this is to hook up a decent pair of studio monitor speakers to your POD and select the POD as the default audio device (playback/recording) on your PC. If you leave the POD connected and turned on when using your PC, the POD and it's connected studio monitors with provide for all sound both from the PC as well as the guitar or mic plugged into the POD. Of course if the POD is turned off you won't have sound from the PC.

Well don't give up hope. I've adapted to direct in to the mixer with my HD500X and a Yamaha DXR12 monitor...and our drummer has adapted to a V-Drum kit, so there's always hope. As I said, I doubt seriously it had much to do with your subs and more to do with your configuration on the GT100. The biggest thing I had to adapt to moving from a traditional amp to this setup was the wider range of frequencies the new setup exposed which caused my patches to sound boomy on the low end and harsh on the upper end. It made sense when I thought about how limited traditional guitar amp speaker cabinets are in that regard. It's exacerbated by the fact that most of the state of the art speaker systems such as yours use drivers and DSP processors in the speakers themselves to widen and flatten out the speaker's response across the frequency range. This results in very responsive highs and lows which is great for the overall sound of the band, but that means that you have to take charge of the frequency range where your guitar will play. There are lots of various ways to do this on the HD500X. I took the easiest route using the Global EQ so it would apply across all my patches as I described in my original response and it works fine for me.

When I see conversations like this going on I'm eternally grateful that I went with a FRFR speaker setup when I moved to the HD500X. I can see now how it made my life a whole lot simpler.

I'll cast another vote for the Vetta comp. I use it extensively in my clean settings sometimes before the amp if I want more articulation (as in country or rockabilly) and after the amp for a smoother "limiter" effect for things like jazz. Sensitivity of around 30 and output around 50 works for me.

Your use of the term "bass bins" is throwing me off a bit as I'm not sure if you're referring to a normal subwoofer arrangement or not. But as an example, in our case we have QSC KW181's as our subwoofers and QSC KLA12's as our main arrays. The KW181's have a fixed high-pass filter of 100hz. Even if we don't have the hi-cut filter engaged on the mixing board channels for the guitar, little to no guitar goes through the subwoofers. The fact is in a true subwoofer arrangement you really don't "hear" much out of them at all in a traditional sense. You feel them more than hear them in an omnidirectional pattern. Often times during soundchecks we have to wait for the bass and drummer to show up to ensure the subwoofers are working correctly. At 100hz you don't really get a feel for the tonal aspects of the bass, that's provided more by the main KLA12's, but you will sense the lower end harmonics or punch of the bass through the KW181. Likewise you'll feel the mostly the punch of the kick drum. So generally speaking the subwoofers really have little to do with guitars. What is an unpleasant artifact of guitars occurs generally right around that 100hz cutoff from the subs roughly from 85 to 125 hz in what a lot of people term as "boominess", but it's actually heard through your main speakers more than reflected through the subs and produces what most people refer to as "mudiness" when applied to the entire band mix. That's the reason I tend to limit that range on my patches as much as I can to reduce that kind of artifact. But that really has little to do with the subs, and far more to do with carving out my own space in the frequency spectrum of the entire soundstage of instruments and voices that are heard through the main speakers. This is one of the key advantages of using an HD500X direct through the PA over a traditional mic'd amp in that I have precise control over where I sit in the mix. Typical amps don't have that precision unless you employ parametric EQ's in the sound chain.

I'm with Cruisinon on this one. It makes little or no sense to me to waste time dowloading patches regardless of their ratings given that you're not likely to be using the same amp or FRFR monitor setup or the same guitar as the author of the patch. All of these things make a significant difference in the overall sound of the patch. With all the time people waste trying to get a downloaded patch to do what they want they could have simply started from scratch and built the tone they're after. The only thing useful about a downloaded patch in my opinion is looking at the details of how the patch was put together. That might prompt a few ideas you could use in building your own.

There really isn't a reason to have an entire patch set aside just for a boost for a lead. You can do it that way but I think most people simply use it in a similar way to what clintmartin noted in his response and put something like a boost effect into the patch that's set to inactive on the patch by default, then when they come to a lead they simply turn on that effect on the same patch and turn it off after the lead. That's why the default configuration is for ABCD mode. In that mode you can turn on and off individual effects using top row FS 1,2,3, or 4 and the bottom row of footswitches allow you to cycle through different patches. It's also important to note that just because you have an effect setup in a patch, it doesn't have to be assigned to a footswitch. If that effect is constantly going to be on in that patch you can assign the footswitch as "none". That's handy for such things as reverb for example and conserves your footswitches for things you DO want to turn on and off. Likewise mulitple effect can be assigned to a single footswitch. So let's say you have a lead on a patch that uses both a boost and a delay. You can assign both of those to the same footswitch so that turning it on engages or disengages both at the same time. In some cases I've assigned something like a chorus and a boost to one footswitch, but the chorus is on by default and the boost is off by default. So when I use that footswitch it turns off the chorus and turns on the boost and vice versa when I hit the footswitch again. You can really get a lot more mileage out of a single patch with inventive configurations of the footswitches rather than always reverting to making a new patch.

That's actually the point of why many people that go direct into a PA from a POD limit the frequency range of the POD so that it will reflect more accurately the frequency ranges heard on most guitar cabinets. Like you our band engagese shelves even on mic'd guitar cabinets as well as voices, because there really isn't much value in either of those below such frequencies and helps to isolate such frequencies from going into the subwoorer. In my case I go direct into a PA and I set the Global EQ parameters to limit the range of my guitar to between 85 to 7300 hz, which is little bit more the the typical range of a guitar speaker cabinet. Otherwise the sound from the PA speakers tends to be harsh at the top end and boomy on the bottom end. Your results will likely be different if you're going into a DT25 and direct out of the DT25's XLR output. In my case I have no guitar amp and go direct from the POD XLR to the board, and another line on the stage direct to a FRFR monitor/speaker. I do this to better ensure the sound I'm hearing from the guitar on stage is pretty well reflective of the sound being heard from the PA. In any case it's worth checking to examine how well the PA speakers reproduce the sound of the guitar rig on stage. Some vary quite a bit.

I can identify with your lack of bonding experience with an FRFR setup. I made the move to an FRFR setup late last year and it's been challenging to adjust to it. The real problem is in dialing back the response of the FRFR speaker to get it closer to the range of a typical guitar amp cabinet so that my patches sound more normal. I think I've now pretty much got it where I need it to be but there's been a lot of adjustments along the way. I think it's worth it because I know what I'm hearing out of my FRFR speaker is what will be accurately projected to the audience through the PA whether I'm practicing at home or on the stage. I also discovered that because of the clarity and articulation of the FRFR it's easier to make my guitar distinguishable in the mix of the band at a lower volume, which makes the whole band sound better.

Your reaction is fairly typical to people that get into modeling hardware as the behavior your describing is pretty consistent with the way it's done by everyone. The fact is that once you've set up your patch and saved it there's really no reason to need to know what the individual values are on that patch. The only thing that's important in a gig is the master volume knob which does reflect the real value of the volume you're playing at. The way I work with patches is (and I suspect this is fairly typical of most people), when I'm setting up the patch I pay attention to the volume level of that patch and try to match it up with my other patches using a consistent master volume setting. There's a variety of ways to do this, but once the patch is gain staged consistent with my other patches I save it. Then when I get to the gig I gain stage my rig based on stage volume using only the master volume knob. With that done I never need to touch anything on the POD during the performance other than switching patches.

dvuksanovich: I've experimented quite a bit with my global EQ settings to find a place that keeps out the harshness/boominess. Like you I pretty much settled on 80hz for the low cut and my high cut is set to 7.2khz. That seems to be fairly consistent with most guidelines from audio reference books and what I see on most scopes when I examine various tones. Your mileage may vary depending on what you're looking for in your sound. To be honest I leave my low cut at 80hz because a few presets seen to need that kind of low end, but I very often offset that up to around 120hz using the parametric EQ for some presets to get real clarity on the low end.

Or....you could just adjust the low cut and high cut parameters of the global EQ for $0 and get rid of the boomy lows and broken glass highs. That's all I had to do.

Line 6 has been pretty good at keeping products alive even though they may be old. Look at how long the Spider series of amps has been around. I think there's reason to believe the HD series will be around a while.

I think I'd pass on the gig. Too many limitations to realistically do what I normally do.

Quite honestly I don't know how much longer bands will be able to continue to use analog systems given what's been happening in live music performance. The cost of buying and maintaining it is a real economic drain when there's precious little money to be made in most bars and clubs. Even at the larger venues it's just not reasonable to cart around the equipment and expect to make a decent living. The good news is the current generation seems more ready to adopt digital as that's what they've grown up with and they aren't as prone to buying in to the "voodoo" legends of analog. What's odd about this is that many of the older "grandfathers" like myself have begun moving into digital because we just don't have the energy to move all that gear...:) I honestly shake my head when I see the parent of a young teenager playing in a band with a dad following them around and telling them they need to have Marshall stacks and all of that nonsense. It makes me wonder which one of them needs to grow up more....

I'm quite familiar with IEM's and wasn't making any reference to a headphone jack. The problem isn't solved by feeding from a mixer. It's a problem related to people not blending together on stage. It's the "garbage in/garbage out" principle of live bands. You don't solve that with gear, you solve that with discipline. IEM's just tend to make it worse by focusing bandmembers on a misrepesentation of what the stage blend REALLY is. Bands that have the discipline do fine with them. Less disciplined bands tend to have the mix heavier for their instrument and don't notice when they're overplaying or leaving enough "air" for the rest of the instruments and vocals. It creates more FOH issues than it solves with folks like that.

Normally it's pretty simple with a standard active/powered monitor by using the 1/4" out set to line and Studio/direct in the system I/O config, but the Friedman doesn't appear to support 1/4" inputs which (in the case of the HD500) allows for mono connections on the left output, only XLR. You might want to call their support line because I would assume they have some recommendations for how to best configure for the HD500.

No matter what the MIDI stream sends all messages through the chain complete and intact for both channels, that's normal. If the second unit on channel two (the PRO) isn't responding then there is either a problem with the PRO recognizing it's set to channel 2, or the message itself isn't formatted appropriately for channel 2. The only other thing (although I can't see why it would make a difference) is if the PRO is configured as THRU also rather than OUT. This would date back to the early days of MIDI operations in which you needed to have a terminator on your final leg. This isn't typical nowdays, but maybe you have to somehow set the second unit's MIDI to OUT to ensure it's paying attention....just a guess...but probably a wrong one.

It sounds like you didn't configure the MIDI OUT/THRU setting to be THRU rather than the default of OUT. Setting it to MIDI THRU makes the OUT MIDI port into a THRU MIDI port.

I'm one that runs my rig pretty close to what you're describing with a Yamaha DXR12 monitor and a line direct into the PA. I'd be a bit concerned about using the F1220D if you're going to be competing with other loud instruments on stage. It's a bit light on wattage and I'd be concerned it could be overwhelved. As someone mentioned earlier, with this type of rig you'll likely have to tweak things like the global EQ to remove the boominess and harshness from a FRFR speaker, but it's really not that big a deal. I'm pretty much in the same boat as you in that I play a wide variety of styles and that's what brought me to using this type of rig. Once you tame the high and low end of the FRFR speakers you'll find much better articulation and clarity than you can normally get from a guitar amp speaker, and that becomes really evident and important when you move between styles like jazz, rock, metal, country, blues, etc. But if you're playing clubs without a direct line to the PA I'd really suggest getting up into a 500 watt or higher speaker just to make sure you have some headroom if you're playing with loud band members in a sizeable club full of people. You might get by with the rated 250w of the F1220D, but you might have to have it cranked up higher than you'd like. I was using a Behringer B212D as my home rig for practicing with my HD500X, but recently got another DXR12 as all of these type of speakers behave a bit differently and I found myself having to adjust my patches too much once I got them into my live system with the DXR12. Also the B212D didn't match up as well to the sound I was getting out lf the main PA which is based on QSC KLA12 line array speakers whereas the DXR12 is much more indicative of what those speaker will sound like. As far as Behringer having a bad rap, there were some fairly serious quality control issues they had 4 or 5 years back, but that all seems to have been resolved after they were bought out by TC Group. They appear to have some pretty solid reliability now and are even branching out into higher end/higher quality units. I'd second the advice of taking your unit down to your local Sam Ash or Guitar Center and trying out a few different speakers to get a feel for them. That's what I did when I got my DXR12.

I agree with Matt on this. In ears are great for each individual to be able to hear themselves, but they tend to get in the way of everyone blending together on their stage mix because they encourage people to focus on themselves and not on listening to everyone else.

Each normal intrument/voice channel on a PA is mono by definition. Some mixers do have stereo audio channels which take 2 1/4 inputs on that channel. I suppose that could work if that channel could also be routed to the monitors so you can hear your guitar, but that varies with the mixer. Personally I don't worry about stereo in live situations because generally the audience can't tell or would even probably notice. Most of the audience isn't perfectly positioned between the speakers to really hear it correctly, and given the acoustics and general ambient noise in most live venues it's not worth the effort.

Like you said, you can't wait until the band is more established and you can turn down gigs like this, but playing in a situation where you can't hear each other or your vocals very well is probably a pretty good way of extending how long it's going to take to get established. Kind of like a vicious circle, huh?

Bear in mind if you're going through a house PA system or one not under your control, a large portion of these systems don't use stereo panning on their channels. The thought process is that it won't appear as actual stereo to most of the people depending on their physical location in the room. Therefore they leave all channels panned to center. I suppose there's some logic to this from a soundman's perspective, but you may have to ask for panning, and you may have to argue with them to do it... Just fair warning...