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saTa

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Everything posted by saTa

  1. I'm afraid this is by design. Bad design. I've used a DT25 for two years and ended up never using Topology I because I would have had to save all my other presets at like 20% volume to match loudness...
  2. It may be a great product, but do you remember the hype? "THE GUITAR AMP, REINVENTED"... really?
  3. After amplifi, I can't take L6 announcements seriously anymore. Let's remain calm and wait and see.
  4. Another thing most people probably don't do but that's possible with the POD: You can use its stereo signal chain to make the Pedal an FX send... If you pack a bunch of FX into the left channel, your dry signal into the right, you can put a volume pedal onto the beginning of the split and only send certain notes of a solo into the delay, for example. I do this a lot. For most applications, you could just assign the FX mix to the pedal, of course, but say you want that last note of the solo ring for another 3-4 bars while you go on playing dry, the "send" is the way to do it.
  5. I use it a lot for delay feedback and mix... If you want to use it for the tone knob on your guitar then probably your best bet is to add an EQ with a low pass filter at the beginning of your chain and to map that LPF frequency to your pedal.
  6. Makes me want to give the 800 model another go... I usually go for the Park 75 if I'm shooting for this kind of sound.
  7. After dabbling about with both extremes, I settled at having both at around 12 o clock as a starting point, depending on surroundings... It's nice to have direct control without walking over to the amp (hence not to have the hd dimed) and it gives me enough power stage breakup for the sounds I'm after.
  8. Yes. Unfortunately, Topology I is virtually useless unless you save all your other sounds at about 20-30% volume. This really needs addressing but it has been this way for years, so don't expect it to change. :(
  9. It's one of my favourite features about the DT25... when we play shows where all we get is a 15min line check I'll regularly tell mixers not to bother with mic positioning and just throw me a cable. During a recent live recording session, I carefully placed a SM57 and a MD421 in front of the DT25... just to be safe, I tracked the DI as well - it compares very well against the two.
  10. 1. Make sure you don't use "same" or "Guitar" as your second input but something else 2. Turn down the Master Volume in the deep edit settings. For some reasons this is always at 100% for all models which almost never sounds good. Dial that back and hear virtually every sound clear up nicely.
  11. I went straight to PA for years and now use a DT25 on stage. I love playing through the DT25 and I probably wouldn't use another amp that wasn't made for use with the POD like the DT25 is. Going to PA is fine and probably better than going through a less than optimal amplifier.
  12. Nevermind... this solved it... it was the output mode... http://line6.com/support/topic/3666-horrible-buzzing-sound-for-most-patches/
  13. Update: I just reinstalled the latest firmware on the HD500 and made sure to overwrite all the sounds on it, so the factory presets are fresh from Line6. It sounds different but still useless, see attached mp3. I'm confused. hd500afterreset.mp3
  14. Hi guys, I've been using a HD500 ever since they first came out. Usually, I use it live with my DT25 and everything seems fine. However, I've been taking it home to do some recording and couldn't for the life of me get it to sound right. The distortion sounds absolutely horrible and I don't remember it being this bad. After a lot of tweaking, I decided to get a HD bean. I've been thinking of getting one forever to save me from lugging the HD500 around all the time or being unable to record ideas and tweak sounds at home. (As a sidenote, I wish getting sounds from HD500 to HD bean and back were easier but that's for another thread...) I just plugged in the HD bean and installed the latest firmware onto it. Then I tried some factory presets and holy lollipop it sounds amazing! I was unable to put the guitar down for an hour or so and it left me completely confused. When I tried the same factory preset on my HD500, using the same guitar, the same cables and making sure the presets were exactly the same (see screenshot), it sounded horrid... thin and unpleasant distortion... I have no idea what's wrong and what I should do? Can you guys point me in a direction? Attached you find the screenshot of both Doctor Crunch presets, as well as an mp3... it shouldn't be hard to figure out which is which and that something's very wrong here! I've also attached both presets just to be sure... but they're factory presets so you guys should have them. Oh... and this happens across the board with all sounds. Even when I build a new sound from scratch and put nothing but an amp in there, the HD500 sounds completely terrible. :( hdvshd500.mp3 Doctor Crunch.zip
  15. Cheers... but it was a strange software issue. Re-installing the firmware and resetting the amp did not work. However, when I used one of the DT editors posted somewhere here on these forums to have a look at the parameters, I found the master knob to be WAY off. When it was turned up all the way on the actual amp, it was at 30-40% in the software. So I cranked it in the software and then moved it on the amp. Now they're back in sync and the amp behaves as expected. Very annoying and many thanks to the guys putting out these DT editors!
  16. I have the same problem and resets don't fix it. Used to have it at 3 in the rehearsal room and couldn't hear myself at 10 today. However, I did install the new 2.20 HD500 flash memory and am thinking that that might have messed it up as well. Will do some more testing tomorrow. :(
  17. I'm not sure this has to do with the update or if something's broken... but all my presets sound completely different (much cleaner and quieter) than before. I have to turn the Master on my DT25 up all the way and still can't hear myself in the rehearsal room whereas it used to be fine at around 35-40% up. Am I wrong in assuming that it all should sound exactly the same with the master on the HD500 all the way up? Well, it doesn't... :(
  18. Any news on this? I just broke my 2nd PSU (DC 3-G, the stock HD500 one) while on the road and I'm frustrated to a degree where I'm researching alternatives to my HD500/DT-25 setup.
  19. Haha, I get the same on a B note...
  20. Thanks for this thread... I too have always switched off cab modelling on my HD500 as I figured the DT25 112 Combo was an *actual* cab. I didn't realise there were live models that the HD500 would switch to when linked via L6Link. Once again, the L6 documentation could do with some improving... I'll give this a shot over the weekend and see what happens!
  21. I sync my HD500 with MIDI to get timed delay fx and mostly it's fine for live use... When recording, I've had to scratch a bunch of otherwise perfect takes though because it does drift and then readjust audibly.
  22. Nothing constructive to contribute but please keep us posted... the PSU is a flimsy POS.
  23. Just to add to the very accurate explanation by DeanDinosaur: Compressors usually have five variables: - Threshold level: Signal that is louder than this specified level is compressed, that is to say turned down - Attack time: Determines how fast it is turned down. Long attack times let transients (picking sound, attack) slip through before turning down the volume, fast attack times can make the track sound flat - Release time: Determines how fast the compressor opens up again after turning down the signal - Ratio: Determines by how much the signal is turned down. - Makeup gain: Global gain to make up for the volume loss incurred by compression (that's right, compressors make the signal more QUIET at first) Example: Threshold is -10dB, Attack is 10ms, Release is 100ms, Ratio is 2:1 Signal coming in at -12dB will remain untouched Signal coming in at -8dB is 2dB above the threshold of -10dB. Ratio of 2:1 determines that the signal is turned down by 1dB meaning the output is -9dB instead of -8. Signal coming in at -2dB is 8dB above the threshold of -10dB... meaning it'll go out at -6dB instead of -2. So, like DeanDinosaur has said, compressors reduce the dynamic range of your signal... pushing down the spikes leaves more headroom so they're used to increase the overall loudness of a track. For guitarists, they're usually used to make your signal more even... examples: less spikey sound when strumming clean chords because the sustain is closer to the pick attack in loudness... more even transitions between solo/melody parts and rhythm chords... increased sustain (if louder signal is turned down the perceived decay time is increased)... and so on. :)
  24. You could also bypass all the fx blocks and then add them back in from left to right while monitoring the output... put simply, the concept of "gain staging" suggests that each gain stage in a signal chain (comps, eqs, pres etc.) should make the signal louder - never drastically turn it down. So if after your stomp box the signal is much louder than after the next effect, tweak the stomp box output. If you stick to this, SNR will be optimal and digital clipping avoided.
  25. What phil_m is saying is: If clipping occurs within your POD, it won't necessarily be reflected by your DAW. It is important to distinguish clipping from signals hitting 0dbfs. Yes, hitting the 0dbfs mark will clip your signal but if you turn it down by 2dB afterwards, the next stage won't recognise that because the signal has been clipped and then turned down. Your EQ would then show up peaks at -2dB and everything would look fine but still sound digitally distorted. As a general rule, clipping is an issue that should have a priority much higher than SNR with digital devices such as the HD500. You could easily track all your guitars with peaks at -20dB and not have to worry about noise... tracking hot, that is to say, as loudly as possible without going into the red if a left-over from analogue days that doesn't really apply to nowadays practice anymore... but that's another subject altogether. But even then, it had more to do with the saturation-behaviour of analogue recording equipment rather than SNR.
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