innovine

Understood, good point :)

If they charged 1,000.00 for a pod everyone would think they are as good.

If that was true they would have done it by now.

Isn't volume and gain the same thing? Gain is the measure of increase of volume in an amplifier. It just so happens that most amplifiers start to distort under increased gain, due to limitations in their design. I guess its the distortion you wish to avoid, not gain..

I use a fulltone ocd, a custom rat, a bass big muff and a custom built pharaoh (not at the same time :), and I run all on front of my pod hd500. I can immediately hear the difference if I put them in th fx loop, but I don't know why (impedence?). none of the onboard effects distort the lower eod in the fuzzy, melted speaker way that my pedals do. Maybe the screamer is ok for harsh high gain metal but it just sounds thin, fizzy and brittle, and far too bright and digital for my taste. Low end muddy face melting fuzz is not the pods strong point at all

then the Looper would appear to be synced - at least it would restart with the correct timing at the start of every phrase.

 

Not really. The above only works if youcan also overlap the audio, ie, have two audio samples playing at the same time, getting mixed to one output.

 

Th problem is that, due to midi protocol, the clock messages are less accurate than the sample rate. In one second of audio, you have 44100 samples, exactly. So, if you try to loop one second of audio, the timing on the midi clock means that there might elapse 44150 samples if the clock is late (a big audible dropout) or 44050 if the clock msg arrives early. This causes 50samples of your audio to overlap or be omitted, both are also audible glitches.

 

Real loopers need to perform tmestretch (keeping the pitch constant) so the 44100 samples get played back in 44050 on one loop, or in 44150 in another. This is generally so complicated as to not be economically viable, which is why you don't see sample accurate loopers slaved to midi clock so often. Considering the amount of other issues in line6 products, implementing this is obviously way way down on their worry list.

Actually a receive midi clock in it as originated by OP would make it more usable

Indeed, that would be nice, but it is very difficult to do correctly as you need to also implement high quality timestretching for the audio. Even fancy softare wih tmestretching like ableton live is not so happy when slaved to an external midi clock.

 

It is easier and better to generate the midi clock from the audio rate, rather than the other way around. You won't notice if a midi clock is off by a fraction of a milisecond, but you sure notice the error if the audio glitches by the same amount.

I will try the Aux in and see what happens. I'm using the fx loop for effects, and I'm connected to a dt25 amp via line6 link and the spdif is unavailable when doing this. Annoying as hell because I could just spdif the dry signal into the computer otherwise.

Hi all,

Can someone help with reamping? I have my guitar getting DI'd into my mixer, which routes to the DAW for recording. I get a line level signal out again (from the mixer for live monitoring, and from the DAW during playback). But, how should I deal with impedence mismatching when I try to put this line level signal into the Guitar In on my pod hd?

/ Niall

 

, you can simply move your files over to the new laptop and keep going without missing a beat.

Sure, now watch while i just plug my scsi sampler into my laptop. and copy the win3.1 drivers over and keep going without missing a beat. I've a parallel port printer too. And a firewire soundcard. Ghosted disk images of a fat16 partition. I like my word documents from the mid ninties. My dos games all just copy over ok, no missed beats there. My 28.8k modem just keeps making its squealing noises and dialing up AOL. Lucky my childhood videos are all on betamax, and videos of my children all h.264. Won't be any missed beats there either.

Can't say I'm feeling at all secure about buying more jj's. But the noise floor with the eh's was terrible. Guess maybe I'll try one last pair of jj's but I'll be keeping a backup nearby

Ok so one of my new jj's just died in my dt25, after 4 months of very light use (an hour session a few times a week). I'm careful to wait a minute after powering on before leaving standby... I'm not playing very loudly. This is my second pair since march, am I doing something wrong or does this amp eat tubes?

I don't know about the music, but that bubble record player fills me with lust.

If you point your guitar in different directions, does the noise get better or worse? If so, definitely do shielding work on the guitar before you look into pedals.

 

Humbuckers can help, but mine still pick up noise from the dimmer lighting in my living room anyway, so they aren't foolproof. They also sound quite different to singles.

You're not using the line6 link at the same time? Cos that disables the pods spdif.

Everything the 4CM achieves can be done when connected by L6 Link.

Apart from the spdif..

 

There is another limitation, I can't remember exactly but I think its that you can't use the midi editor for the dt25 with the l6 connected. Haven't got mine hooked up so I can check. The l6 link clearly disables several features, but I agree with using it in the vast majority of cases.

Fx send is about 3db quieter, and noisey. Put the pedal on front of the pod instead, if you can

Just to nitpick.. the spdif output can provide the clean guitar signal from the pod. Very useful for reamping. But the spdif deactivates when you use the l6 link. This is infuriating, but a possible reason to use 4cm, as getting the clean signal is probably more important to a recording than the integration features or the extra noise in 4cm. Then you got to re-enable the l6 link to get best quality when reamping :)

So if reverb or delay effects are placed after the amp block and mixer on the 500HD, they sound the same as if they were in the effects loop of the DT25 effects loop?

Yes, and they are placed prior to the actual fx loop. Guitar->pod effects->amp block->pod fx->line6 link->dt master vol->dt fx loop->dt power amp.

 

Note that the use of cab models and lvm may possibly change or add processing in the dt, this is a bit unclear.

Oh, and for what its worth, I frequently skip my pod and use the dt25 like a classic guitar rig. I find it much more fun! Pod is too fiddly and too many options, plus I have a couple of pedals which get me the tones I want, so no real need for pod emulations. Pod opens up lots of options but often kills my creativity and actual playing time. Ymmv of course but if you find yourself constantly moving fx around and not playing guitar, disconnect the damn thing for a while.

Is the phaser before the overdrives nice? I thought its a rule of thumb to have such things after the distotions. Haven't used it myself..

If you can survive without the wah, I'd be tempted to put 1-5 on front of the pod and so skip its fx loop.

 

If you have money for more gear, adding an eq before and/or after the overdrives would open up a lot of tone territory. You could investigate this by losing the wah and putting some gear before the pod, then a pod eq, then the rest of the gear in the pods fx loop, but the eqs on the pod are imho nowhere near as nice as my boss ge-7.

 

 

*** basically you have 8 presets, BUT it's still a 2 channel amp***

 

It's not quite that, since there are only four presets for channel A, and four different presets for channel B.  That's 8 presets in total, yes, but you can't assign them to the channels freely. Presets 1-4 to channel A, and presets 5-8 to channel B. It's more like having two banks of four presets, and the ability to toggle bank.

Makes no difference, they do basically the same thing.

 

If you are using a pod hd500, then it is useful to set the channel volume per patch, to level out the differences between patches, and use the master for overall volume depending on how loud you want it in the room. If you are just using the dt25, then you can balance the relative volumes between the A and B channels, and again use the master to set the overall room volume. If you are just using one channel only, then the controls are, as far as I can tell, totally identical.

 

There are a couple of people who claim otherwise, but they are neither putting up an explanation nor are they posting sound samples for analysis. From what I can gather, the channel volume attenuates the signal, and then the master attenuates the signal. The result of this is optionally passed through the fx loop and then sent to the power amp, which is always operating at full strength. If you get power amp distortion, it is by putting a higher signal into the power amp, and it doesn't matter which of the volumes you are using to achieve this, it is a simple factor of how LOUD the amp is running. (Class A operation may differ, I don't know, I never use it).

 

Can the monkey software read a serial number from the amp via software?

I tried those settings (I assume you mean 4 as in 4-outta-10) with dt25. I use jj el84 tubes which are a little louder I think, but not by much. At the mentioned settings, for voicing 1, I can hear the amp but it's not loud at all, too low for me to sing along with anyway. Adjusting the drive up a bit raises the overall volume a lot, so going from 3 to 5 on the drive brought it up into what I'd normally sing along to without a mic.

Changing voicings, all the others are very loud indeed, so much so that I'd back off the master to sing along without a mic.

 

Just realised, I've also messed with which preamp is in which slot, so could be a bit off the defaults... but in summary some amp models are very quiet, and some are very loud at the same knob positions, but you should be able to hear something at least.