pfsmith0

I've used both Axiom (for many years) and the Behringer system (for ~2 yrs) and your buddy is correct. The Behringer system is better. There are a lot more channels and a lot more flexibility and are a lot less intrusive on stage. However, it does not have headphone drivers like the Axiom does, so you'll still need to add those.

Here's the post.

I use the HD500 with my Taylor 814 as well as my Deering banjo, Red Diamond mandolin (my avatar), several ukes, and a lap dulcimer. I mainly use it to equalize gains, set EQ to minimize feedback, and use on the the FS to give me 6dB of clean gain for solos.. As mentioned previously, no amp sims. It does not color the sound at all. Whatever the pickup sounds like, that's what you'll here. A side benefit is that you can set the input Z to 3M (maybe more?) which is required for piezo transducers.

Be sure to get dual-drivers. Triple drivers are overkill. Single drives lose either high-end or bottom-end. Westones are the standards and are notoriously very nice. If you can afford those, I like them the best. I have both Westones and Shures and I like the fit/feel of the Westones better. But signal fidelity-wise, I don't think I can hear the difference between these two. Etymotic is a 3rd company to look at. They're somewhat more boutique, for lack of a better word. Some people really like them but i can't say one way or the other since I don't have any. Then there's the R-50 from Rock-It Sounds. Less expensive but with less bottom end. They are very clear but can bottom out when the signal level is cranked. Not so good used live but I use them when I'm flying. IEMs are best used when each member can have their own signal feed. We use the X32-Q app that works with the Behringer x32 digital mixer and, wow, the system works very, very well. I use my iPhone or iPad as my own personal mixer. So does everyone else in the band. I can tweak what I hear to maximize whatever I want or need. And this is all independent of what the house hears. It's great. I must say, though, that it took about 1 month of use for us to become comfortable with using IEMs. At first you feel like there's a barrier between you and the other members of the band as well as between you and the audience, but this goes away as you get used to it. I don't even notice it anymore. We still give each other visual cues, we still feel the groove (actually, we feel it better now), we still get turned on by audience appreciation, etc. Stage volume is very low (drums are in a cage). Mic bleedthru is gone. Hearing is saved. Can't say enough good things about the system.

Just a point of clarification: setting the input to impedance to AUTO does not let the POD manage input impedance "correctly". It just sets the impedance to that of whatever is in first FX block. So as you change patches with different 1st FX you may get different tonal characteristics. It introduces another set of perhaps unintended variability. If someone is having a hard time dialing in their tone they may benefit from setting the input impedance to a constant value. Higher value = harder/brighter.Lower value = more polite/darker.

Didn't Keith Richards use a fuzz box to emulate a sax on "Satisfaction"? I'd start there.

You may find some interesting discussions on pre- and full amp models here and here where I summarize the frequency response, gain, and distortion of the various models.

When you say you can "hear it fine through the headphones", do you hear one output in the L channel and one in the R channel? Perhaps you need to pan both signals to the center in the mixer. I'm thinking taking only the Left XLR output is losing one of the signals. Or, plugging in a standard 1/4" mono plug also messes with the stereo image, automatically summing both channels to mono. If you unplug the 1/4" plug, is the XLR signal still messed up?

No, the numbers to the right are the EQ settings in dB as dialed in on the HD500. The numbers to the left are the output level obtained in dBFS. Using the Graphic EQ for example, when I set it to 0dB I got -20dBFS output (because I'm inputting a signal that is -20dBFS). For this EQ, the gain is accurate. At 0dB, -20dBFS input give a -20dBFS output. Gain = 0dB. Gains for the other EQs are: Graphic EQ: 0dB gain setting = 0dB gain Vintage Pre: 50% gain setting & 75% output setting = 0dB gain Mid-Focus EQ: 12% gain setting= 0dB gain Studio EQ: 4.6dB gain setting = 0dB gain Parametric EQ: 50% gain setting = -3dB gain 4 Band Shift EQ: 0dB setting = -3dB gain Q filter: hard to tell exactly but if you set mix = 0% then 0% gain setting = -3dB gain Note that for Global EQ, which I measured much later (since this feature was added much later), I actually DID measure true gain. At mid-settings it has a gain of -15dBFS/V. That is, a 0dBV input signal will give a -15dBFS output signal.

Hi, the graphs don't show gain per se, as in 0 dB mean gain =1. This is because the input is in volts (dBV) while the output is in fullscale (dBFS). The units of measure are different. Rather, I adjusted the input level until I got a -20dBFS signal for a blank, default signal chain. The output of a filter set to 0dB would then be -20dBFS. The graphs show the output level in dBFS, not gain. Check the units of the Y-axis. It's the absolute signal level in dBFS, not gain in dB. But, if I were to to do this again I would convert it all to dB as I think that would be more useful. Basically add 20 dB to everything and call it dB. Gosh, this is harder to describe clearly than I thought...

+1 Studio EQ

Does the Helix still have this issue? That will tell us how high this on their to-do list.

It isn't so much a question of spill over as it is mute. Your signal is just.... gone... for a short blip. You really have to adjust your arrangement so it doesn't impact anything. I can't do it. It cramps my style too much. It is so much easier for me to just use the FX switches within a given song. That gives me tons of variations.

The only way to NOT get the "latency" is to NOT switch patches within a song but just turn on/off FX. If you need more sound variety within the song (the main reason to switch patches) then switch to FS1-8 mode. Only switch patches between songs when there's time for the mute. The mute drives me CRAZY so that's what I do, Run the HD500 in FS1-8 to get all sorts of FX within a song. Then switch patches for the next song. I'm not sure, but I think MIDI can be used to turn on/off FX in FS1-8 mode (someone will correct me if I'm wrong), so there should be no loss flexibility using MIDI. FYI, what you're describing is not really latency but an outright mute. I.e., there's no delay in the sound.

Does it change when someone adjusts the lights?

I measures the Global EQ frequency responses and posted the results in this thread.Perhaps it will give you some insight into what they're doing.

I had the exact same question! So I measured it, along with all the other EQs. See if this thread helps explain it.

OK, that explains why he gets phasing issues. The noise gate mixes L/R together to get the mono output. If you don't want to worry about phasing issues then you'll need to keep the L/R stereo pair all the way out to L/R amps and speakers.

But remember than when you used two amps for years while gigging, they came out of different speakers. If you use dual amps in the HD500 and pan one left and one right and come out stereo going, again, to two different speakers you won't run into phase issues. Once you get a few feet away from the speakers the phase is all screwed up due to room acoustics. The phase between the two speakers will interact with each other in exactly the same way your gigging amps did. If you had no issues then, you will have no issues now. The reason there was phase issues in the video was because he was summing the outputs together in the electrical domain. The phase is very well-controlled inside the unit. Shoot, you even get phase issues with just the mix control of the wah models (there's more phase shift thru the wah processing than without and the mix control adds those two together with the result of notches and peaks at various frequencies - granted it may be a desirable effect in a wah but it phase cancellation none the less whether intended or not). As long as you set the mixer for path A = 100% L and path B = 100% right (or visa versa), then you won't have any phase issues.

To answer one of the original questions, a low E on a standard tuned guitar is around 82.4Hz. That being said, you can have a highpass rolloff somewhat higher than that without losing much. You'll still hear a "low E" even if it's attenuated quite a bit. The ear is a wonderful and somewhat mysteries thing. Even if the fundamental is gone the brain fills it in based on what it hears in the harmonics.

Oh, it wasn't just those. Over Christmas I was watching an old Fred Astaire movie and he had a scene in "black face". Wouldn't see any of that nowadays, And Hollywood prided itself in how liberal and inclusive they were....

I can verify that SPDIF is 24 bits.

The whole idea is the biggest source of signal degradation in this system are the data converters so you want to go thru as few of those as possible. It's probably true, but whether or not you can hear the difference is another issue. I figure if it's true then why go there, so I use SPDIF and only go thru one set of conversions.

Lots of people turn off their wifi adaptor in Device Manager before doing recording as it has bursty demands on the CPU and can cause audio pops.

I have been very frustrated trying to get single coils and humbuckers to share the same signal path. The single coils have such a low level output that all my compression and overdrive settings wouldn't work as expected. I could have used an FX block to attenuate the humbucker output but I wouldn't always remember to engage that. It was easier to just have 2 different signal chains. One option may be to plug one guitar into Guitar and the other into Aux. Attenuate the humbucker and then mix the two into a common signal path. That would probably work, although the tuner would only work with the Path 1 guitar.