innovine

My first pair of jj's were a great improvement in the noise floor of my dt25. Tone wise I don't know, a bit clearer and brighter maybe, but certainly no worse. They lasted about 3 months before one blew up. Second pair lasted about 3 months too (I play a few hours a week, always warming them up before leaving standby, properly biased, etc). After that I wasn't too enthusiastic over buying a third pair. So I stuck the remaining, unmatched tubes in instead, and even tho they are off by a good amount they have lasted much longer, must be over 9 months on each tube now. There's a slight hum but otherwise no other side effects that I've noticed from being unmatched. Perhaps its only SOME jj's which blow up? The sound quality difference is enormous compared to the stock tubes. Bogner designed a piece of lollipop if the best the amp can do is a choice between sounding noisy, or burning out the tubes every few months.

Also, I have found much better results with the master low and LVM off, rather than the other way around. YMMV of course, but in my experience I haven't found a tone with lvm that I couldn't better with lvm off. It sucks.

You can only get the tube distortion by having it really loud. Well, you could add an attenuator which will let the tubes work hard yet cut some of the volume, but for that tube tone you need to have them tubes working hard. You'll find plenty of good tone without having to crank it though. In this case you're hearing preamp distortion only, so its a lot like just a modeller, but you get a) a real cab and speaker and b) tons of volume and headroom. These matter, probably even more than 'tube tone'. I just love the sound of the amp. Even though I could run my pod into decent studio monitors I always play with the amp. Just sounds loud and solid like that. You're probably overvaluing tube tone. Learn to dial in a good tone just with the preamp first, and worry about magic tube warmth later.

Who the hell buys an amp and sells it after two hours? Sounds very weird to me. My dt12 tubes are too hot to touch, and smell warm if I put my face to the back of the amp, but the rest of it is cool to the touch. Sometimes, if the amp has been idle there can be a little dust on the tubes which burns and smells a little when it warms up, but your description sounds a lot worse that these..

Well, the point is the same. If you listen to just the dt for a while you will loose some objectivity in your judgements of how much bass and treble you want. Your ear automatically recalibrates things which is why your patches sound crap the next day. For instance, if you have a very bright tone, your ear will start to automatically dampen high frequencies, to better hear mids and lows. After a while you might even notice this, and so yoU dial in even more treble to compensate. The day after, you wonder how you thought that ice pick tone was dialled in. Listening to anything else at all (even other patches) every few minutes can help avoid this.

When dialing in tones, you shouldn't listen to the same thing for more than a few minutes.. your ears will start to compensate and even out your perceptions the frequency space. Switch to a well mastered track or some reference material and listen for a minute, and then back to your preset.

Knob jitter? Give it a squirt of deoxit. Check that the soldering is good, and redo it if unsure.

When will they be available?

I am also a software developer. Creating a lookup table in the UI is a perfectly reasonable solution to something like this. You don't rewrite a whole chunk of the code base, you just calculate or empirically measure 64 or 128 points and make a lookup table. Takes less than 1k of memory, and no cpu use. "rewrite a whole chunk of the code base". Tell me, are you a consultant who gets paid by the hour?

I tend to leave my cab modelling on, and set to the default which matches the selected pre-amp. I don't think I've found much more interesting tones by changing cabs from the default, so I rarely bother.

"it is impossible to make the percent into dB and Hz" This is just a load of bull from line6. You can create a lookup table... 5% = 100Hz.. 10% = 500hz 20%=1kHz. You could figure this out with white noise and an analyser (I think even meambobo did something like this already?) Who cares if it's slightly off, it'd be a lot better than mystery numbers. Line6 are flat out lazy here, and we should NOT accept this level of incompetence.

AFAIK the looper does not sync in any way to midi clock. Only the time based effects (delay) do, and then, only if you set their length to be a note number. Many will be using milliseconds by default. 1) I don't know about send. It receives automatically if you put clock messages in the midi port. 2) Use the midi IN/OUT interfaces. Line6 were lazy and did not implement midi over usb. Which is a terrible shame. I guess the features looked good enough on paper. 3) No. The time based effects will sync to the clock, so the echos will be synced, but the start and end points of the loop will not. 4) No. It's rubbish. Get a Boss RC looper.

I don't know much about either the tubescreamer or the exp2, but since no one else is answering... maybe the output changes the tone of the pedal, probably not, but what it certainly will change is the signal level going into following devices. So the higher you set the output level, the more you'll cause the next device in the chain to overdrive or distort.

I've had bugs when setting channels for cc's from the pedals, the pc editor is a bit weird sometimes. Works fine to set them via the pods own controls.

You can buy that fridge magnet stuff in sheets. Comes with one side already sticky, too. Just peel the backing off, stick it to a photo you cut from a magazine and put a hole in the middle. Great idea tho. Might help me organize my presets a bit better if the buttons are already laid out for certain fx :)

Like a lookup table is hard to do. Who's writing the code over there, ffs. That's the lamest, laziest excuse for lazy programming that I've ever heard. Architectural changes??! what architecture, your product is such a mess that not even your own support how how the signal chain works.

If you want to watch the pods eq in action, don't use the guitar, send white or pink noise through the pod iostead. You'll see the frequency response change more easily. This will let you see where the cutoff actually is, the slope of the filter, etc

A q filter makes the signal nasal. Thats not a side effect, thats what it does.. if a pedal doesnt make your sound big and open then stop using that pedal. Stop complaining. Next you'll be whining that turning up the distortion makes it more distorted.

So why is the fx return amplified to the max even with the dt master all the way down? I think you are incorrect on most of your points above tbh.

Might also be of interest, the pods spdif out stops working when you hook up an l6 link :p

True that! Aes/ebu is usually carrying PCM encoded audio. Single bit errors in PCM probably disappear in the background if its not a significant bit, but on the other hand you've the same statistical likelyhood of the error flipping a significant bit, producing a massive impulse. My point is that you'll never mistake that for a working cable. Analog transmission might dull your signal, suck tone, cause faint hiss, etc, but digital errors, when they occur, occur in really dramatic ways. If this occurred to me, I'd say the cable doesn't work. So if you aren't experiencing massive bangs and pops and clicks on a cheap cable, then you're going to get the very same with a much more expensive cable. Maybe how often these interruptions occur might go down on a better cable, true!

I work with digital transmission of mpeg video. Of course there may be retransmission of packets, and so long as that mechanism functions you will not receive any errors in the signal transferred. You just retransmit more packets in a crap connection, and the user is none the wiser. I think you need a damaged cable to get to the point where retransmission can't compensate (if they are even using retransmission). I got a response from l6 support once, saying that no information travels from the amp to the pod, but we were discussing audio signals then, and possibly it was a simplification. Even without retransmission, it's possible to encode redundant bits of information into the packets which help make them more robust. Here you can suffer some degree of noise in the transfer without losing anything of the signal. In both scenarios, despite damage in the transmission, the received signal is identical. If you DO lose any data in the received signal then it's a total trainwreck and you'll get a 'chirp' or very loud glitch in the audio as you can't decode it properly. Not a mild, general 'loss of tone' or subtle noise floor; an error will be glaringly obvious. This is what I mean by it working or not. Since I've not heard anything chirp artifacts, ever, with my cheapo xlr I take that to mean that the cable works well enough so no errors in the received signal ever reach the amp (although they might occur in the transmission layer, this doesn't affect the result).

It's digital, it'll work, or it won't. There is no middle ground. I'm happy with my random, bargain bin no-name xlr

My own experiments lead me to believe that you can think of the pod master as being on one end of the l6 link, and the dt master on the other. I think they are serially connected and it matters not which one you use; I use the nearest. This might be a bit different in LVM, I haven't done much with that, but I'm not getting any voodoo magic tones by tweaking one over the other, this after extensive a/b comparisons of the same riff looped and recorded with a variety of master positions.

That is very likely digital noise from a computer, possibly from the power supply. Turn off your computer, or run your laptop on batteries. Check that you don't have another computer or digital device running in the house (noise like that can spread through your ground connection from quite a distance). turn everything off one by one till you find the culprit.