pfsmith0

Put a volume control in front of a delay. Rock the volume back while you strum and then rock it foward. You want to feed the delay with only the sustained part of the chord, not the attack. Keep doing this at a rate that keeps the decay from the delay minimal. Usually you can set it up so you only need to strum every few seconds. It'll sound like one hugely long chord as long as you keep your strum volume consistent.

Windows may have changed your sound palyback/recording device. Right-click the speaker icon in the toolbar on the lower right of the screen near the clock and select Playback Devices and see what is listed. You may need to change the default device. You may also need to do the same for Recording Devices.

The idea of using the FX Loop is a wonderful idea! That should solve your problem. Set it up for Combo mode and feed Channel A to the normal output to your amp. For channel B, insert the FX loop at the beginning (but after any acoustic FX blocks you may want) and connect Send to your PA. That should work great! This just shows how flexible the IO is on this gear.

Oh. Right. I missed that part of the problem. Yes, I believe you're stuck with one mode or the other. I always use studio direct into the PA to minimize stage noise.

You can get two outputs by taking advantage of HD's stereo signals. For example, dedicate Channel A to electric and Channel B to acoustic. Set Channel A input to Guitar and Channel B input to Aux (1/4") or MIc (XLR). Move all the FX and Amp blocks to the left of the mixer, putting them in either Channel A or B as needed. Keep the two channels separate all the way to the mixer. You don't want any FX blocks combining the two signals. Then, in the mixer you pan the two channels hard left/right and take your two outputs to where you want. I do this all the time with acoustic and electric instruments and it works fine for me. The only downside I've found is the built-in tuner is only available to the Guitar input, but this can be fixed by using different patches for acoustic and electric instruments. Just make sure the patches pan hard right/left so the acoustic/electric come out different outputs.

Check out the Seymour Duncan comparison guide at http://www.seymourduncan.com/comparetones. You'll see lots of variation among the single coils and lots of variation between the humbuckers. For the sake of an example, though, many single coils have peaks around 9kHz, so you'll want to dial in an EQ to cut that some. You'll also see many humbuckers have peaks around 6kHz, so you'll want to use the EQ to amplify that somewhat. Then there's output amplitude. Humbuckers are louder than single coils, so you'll want to add a little gain, maybe 3-6dB? Not sure about that... Then there's alnico II vs alnico V magnetics, but you'll these in both pickups. However, humbucker's have more windings (which is why their output is louder and their resonant peak is lower) so they distort the magnetics more than single coil. So you'll need to dial in some mild distortion, but I don't think any of the FX in the HD500 will do a very good job of that, but you could try. All-in-all, you won't have very good success. But if you're requirements are lo-fi and you're willing to try something, maybe some of these points will provide some guidance.

I've used the HD500 for a while now and have settled in on the Soldano clean and crunch amps. A noise gate (P90s and automated stage lighting causes lots of hum) and some mild compression are always on, as is a small chamber reverb. I use digital delay sparingly, usually with a 1/4 or dotted 1/8 settling. I add in a tuber screamer for solos and the occasional chorus, phaser, flanger. The delay, tube screamer, & modulation leaves me with one foot switch left available for something special when needed. I usually set up ABCD to be clean, mild crunch, major crunch, hi-gain, but I'm starting to get a little frustrated with the 5msec delay when switching presets. I don't always have that much time to switch sounds so I'll be moving to the FS1-8 paradigm. 5msec doesn't sound like much but it mutes the sound while the new preset loads in and a 5msec mute is very much noticable. I use Channel B (mic input) for my acoustic instruments. This only contains gain and EQ. Different patches are switched in for acoustic guitar, mandolin, uke, and banjo, so they all come out at about the same level. The EQ is set to null out the few major feedback modes I get with acoustics. Thsi works very well, except the tuner is only active on the guitar 1/4" input, so I have to use an external tuner (kind of defeats the purpose of using the high quality XLR mic input). Because of this I'll be reconfiguring this to allow me to plug into the guitar 1/4" jack for all instruments and use different presets as I change instruments. I can't change instruments in 5msec, I'm good with that. This all comes out the XLR to the main mixer. I'm very happy with this setup. It's quality is about 80% of a purely analog setup but the convenience is well worth it. Almost all of the people in the congregation don't notice the difference and half of the other guitar players who use analog systems don't get the best sound out of them anyway (that I've noticed in our church anyway), so it's pretty much a no-brainer to go the HD500 rout.

Check out http://line6.com/support/topic/848-gain-line-up-measurements/?hl=pfsmith0&do=findComment&comment=4669. I measured a 5.2dB of attenuation when Normal is switch to Pad.

1) not sure what you're question is. I hook up the XLR to a PA all the time. In fact , it's my preferred way. No DI box necessary. If you use an on-stage amp, chances are it will change the frequency response too much. 2) I use GROUND. If you hear 60 cycle buzz you can switch it to LIFT to see if it helps. You can't hurt anything (if the venue is wired in accordance to code) by using one or the other. It's purpose is to reduce ground loops if 60 cycle buzz is a problem. 3) I use Amp. The only difference is the level of the signal coming out the 1/4" output. If you don't use the 1/4" output, then don't worry about it as it has no other effect. If you DO use the 1/4" output, you can flip this switch to the one that does not overdrive whatever the POD is plugged into. 4) I'm still struggling with this. In general, headphones seem to accentuate the mids/bass, so you'll tend to add too much highs. Drop the highs back a bit in the headphone mix and it should sound less "fizzy" in the PA. Other than that, it's an iterative process - try, adjust, try gain, adjust again, etc. t took me about 6 tries before I felt good about the live sound. 5) Yes, if you're just going thru the PA.

I've done quite a bit of measurements on the HD500 and summarized the results in http://line6.com/support/topic/848-gain-line-up-measurements/?hl=frequency&do=findComment&comment=5206 and http://line6.com/support/topic/335-frequency-response-graphs-for-hd500-eqs/?hl=frequency. I've found that digital clipping occurs long before any of the analog output stages clip. Also, except for the the Mic input (which has it's own trim control), digital clipping occurs before any of the analog input stages clip.

Couldn't you do this by putting each amp in it's own A/B signal path and then using the foot pedal to change the mixer gains between 100/0% and 0/100% for A/B? It wouldn't be a push button but it would still be a single foot control. No muted delays. Plus, you could then vary how sudden the switch happens and you would also maybe get some interesting mixing effects in the 50/50% region. In addition, since the reverb/delays are part of the common signal path, you keep your spillover tails.

I just installed the new Line6 Monkey and HD500 driver update. Has anyone figured out what has changed?

Does Garageband allow you to meter the input levels? How close to 0dB is the guitar getting? Did the input impedance of the Guitar input get reset to a value you used to not have? Check your USB connection. The jack on HD500 is fragile and can easily break. I wonder if the pins are losing connection intermittently. I'm not sure, but normal USB links may have automatic retries for digital communications, so it may work seemingly fine for non-realtime tranfers like patch downloads and such. But when you try to stream audio across it you may be getting lots of missed samples.

Yes, this happenbs to a lot of us. The USB connector is not very robust. Don't even try to straighten the pins. It'll be futile. If you're handy at soldering you can replace the USB connector yourself (since it's out of warranty anyway). http://line6.com/support/message/381500 has a link to Mouser where you can buy the part you need. They're less than $1 USD each.

Take out everything in the signal path - no FX, no AMP. Then use the S/PDIF output into a DAW and look at the meters. That'll tell you how close to clipping you are. Some DAWs have a clip indicator, so that would be even better. The key is to have your input signal transfer directly to the digital output with nothing in between. That way you're measuring the input signal.

It is my understanding you can use a MIDI foot controller to send MIDI commands to the HD500 to do the bank switching, Then you would have 8 buttons for direct patch access.

Hello again, I re-ran the tests with the amps turned off and the results are much clearer. I saw exactly 6.02dB of difference which is what I would expect (6.02dB = x2). Using just input 1 was x2 lower than the other two - Input 1&2 and seperate path A. Apparently there's enough compression/clipping/overdrive in the amp model that even though drive level changes (and you can hear the difference in crunch) the output level does not change. I've deleted the earlier file and uploaded a new one here. The first tab shows the frequency response with no amp (and no FX or anything) and you can clearly see the 6dB difference in signal levels. The second tab shows the same thing but with the amp turned on. No difference in output level (even though the input has changed by 6dB). This is also a great example of why guitar amps make lowsy general purpose audio amplifiers. The frequency response sucks (literally, there's a ~40dB notch at 6 kHz). In addition, I can also verify that the HD500 qualifes for "HD". The raw frequency response is quite flat (+/-.2dB) over 20-20kHz and the 24 bit noise floor easily lets me discern signals near -100dBFS. FR.zip

Hi Hurghanico, I went back and remeasured the HD500 and got the same results as before. The inputs were indeed set to preset. As a sanity check I took your 1&2 input configuration and panned both channels to center and the level increased as expected, although the increase was 3.5dB which wasn't what I expected but it's reasonable I suppose. I was using a 0 dBV signal which isn't particularly small but it wasn't clipping. Perhaps your signal is larger and the amp is being driven into saturation at different levels for the various cases. I believe this was your interpretation as well. An idea may be to insert the FX Loop in one of the channels to allow me to tap into the middle of the signal chain somewhere an read the signal there. If that idea is worth pursing, please send me new presets with the FX Loops where you'd like to see the signal. Wait, wait! I just read your original request again and you wanted the amp turned on/off. Sheesh... how did I not see this before? All my measurements were with the amp on. Let me get back to you (again). Sorry...

Good point! You told me preset, didn't you? Let me re-check that. Standby....

Yes, I wish the HD500 used TOSLINK instead of S/PDIF. That would have solved your problem. it seems like a natural fit with all the ground loop issues common to electric musical instrument setups.

Hi Hurghanico, I was able to make your measurements today and attached them here. I saw almost an exact match between left & right channels for each of the 3 configurations. All 3 configurations were basically the same. I saw nothing significantly different between them. File removed - see updated file in my post below...

Hi hurghanico, that should be very easy to do. With Father's Day coming (in the US), it may take me a few weeks. I'll post the results here, so keep an eye open for it.

In order to get a feel for the signal levels suitable for the HD500, I've made a few measurements of the gains available from the different inputs to the various outputs. I've also measured the gain of all the FX. A small Excel spreadsheet containing the data can be found here (unfortunately, I can only upload a PDF version of the data, which I've done here). A brief summary is shown below: 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 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) By far most of the FX have 0dB of gain, but there are several exceptions (e.g., Graphic EQ = 2.9dB, Tape Echo = -5dB, and others). FX were measured with Mix=0% (to eliminate comb filtering effects) and Gain=0dB (usually). The PDF files contain the whole list. Using some of this data you can see the clip level is essentially set by digital clipping, although the mic input stage only gets you to within 1.1dB of fullscale (close enough to call it fullscale in my opinion). The output stages will not get close to their clip levels (unless you use the CD/MP3 input which bypasses the digital engine). That is, it's sufficient to look at the S/PDIF signal to keep your signal below fullscale. You don't have to worry about clipping the input stages. You also don't have to worry about clipping the analog output stages. I hope you find this useful in optimizing the setup of your equipment. IO Gain.pdf FX Gain.pdf

Thanks for the responses so far. I play direct into a PA in a band setting. I don't know if anyone else in the band (or audience) hears the 15msec mute but I sure do! It's not the same as a 15msec delay (which my foot tapping can easily cause) because of the mute. That's the problem. Not delay. It sounds like most of you live with the mute and work around it. Said another way, the flexibility that ABCD patches give you far outweights the hassle of mitigating the effects of the mute. Maybe I just have to try harder but, man, it's really annoying and takes away from "the moment".

I originally planned on using setlists for different, um, sets. And then I’d select different banks for different songs, 16 banks is plenty for my sets. Then, within a song, a can select ABCD patches for different parts of the song – clean, crunch, lead, and maybe a high-gain lead or something. Then I also have FX 1, 2, 3 & 4 when I need something special for each of the 4 ABCD sounds. Sounds like a plan. I decided to save one of the 1,2,3,4 slots by relying on amp models for my distortion, like selecting between Soldano clean, crunch, and high gain. That way, I select ABCD for the type of distortion and use 1234 for delay, chorus, tremolo, and some other effect (compression, noise gate, reverb are always on – no need for a 1234 switch on those). The problem I’m finding is it takes 15msec for a new ABCD patch to load in (I measured it). This is too long. I can’t always anticipate the timing to punch it in, or I don’t have that much time. The audio is muted during this 15msec and you hear it when I switch between, say, clean rhythm to scorching lead. So, now I’m thinking about going back to 1-8 FX buttons and just bank selecting between songs. But then I’m stuck with a single amp model, which is workable I suppose by using more of the FX buttons to switch in different distortion FX, but not as flexible. So my question for the group is: what is your strategy in using setlists, banks, ABCD patches, and 1234 switches. How do you use them? Are you stuck with a single amp model for a given song? How do you overcome the 15msec audio mute when switching ABCD patches? When do you use ABCD patches and when do you use 1234 FX switches? Thanks for any insight you can provide me.