DunedinDragon

For heavier rock tones I start with the Marshall JCM 800 through the closed back Marshall cab with SM-57 off axis. For crunchy british sounds like the Who or the Stones I go to the Vox AC30 Fawn Bright from the vintage pack through Vox silver bell speakers with SM57 off axis. For cleaner blues similar to BB King and some jazz stuff I use the Roland JC20 from the vintage pack through it's normal speakers with an 87 condensor. For classic rock tones I really like the Pete Anderson custom through the Fender Deluxe cabinet with SM57 off axis. I've also found the Soldano through a Marshall cab with SM57 off axis to be a very versatile combination as well as the HiWatt using a condensor mic.

Apparently you trust your sound guy a lot more than I've ever trusted one. Heck, I feel pretty good if he can keep the vocals heard above the instruments..

I guess I'm at the opposite end of how I use patches. I use a different patch designed for every song so the guitar matches the specific style of the song. But then I don't mind switching patches between songs since in many cases I'm swapping out guitars as well. Plus our sets are determined in advance of every performance.

I think how much you cut out of the top end is a factor of several things; what styles of music do you play, what type of guitars/pickups you use, your preference for the tone you like. I'm a little cautious about what I cut due to those factors. I play a very wide range of styles with a range of guitars so I'm a bit conservative in that regard and cut at 8.0 khz. I figure if a given patch/guitar combo on a song needs more than that I'll deal with it using a parametric EQ, but I'm trying to avoid slicing out so much that I lose the crispness and articulation I look for in some of my patches.

Okay...so maybe it's a revived zombie thread, but I think it's worthy of new life since there appear to be quite a few of us using PODs in worship environments. And sharing our approaches can probably help others in new ways to use this great technology. This isn't something I'm new to. I became a member of The Posse Band, which is the band associated with a biker church known as the Salvation Saloon located in the Tampa Bay Florida area about 5 years ago. We're a bit different than many worship groups in that we are a band. We're the church band and perform every Sunday but we also play a number of outside events and concerts in the area and have a pretty broad following throughout the area. Our musical selection consists of a very wide range of material from some of the contempory christian songs, to older christian rock/blues out of the 70's and 80's like Darryl Mansfield and such, and a number of even older songs from people like Elvis, Blind Boys of Alabama, and some re-arranged older black gospel tunes dating back into the 20's, as well as some originals which tend to have a classic rock feel to them ala Doobie Bros., Rolling Stones, Who, Steely Dan, or Aerosmith. Because of the range of material modeling is a significant part of what we do. Not only in terms of guitar, but also in our drums using various styles of electronic kits and samples that match the feel necessary for each song. I've been using modeling for quite a long time in the studio and brought that concept with me into the live environment when I joined the band. Recently I've been noticing the trend of other guitarists using pedal based modeling going direct into the PA with an FRFR monitor on stage in place of a traditional amp. I began experimenting with the POD HD ProX in the studio and found I could get a much more clear and articulated modeling similar to that in the studio using that type of setup and recently made the leap to that type of setup a couple of months ago. The first challenge was to have a home practice environment that mimics the stage environment so I can develop my patches with some degree of confidence in how they will sound live. Many have already noted the challenges in this setup due to the wide and flat frequency response you get going direct into a PA, which is VERY different than the sound that comes out of a traditional amp and cabinet. So I've had to work my way through cutting out unnatural guitar frequencies that people aren't used to hearing on electric guitar. Using the global EQ on the system setup I've placed my Low Cut at 80Hz and my High Cut at 8.0 Khz. That pretty much leaves me inside the normal range of guitar amps and cabinets and allows me to use the EQ in each patch in a more natural way. I still may have to further limit some highs and lows on certain patches either due to the guitar I'm using or to keep out the rumbling or brashness, but generally this leaves me with a very clear and articulated range that readily accepts distortion or other effects. In our style of music I don't tend to have to go overboard with effects, just mostly the basics: drive, distortion, compression, phaser or chorus, reverb, and delay in whatever measure they're required or not required for any particular song. I do employ a range of amps and cabinets and mics to get the right basic sound then add in effects as necessary. In most cases this will be similar to what was typically used by artists that specialized in that style. For example BB King style blues would use a Roland amp, early british rock like the Who will use a Vox, and harder 70's rock similar to Aerosmith will use a Marshall JCM...and so forth. In my case, because of the range of styles, I create individual patches for each song and store them all in the User 3 library or backed up onto the PC. That way I can retrieve whatever songs are needed for each performance and place them in the User 2 library and then just cycle through them throughout the performance. Each patch is named the name of the song it's for, so this gives me the added benefit of having my set list displayed on my POD. The other advantage is each of these patches is pre-gain staged so all the volumes match and I never have to worry about one patch being louder than another, I only have to worry about my master volume, which keeps my band mates happy. Another distinct advantage to this approach is I don't typically need to have a lot of footswitches setup for each patch. Generally a given effect such as reverb, delay, chorus, compressor, or drive will be on the whole time I use the patch and I only need to assign a footswitch to things like a screamer for a lead or possibly a harmony for a dual lead which leaves my feet free most of the time. Admittedly I'm still learning the nuances of how to use this system effectively, but so far it's been a great success. Due to the clarity and articulation of the patches and the FRFR speakers and monitors, I don't get lost in the mix of other instruments even if my volume is lower. Also, because of more precise control over the EQ of my guitar, it's left more space for other intruments in the band to be heard more clearly, so the overall mix has improved considerably and is much closer to a polished studio sound than a traditional live sound, which is one of the goals for our band.

Your headsets may be very deceiving when it comes to how things will sound through your Mesa Boogie depending on your setup. In an ideal world you would only want your amp to work for pure, clean, flat eq'd amplified tone and leave everything else up to the POD. The reason being is if you intend on using the amp models along with their cabinets and mic placement, you don't want the sound of that amp model to be colored by your actual amp and cabinet (and possible mic). But guitar amps and cabinets naturally add a certain amount of coloration to the sound because of their design..that's what they're supposed to do. Your headsets on the other hand, if they're decent ones, are trying to provide a very flat and even tone like you would hear from a stereo system so it's likely there are going to be some fairly big differences once you play your patches through the amp. Generally speaking you'll likely be missing some of the crispness and articulation you were hearing in the headsets, in effect a bit muddier. You can mitigate some of this by following the setup advice cruisinon2 pointed out, but you'll still likely have to make some adjustments unless you set up your patches in your band room. I play my POD direct through the PA and through a personal FRFR monitor on stage which is a Yamaha DXR12, so I end up with something of the same problem. I have a different powered speaker at home, but it doesn't respond at all like the DXR12, so I tend to have to make some adjustments once I get to the rehearsal space. I'm intending on replacing my home powered speaker with a second DXR12 for home use, but that's the only way I can be sure the sound I'm building at home will sound the same on stage.

From watching the video of it I'm not sure you couldn't achieve the same boost and sound with one of the several compressors included in the 500X. At first hearing I'd say it's most similar to the Vetta comp as it has a fixed ratio that sounds similar. It may take some playing around to get the tone you want, but I think you already have everything you need.

As far as I'm concerned the vintage pack was the only one of the packs worth having. I use the Roland quite a bit both with my Gretsch and with my Strat for bluesy BB King sounds and jazzier tones. It has a super clean and super rich sound, but without the thinness of the Fender Twin. The Vox is really the reason I got the pack however. I match that up with a compressor and a bit of crunch and I can get a really convincing british crunch sound ala Pete Townsend or Keith Richards. I haven't used the Champ yet, but I've used a champ model previously to get a Joe Walsh sound as that's what he used on a lot of his earlier tunes like "Funk 49". If you're only into heavy overdriven metal sounds you might not find a lot of use for most of the these models, but if you're looking for some very recognizeable sounds with a wider range of music styles you'll be very happy with it.

It could be interesting I admit. For gigging you'd still probably have to mic it, which would add some coloration to the tone based on mic and it's placement. The benefit would likely be that you wouldn't have to do all the tweaks to the tones necessary on a regular FRFR monitor because the amp itself would tend to impart the same limitations in frequency response you'd get on a regular amp, but with a flat EQ (I'm presuming).

First off, the folks that are asking you this question apparently don't understand the difference between their effects processors and a modeler. Your POD HD is a production level piece of equipment meant for use in a studio recording environment or stage application. Although it could be used as an effects processor, it's designed to be much more than that. No one in their right mind would suggest a Zoom G1 is anything like that. It's really a pure pure practice rig. As far as drum beats I think you have some pretty good suggestions here, but it depends on what you need it for. If it's just for personal practice to help you with timing you can use any metronome app on your phone through the MP3 input. If you just want a drummer to practice with then loop or plug in a drum machine. If you want to record your best bet will be a virtual drummer plug-in like an EZDrummer2 so you can develop the specific drum line you want with accents, builds, finishes, etc. where they need to be and they can exists independently on their own track.

I hate to revive an old thread like this, but I'm coming into a situation that makes me consider using GEQ full-time in my setups and want to gain some insight from other folks. My gigging rig consists of a Yamaha DXR12 monitor speaker for myself on stage, and my output goes direct to the PA. I realize GEQ is what it says it is...GLOBAL and it will affect both outputs. But what I've found in virtually every patch I've created I go through a lot of the same things. Generally I cut 125Hz and below as well as cut the highs around 9000Hz. This applies to clean, crunchy, or overdriven tones regardless of the guitar (Les Paul, Strat, Gretsch hollow body). The reason being is I come from more of a studio background and nothing interesting happens with guitar in those frequencies and very often they can be quite problematic in the blending with the rest of the instruments and leaving space for everyone to play. Here's my take on it. On the low cut side, I have a low cut I can apply to my monitor and most boards use the high pass filter on guitars anyway. Even if that don't, I'd prefer they do, so I'm not losing anything if I have it engaged globally. It may be redundant in some cases, but once it's cut, it's cut and it can't be cut anymore. And cutting at 125Hz leaves ample room for the bass guitar and kick drum to operate On the high cut side, most people are used to hearing guitars through guitar amps which don't produce anything with any real clarity about 9000hz anyway. I guess to be safe I could take it up to 10,000 but as I said, I don't see anything useful in that range for guitars anyway. Even acoustic guitars aren't really useful above 8000 and the higher you get you begin to interfere with creating nice airy-style blended vocals with the singers. None of this would really be a consideration were it not for the efficiency in modern FRFR speakers, or if Line 6 had really built in the typical high end limitations of most guitar speaker cabinets in their modeling. But they didn't, so it seems to me I might be leaving myself more room to adjust on my individual patches if I weren't having to constantly make these high and low end adjustments. What's your take on this?

DSP gets used up based on the number of effects and amps in your signal chain. Generally speaking this comes into play when you have built a fairly exotic signal chain but not always so. Some effects are more demanding than others and you can run into these limits earlier with extensive use of demanding ones. But generally speaking from what I've seen it typically shows up when you begin to split the signal chain to use double amps or multiple parallel effects. Probably not much of a concern with punk, but some metal styles can use some pretty wild setups.

Have you tried eliminating your effects in the signal chain one by one individually to see if one of the effects is causing that artifact? That's what it sounds like to me. It's not so much a hum as it is a chattering sound that dies off slowly when you stop playing if that's what you're referring to.

I've noticed a lot of confusion about developing patches on the Pod HD systems that are connected directly to a PA and/or FRFR monitors. I've been working with my Pods (500X and ProX) for a month or so and have learned a few tricks and tips that may be helpful to others. I think it might be nice if people share some of their findings here as well as we could all probably benefit. For the record I use an FRFR monitor on stage (Yamaha DXR12) and connect directly to a PA using QSC KLA12 speakers. Here are some of the things I've found helpful as a baseline in almost all of my patches. To get rid of boominess in the bass response: Adjust the low cut on the cab parameters to up around 100 to 125 hz. I do this on almost all my amp models and cabinets. It gives a nice, clean articulated bottom end in almost all cases. Use the bass knob sparingly. I never go above halfway and am typically at about 30% or just slightly above that. Even with high gain models it keeps it from getting muddy. To get rid of thin or tinny sound at the top end: Put a parametric EQ toward the end of the signal chain. Sometimes I slightly roll back the highs to 45%, but generally I take care of the very tinny sounds by taking the EQ up to around 80%, the Q at around 75% and the Gain down to around 30% or even lower. From what I can tell this rolls back the very high end of around 3500Hz to 8000Hz which is a bit piercing on most amps and cabinet setups. To get a tighter and better articulated overdriven sound without the muddiness: Share the drive duties between the amp and and a pedal. I generally use a Tube Drive set at about 28% and set the drive on the amp such as a Marshal JCM at around 30%. This of course would vary with the amp, but sharing these duties seems to give it some good pump but not sound all mashed together when playing chorus or verses. For leads I add in a Tube Screamer with the drive set around 30% which gives it pretty good sustain but with a lot of clarity. To get a good british crunch style sound similar to the Who, Stones, or even Doobie Bros: Use a bit cleaner amp like a HiWatt with the drive around 40% (or more if you want a bit more crunch) and combine that with a compressor toward the front of your signal chain (I prefer the Vetta Comp) with Sensitivity around 30 to 50% and Level around 75 to 80%, but mostly follow your ears. I often use an analog chorus at very low speed, depth around 35, tone about 50% or slightly above and the mix at 40% or 35%. Not so much that you really hear the chorus but it adds a bit more body to the sound. Those are all the major things I've come up with. Feel free to share your lessons learned.

There's a big difference between live sound and processed and mixed sound like you would normally hear on your stereo. Live sound has a much greater dynamic range and stereo systems are designed to accurately produce sounds that have been mixed and mastered in a studio.

The advise about using something that's designed to do a million things and something that's designed to be very specific is worth listening to. I spent the vast majority of my life as a developer and designer of software and hard-coded real-time devices. Although a generalized device like an iPad or computer can be very flexible, those programs are run in free memory and have to negotiate with the operating system for access to hardware devices, memory, and embedded system functionality while interacting through a layer of security and communications software. There's a cost involved in these things that direct hardware programming in a specialized system doesn't have to pay. Where this typically plays out is not in a casual testing of functionality, but in a production environment where things are changing quickly and relentlessly...like in a live performance over a span of several hours. Places where it's critically important that things don't fail. Playing around and getting something to work is not the same as stress testing, which is where these type of generalized devices will most likely have problems as they try to allocate, discard memory, and cleanup unused segments of memory and such...things a hard-coded hardware device doesn't have to deal with because it's got a very specific job and only that job to do. This is one of the reasons you see the DSP limits which are hardcoded into the Pod..so it doesn't exceed it's real-time performance requirements.

You're going to find there are two schools of thought here for connecting output devices. Those that want to continue using their existing amps on stage and those (like myself) that have abandoned using traditional amps and use direct connections to the PA and/or using a powered FRFR monitor as an output. What AlexKenivel mentioned sounds like a reasonable way to connect to an amp and still take full advantage of the abilities of the Pod. The one thing I would mention is in setting up your patches. Ideally you'll have your amp available while you work on the patches as the sounds can vary greatly from what you would hear from headsets or other sources. I try to make sure I have the same FRFR speaker at home when I'm working on patches so I don't get surprised when I go live with them. Personally I have a basic set of patches for things like a clean tone, jazz, country, bluesy, crunchy, overdriven, etc. that I draw from and I tinker with them to create specific patches for each song we do. I keep the patches I create for each song stored on my laptop and then pull them in for each show in the order we have in the set list. Often I'll just copy an existing song patch that similar to another song we're doing, tweak it for the differences and save it under the new name. That way they stay very consistent and it keeps all the patches less confusing to use because I only assign footswitches within the patch to things that will change in that particular song..like where a lead kicks in. Otherwise the other effects for that patch in that particular song are always on. I name and save the patch with the same name as the song so I have my own setlist right there on my Pod during a performance and just click through them one by one as we perform. It's a little more work up front, but it really pays off when we're performing live as our setlists change week to week. Just one way to approach things. Good luck.

One of the problems with downloaded patches is you have no idea how they were created, and that can make a HUGE difference in how they will sound on your system. But I have to go along with prsmith0 on this. If your patch is making use of an amp model and then you're pumping it through a regular amp on top of that it's bound to be an unholy mess. You can mitigate some of that sometimes by making sure you are using the right output selection of amp rather than line, but you're still going to contend with a number of factors one of which is two amp stages, one modeled and one real. Alternatively you can take note of the modeled amp settings then replace the modeled amp with the preamp version of it and use the settings from the full modeled amp you noted down. Even then if the patch was created and EQ'd using a FRFR style amp/speaker, that's going to produce a much wider bandwidth sound than what's produced on your MESA half-stack. And again in this case you're also dealing with a modeled cabinet with a given mic in the patch being pushed through a real cabinet and possible different mic from your live rig...yet another mess. Although I'm relatively new to the Line6 HD Pro series modeling and to this forum, I've been working with modeling for many years in various forms and it's something you have to adapt to. From what I can tell there are a couple of different approaches being used by people on this forum. Some people that want to retain the use of a traditional amp simply use the POD HD as an effect processor and leave out any amp/cabinet selections in their signal chains, or at a minimum use the preamp versions. Evidently with enough tweaking and settings in the signal chain many are happy with this arrangement. As for myself I've embraced a complete modeled environment. Therefore both my stage rig and my home rig consist of powered FRFR speakers. On my live rig I go direct out to the mixing board and use a FRFR speaker on stage as my personal monitor. Even then you can run into some slight differences in how various speakers respond, but that's pretty easy to adjust. For myself I don't bother with downloading patches for use. I might download a patch to examine what kind of settings and effects they used to get the sound, but I'll typicallly just use that as a guideline. I generally develop my patches on my recording studio rig which consists of a Pod HDProX going into my studio moniors which are Yamaha HS-8's. I then convert and transfer the patch to the home practice rig which consists of the PodHD500X connected to a Behringer B212D (which I hope to replace soon) to work out my live sound. Generally there's not a lot to adjust and over time I've developed a set of practices with my patches that help limit some of the bigger issues with very high and low frequencies which simplifies things. I then use the Pod HD500X as my stage rig and connect it to a Yamaha DXR12 stage monitor and send the XLR out to the sound board. Sometimes even there I may have to adjust some things, but it's generally getting pretty consistent now that I've been using this rig for a month or so. My personal approach with this type of modeling has been to approach it more like as if I was developing my tone for a studio setting rather than live environment..thus the reason for having FRFR speakers as a primary element in all that I'm doing. By doing that I can take much greater advantage of the possibilies in the modeling signal chain and I get a much more precise, managed, and articulated sound on stage that blends MUCH better with the rest of the band and produces a much more polished sound for the audience.

If by amp you mean a headphone amp then the only value it might have is in attenuating the volume output of the POD. That's only relevant if you need to use the POD's volume control for the amount of signal you're sending to the PA or to line into a recording rig. In that case you could keep your headphone volume lower than the outputs of the POD for other purposes which might be need to be louder. Personally, if I'm using the headphones I'm not feeding the other outputs anywhere else, so the master volume on the POD works fine. Bear in mind however that even some of the most high end headphones are going to sound different generally than an FRFR type speaker used on a PA or powered monitor. So if you dial in your patch on the headphone you may have to adjust it for live application.

The JBL will probably be closer to what your FOH is at church depending on what you're using at church. I use a Yamaha DXR12 for setting up my patches which gets played through a QSC KLA12's at church. It's a pretty good match but there are a couple of very important things I've learned about building patches for use on FRFR systems. First is what you've already found about the muddied bass. FRFR systems are FAR more responsive than any guitar amp in the low frequences. They're also very good at accurately separating and projecting the mid frequencies and it's the combination of those two things that are different from guitar amps and cause what you refer to as muddiness. To my ear it doesn't sound muddy as much as just unnatural, because on a guitar amp you'll mostly just hear the bass which tends to cancel most of the mids. As a rule of thumb on every patch I build I use the deep edit paramaters on the amp model to set the high pass filter up to aroun 100hz or even as high as 125hz. This gives me a much more crisp sound on my low bass strings. Not exactly like an amp, but more like a studio recorded and processed guitar which is much cleaner and tighter. Even with that you'll have to be careful about how much bass you dial in on the patch. FRFR speakers will make things sound bassy to your ear when compared to traditional amp speakers. A little bit goes a long way. The same logic applies to treble but it's not as dramatic as the bass. Most of it you can deal with by dialing the treble back. I also sometimes use a parametric EQ at the end of my signal chain to reduce the very high end at around 8000 hz and up depending on the amp, cabinet, and microphone. The best advice I can give you is to approach building your patches as if you're building them for a studio recording effort rather than a live performance when using FRFR speakers. In a studio a lot of time is spent finding where each instrument needs to sit within the mix so as not to interfere with the other instruments and vocals yet stand out in the overall mix clearly. Sometimes this may not sound as natural to your ear when coming from a traditional amp setup, but it pays off in HUGE benefits to the overall music when played with the rest of the band. If done correctly you won't need volume in order to stand out in the mix and it makes everything sound much cleaner and tighter. Of all the differences I've noted in dealing with FRFR systems, the most noteable is in the pickups and type of guitar being used. A strat sounds much more "stratty" and a Les Paul sounds much more "Les Paul-ish" and a Gretsch sounds very "Gretchy". When coming through a traditional amp I could sometimes dial in more of a Les Paul sound with my Strat than I can now and vice versa. That results in me being much more careful when selecting the type of guitar I'm going to use on a given song. Finally, the clarity and articulation of an FRFR system is so much better than a traditional amp I've had to adjust my technique on certain types of things. In essence get rid of the sloppiness, because it won't get covered up. For example, when I was palm muting on chords I was often not constraining my picking to the two or three key strings I wanted to mute. It didn't matter so much with an amp, but it stands out clearly on an FRFR setup. The added benefit however comes in arppegio-style strums of chords. I find I use them much more often because the articulation is so much clearer than they used to be on an amp.

Hint: If the computer is crashing only when the HD500 is plugged in, it's not a Microsoft issue. It's a Line 6 driver issue. Microsoft provides ample documentation and plenty of examples for how to write crash-proof drivers to anyone that requests it. If someone chooses to not follow the guidelines and go their own way, this is the price they pay. There are TONS of audio interfaces on the market that work just fine on any of the above versions of Windows. I have a MOTU that's been running for years, and several others before that one. It's ludicrous to place the blame on Microsoft for the failure of companies that don't do appropriate testing of their products. This is also the reason I don't use my HD500x or HDProX as an audio interface. I use a company (MOTU) whose business depends on it being a stable audio interface. The audio interface on the POD is just an added feature. If you want things to work right put a tested and proven audio interface on the computer and plug the POD into that via the XLR outs. Guaranteed that will work just fine.

I've never felt comfortable totally putting my fate in the hands of any venue unless they're a well-known quantity, which can be pretty rare. That's why I'll at least bring my own monitor. If nothing else I can drive that off of my 1/4" outs and feed my XLR to the board and be pretty assured the audience is hearing what I'm hearing as well as get a sense of how I'm blending with everyone else on stage.

If you want consistency in your sound why wouldn't you just send a monitor feed back from the sound board to the stage? Certainly if you split the signals between amps and FOH Direct like what you're talking about, your stage sound won't be consistent with what's broadcast/recorded. Also have you mentioned to the sound engineers that you need stereo separation? Typically most sound engineers don't separate channels left and right for live performances because only a small portion of the audience are in optimal positions to be able to hear the stereo separation consistently.

I have a feeling the white noise is coming from the EQ as well most likely due to line level signals being out of whack. Generally something like an EQ would be setup as a pre or post insert effect either on the mixer's channel or overall output prior to output to the speakers. Alternatively it could be placed as an external effect from within the HD500X signal chain. Neither of those would affect the line level signals from the POD to the mixer or from the mixer to the speaker.

How are you listening to the HD500X when you're dialing in your patches at home? At home I have my POD plugged into a powered PA speaker using a 1/4" output with the LINE switch on and I have the STUDIO/DIRECT setting on the POD for the XLR output which goes to the sound board at church. That gives me a pretty good match between what I hear when I practice at home and what comes out at church. As mentioned before, with this type of setting you'll find there's a lot more responsiveness to high and low frequencies because PA speakers are very different from guitar amp speakers or headphones. Make sure the people running the sound board at your church have your EQ set to flat or possibly roll off a bit of the high end on your guitar channel. As a general principle I tweak the high pass filter settings on the whatever amp model I've selected to change the high-pass filter setting to around 100 Hz which takes out a lot of the bassiness and gives me a pretty clean tone on my low E string on my patches. In some cases I'll also use a parametric EQ effect at the end of the signal chain to roll off very high frequences. As far as stereo output, it's not likely your sound people are using stereo separation on the mixer anyway. Live sound people rarely do since it would only sound right to people in the center of the crowd and would sound odd and unbalanced to everyone else seated to the left or to the right. So as a general rule most sound people don't pan the channels.