pfsmith0

And make sure your expression pedal is all the way to TOE.

I use the foot controller to control this so when you go toe-heel-toe you manually get the "whoosh" effect. Kind of like putting your finger on the flange of the old tape reels. As you press harder and softer (toe - heel on the controller) you adjust the amount of delay.

Amazon has a bunch. For example: http://www.amazon.com/eforCity-TOTHGUTR10F1-4-Inch-Straight-10-Feet/dp/B005118F4G/ref=cm_cr_pr_product_top

A tech should be able to fix it. It's not very difficult if you have a little experience getting into things.

I imagine the impedance is a real resistor. It needs to interact with the pickups and emulating that would be way more expensive and complicated than just using a resistor network.

Man, I'm with you on the delay. It's intolerable in the middle of a song. I only switch patches between songs. I don't use the 4 footswitches to switch patches but to switch 4 more FX blocks. That is, I don't use the ABCD configuration but the FS5-FS8 configuration. It somewhat limits the variety of sounds you can get within a song but all the delay is gone. I can still get good clean, crunch, and lead sounds. With 8 blocks I can get a wide variety of sounds without switching patches.

I actually measured the gain and frequency response of the Vintage Preamp EQ and you can find them here. As you can see, for the signal level I was using (-20dBFS), I got clipping above Gain = 60% (when Output = 50%) or Output = 90% (when Gain = 50%). So, yes, it will easily clip but this is part of the model. It is not nasty digital clipping (all my graphs show the output remains well below 0dBFS).

Usually humbuckers are dark enough by themselves and work best working into a high impedance. I'd start at 1 Meg. You can then raise (or lower) it to get more (or less) high end as you desire.

Hmmm, mine was setup that way by default.

Yes, it's stereo. Plug your headphones into something else to make sure you're hearing both sides. Wiggle the chord, too.

1-C: although there's waaaay too much delay to switch ABCD patches in the middle of a song for me. So I am in the process of changing to F: FS1 thru FS8 for the entire set. If FS1 thru FS8 isn't enough for the whole set then I'll do F: FS1 thru FS8 plus several patches for the entire set, changing patches only between a song. 2- A: I can't think about playing with the guitar AND playing with the HD500, so I put them in song order, if the patches need to change. 3-C: I use the other Setlists for place holders and storage.

The exact same thing happened at our church. People have dealt with in in their own ways, but one of these is the most effective (by far): Rocktron Hush noise reduction pedal - don't know what they do, but it's like magic. Shield the interior cavities and pickups of single coil guitars using one of the shielding kits available at Stewmac. This GREATLY reduces the hum to the point where it's a non-issue now. Use guitars with humbucking pickups. That's what humbucking pickup are for. Everyone else just deals with it as best they can with "non-Hush" noise gates or volume pedals, but they're only marginally effective. Both the worship leader and I use the HD500 and he went with option #3 above (humbucker) and I went with option #2 (shield the guitar guts). Good luck. It's a real tough problem.

Yes, you can do this using the HD400 FX loop. Guitar -> POD input -> POD FX send -> amp input -> amp FX Send -> POD FX Return -> POD output -> amp FX Return.

Nope, the pedal 2 input requires a special pedal that controls a 10k potentiometer. You'll need to plug the mico pog into the FX send/RX loop of the HD500X, using the FX Loop block in the process. But at least you can now put anywhere in the chain you want.

If the Les Paul is x2 louder than the Strat then engage the 6dB pad switch for the Les Paul. That's why it's there.

Perhaps you're overdriving your inputs. Try moving your 2 inputs from Guitar/Same to Guitar/Variax.

I agree with SPDIF as being the best. Only a single ADC conversion takes place. This also happens with USB but I don't trust the reliability/robustness of that - maybe I'm just remembering the early USB days and it's better now. But SPDIF has always been rock steady. Always will. Digital audio is what it's made for.

Although not ideal, you can find a table to translate % to frequency here: http://line6.com/support/topic/335-frequency-response-graphs-for-hd500-eqs

I'm with you - a grave under sight. I monitor overload by taking the SPDIF output into my sound card. That way you can at least tell if the output is clipping. I don't know any way to tell if the internal stages are clipping other than to start at the input and route it to the SPDIF out to check, Then add the 1st FX and check again. Then add the 2nd, etc. until finally you get the whole signal chain checked out. What a pain and to tell you the truth, I've never done it. I just look at the final output. In my setup that's usually the loudest signal anyway.

I believe 7msec is the delay incurred when switching from one patch to the next. Very obnoxious. The latency is much faster than that (or we'd be able to hear it), but I have never measured it.

I've made these measurements and recorded the results here. The net result is: All measurements were made at 1kHz with Master Volume = max, S/PDIF gain = 0dB, and Input2 = Variax (although this didn't make any difference because I had Inputs 1 & 2 panned hard Left/Right). Guitar input pad = -5.2dB compared to normal. 1/4" (line) output has 6.1dB more signal than 1/4" (amp). Aux & Guitar (normal) have the same gain structure (except Guitar has the tuner and programmable input impedance). Guitar (normal) input to 1/4"(amp) output has 4.9dB of loss with no FX, no Amp, and Mixer = 0dB. Compared to 1/4" (amp) output, the XLR output is 9.4dB less while the phone output is 15.4dB more. Compared to the Aux/Guitar(normal) input, Mic (min gain) has 6.2dB less gain while Mic (max gain) has 38dB more gain. Aux/Guitar(normal) input clips at 0dBFS (measured at S/PDIF) with 8.3Vpp Mic (min gain) input clips at -1.1dBFS (measured at S/PDIF) with 14Vpp Mic (max gain) input clips at -1.1dBFS (measured at S/PDIF) with 91mVpp CD/MP3 input clips at various levels near 20Vpp, depending on which output is used (XLR, 1/4", or Phones)

Hi Strato, I've done this quite a few times using the 3rd technique described above (Input 1 = Guitar, Input 2 = Aux). I then have two completely independent signal chains of different FX (mostly EQ for the acoustic channel) for the two instruments. At the mixer I pan one L and the other R to get two separate, independent signals coming out. You can also pan them both center to get them both coming out the same output (not a problem if you're playing them 1 at a time). The only problem I've had is that tuner is hard-coded to read the guitar input. You have no tuner on the Aux input so you may still need to add that externally.

Studio EQ is the cleanest boost. The problem with the FX idea is that you go thru an additional set of DAC-ADC conversions. It also doesn't save any FX blocks, although it DOES save CPU horsepower. I use the Studio EQ for this.

Here's a possible explanation that builds on something that was said earlier. Many of the FX have a MIX control the combines the incoming dry signal with the processed wet signal. I've noticed there's often an unintended notch in the resulting frequency response due to the delay incurred processing the signal (I haven;t measured ALL the FX but I've measured quite a few - see http://line6.com/support/topic/335-frequency-response-graphs-for-hd500-eqs/?hl=pfsmith0&do=findComment&comment=1519 and http://line6.com/support/topic/848-gain-line-up-measurements/?hl=pfsmith0#entry466). The HD500X uses a faster speed DSP and the delay thru the FX should be less, moving the notch up to a higher frequency - maybe high enough that it becomes less of an issue. This would also be an issue for dual-path signal chains and maybe even within complicated blocks like amp sims that may have several signal paths internal to them. I can clearly see shorter processing delays as having a noticeable impact on sound quality.

Oh gosh no. I do this all the time, but I use the SPDIF connection into my soundcard (RME UFX), not the 1/4" or XLR or USB outputs, although I believe you can use any of those with no ill effects. Plus, I use the soundcard's "zero-latency loopback" feature and adjust the DAW's latency compensation to match. Not sure how to do this in Cubase (as I use Sonar) but it is a full-featured DAW so I'm sure it has this feature as well. Zero-latency monitoring has been around a long, long time. In fact, Google "zero-latency monitoring in Cubase" and see what you get.