edstar1960

1. In a mono amp patch where only one path is needed, the provision of two available paths doubles the available signal which is then summed at the mixer block. To avoid unintentional signal boosting in that scenario you have to counter the doubling effect of splitting the single path into two separate paths, and the simplest way of doing that for a mono patch but still maintain the ability to have a stereo output to two amps or a mixer, is by muting the duplicated L+R signals introduced by the path split. This emulates what would happen in a real world single amp set up where you would not have a splitter giving two paths from the amp and your post fx. However, there is no reason why you cannot choose to centre the pan controls and utilise the boosted signal output if you prefer the tone produced. 2. If you just want to reduce gain to reduce clipping or to even out volumes across patches then sure you can use the dedicated mixer gain faders. In fact for a simple mono patch, I guess cantering the pans and setting gain faders to -6 db would produce the same effect as the default mixer setting in terms of overall volume. However, why bother when for a mono patch you won't be losing any audio details as each L+R channel should be identical, BUT if you use stereo fx in the signal path THEN the default should still be ok if the fx is in the PRE path but if it is placed in path A or B then you WILL have to centre the Pan for that path to get both parts of the stereo signal if it is not followed by a mono fx block. I am sure this will be easy enough to test out if you want to be sure.

Correct. It does not attenuate on the default settings but it only provides half the available signals, ie: just L from path A and just R from path B. cantering the pan controls will give L+R from both path A and B, which is twice the amount. The mixer can be used to attenuate the signal strength by using the volume controls for path A and B, so for a dual amp path a hot mixed signal can be adjusted down if necessary.

I just thought of a better way of explaining the reason for the default mixer pan settings. It is effectively dealing with the same situation that a mono fx block has in the PRE path. Summing two inputs and producing one output. The mono fx block is hard coded to sum and then attenuate so you get half the output signal. The mixer default settings essentially do the same. It has two paths coming in and only one going out so the default mixer settings mute half of the input signals. For a standard mono patch that is the correct logic to apply to maintain the same signal level throughout. Obviously we can adjust the balance of the two stereo paths in the mixer according to taste and what amps and fx we have deployed in the patch.

I agree its good to bring this up for discussion and it is good to experiment and end up with a tone that you like. I was just trying to explain why I think Line 6 set the defaults as they did, which is to keep signal level constant and not inadvertently double it and also to maintain stereo separation for any stereo fx that may get deployed. The defaults are a good place to start and once we understand what each component does we should experiment and adjust things to get the tone we are happy with. :) From peraperas schematics and my own experiments I think that the MIXER is there to mix together the dual stereo Paths A and B. The default values for the PAN settings provide half the available signal paths to the POST path, which effectively balances the signals to what was in the PRE path (ie: one L and one R). The split into paths A and B is where the signals get doubled, because you get two L+R stereo paths. Adjusting the default PAN settings so you maintain both L+R signals from both A and B effectively means you have doubled the signal strength (ie: two stereo paths summed instead of one). The end result may well be pleasing when compared to the default and depending on our own personal tastes. This is certainly something everyone should try with their Mono patches to see if they prefer the resultant tone. :)

There are plenty of full range powered speakers out there. You can spend a little or a huge amount, depending on whether you go for budget or top end. Many people on these forums have raved about the stagesource speakers, so if your budget allows I would suggest you try the L2m which is the cheapest and smallest of the Line6 options. If you want something cheaper, then try the Behringer B210d or B212d. There are plenty of other options as well.

The default values for the mixer PAN A and B are set 100% L and 100% R to ensure you do not double the signal level output. These values ensure the signal level returns to what it was before the Path A and B split by keeping just one L and R signal rather than two L and R signals. From perapera routing schematic, there is a single L+R path for PRE. This is split into two L+R paths, one for Path A and one for Path B. The mixer then allows you to blend the Path A and B L+R paths as you wish. Default keeps just L of Path A and R of Path B providing a single L+R signal for the POST path section. Changing to 0 keeps both L+R of both Path A and B which is double the signal level. For simple one amp scenarios the default setting is correct to maintain the signal strength. Of course if you start using two amps and using stereo effects in the path A and B then you will need to preserve both L+R from paths containing stereo signals. You will also have to work out what is happening to your signal strength and adjust subsequent gains accordingly. The mixer has a very wide range for negative values presumably to allow users to attenuate signals down to adjust for combining two paths before the POST path effects. Once again we can use whatever values we like to get the tone we desire. If increasing the volume at the mixer stage by changing the PAN values to 0 is what works for you, then great, but I don't think it should be applied to every patch without first considering why you might need to do it, and what it is actually doing regarding mixing the two L+R paths from A and B, and in many cases it will just be doubling up a mono signal.

Thanks! I tried this and it really gives a much fuller sound. :)

It depends which effects you are using and what you want to do with them. For a basic one amp setup, I would place effects as if it was a physical setup. So if I was using a distortion pedal or chorus or flanger, I would place it on the common pre path before the amp. I would then place effects like reverb and delay in the post path after the mixer. I have not had need to place effects post amp but before the mixer in paths A or B, but there are some demo patches that do this. Also, if you ever set up a patch where you want to use some effects that don't use an amp or if you want to blend variax model (eg: acoustic)and mags sounds and utilise the different Inputs, you may wish to have different effects on paths A (pre amp) and B with no amp, which are then MIXED at the mixer stage. The routing options are very flexible but you don't have to make use of them if you don't need to. If you just need one path then just use one path reflecting a typical physical setup - or at least start with that and then when you are confident you understand how things work together - then you can start experimenting and put things anywhere to get the tone you want - but be careful, if you don't understand the routing or how an effect will behave when placed in different positions then you may get very undesirable results or results you did not expect such as not getting the desired effect output or clipping etc etc. Hope that helps.

For input settings simply use VARIAX for input 1 and SAME for input 2. If you find that those settings are overdriving your amp model too much and you want a cleaner sound then try setting input 2 to GUITAR (which will reduce the input gain by 6db) and see if you prefer the sound. Keep whatever you prefer. Centering the Mixer on both L and R channels will effectively double the output volume, but as long as that is not overdriving the stagewedge or FOH mixer then that is not an issue, so you can continue doing that.

Yes, as a standby/backup and also, as I have a JTV59, I would like the JTV69 for the trem but also because you are able to replace the neck with whatever neck you would prefer.

I think this will be due to the fact that the DT25 will be using a topology based on whatever you have chosen for AMP A on the HD500 - so it will be using the physical relays and tubes in the amp. The FOH feed will just be using the modelled output of whatever amp chosen. The different topologies on the DT25 do have widely different volumes - for example, topology 1 is much quieter than topology 4, even if you max it out on channel volume. The direct model sounds won't be affected by the difference in physical topologies.

Yes you can. Once on your chosen patch - navigate to the setup screens - ensure INPUT screen selects VARIAX as at least one of the options - and then scroll through till you find the Variax screen. Change the setting from DONT FORCE. That will then allow you to pick a model on the Variax and will show it on the display - or you can scroll through the models from the HD500X. Once you have the one you want you can also set the TONE knob value if you don't want it at 100% for that patch. Hit save and it will remember the setting and change it whenever you hit that patch. EDIT: PS: Forgot to mention that you can also select the tuning for the model and save that with the patch too.

I raised a ticket with L6 Support and they came back with advice to try re-installing the firmware first and if that did not fix it then it would need to be sent in to their service centre to take a look. As I am now out of warranty, sending it in will be an expensive exercise, so I will only do that if it becomes a persistent problem and affects me when I am connected to the HD500 via VDI as that is my usual setup. For now I can live with it being a bit flakey for the rare occasion I use it with mags stand alone. But first, I guess it is worth trying the firmware re-install, although I am concerned it may go wrong and cause more problems and more importantly, I don't see how the firmware can affect the mags when modelling is switched off as is the case with this problem. I may see if I can get a local guitar tech to take a look as it may just need a contact cleaning or something very simple to resolve.

I haven't noticed a buzz with mine but then again I have not stood behind it with my head near the grill, but mine does produce a mains hum as soon as it's switched on while still in standby, I guess that its just the transformer. I also have a Fender Blues reissue and that doesn't make any noise when it's switched on - well none that I have noticed - neither hum or buzz. I think all tube amps are prone to some level of "noise" and I think even the same make/models can differ between units. Having said that, I am no expert and have not owned many tube amps, so I will leave the question for more experienced tube amp users to chime in with their answers.

5A fuse is standard for a guitar amp. My DT25 ordered from a UK supplier came with an IEC power cord protected by a 5A fuse. The fuse protects the device that is plugged in and is normally rated at a level that allows the device to take the power it needs but protect it from getting too much power if a fault occurs - so if there was a power surge then the fuse blows. A 13A fuse is designed for devices like kettles and cookers that draw a lot of power to heat something up. A 3A fuse is designed for devices like lighting lamps that take very little power and 5A fuses are for things like amps and TVs. Options 1, 3 and 4 will work for you BUT option 2 is incorrect as it has the wrong connector for the back of the amp - the correct amp connector is shown in option 1. You can use a lead that has a 13A fuse in it because the amp won't draw more power than it needs BUT you run the risk of a power surge not being stopped at the plug and then blowing internal fuses in the amp and possibly damaging the internals or worse causing it to catch fire. It is much safer to use the correctly rated fuse, so I recommend using 5A fuse for the DT25. The first link above is for an IEC lead that comes with a 13A fuse, but it looks like it can easily be swapped out for a 5A, so I would probably go for that option and swap the fuse when it arrives. Hope that clarifies things for you.

Any standard amp "kettle" IEC lead will work. Just ensure it is one with a 5A rating and NOT a 13A which is what you would use for an electric kettle! If you ordered it from Thomann, then they will most likely just send you an adaptor so you can use their European supplied 2 prong mains plug in the UK 3 prong socket. I had a similar incident with another product and that is what they did for me.

Hi chuberto - firstly, thank you very much for sharing this idea with the rest of us, it is very much appreciated! :) If it works and it produces the tone and sound that you want then that is all that matters. It is the end result that is important. :) Secondly, understanding how you get the end result is necessary if you don't like it and want to tweak it, or if it is bad and you need to fix it. You are happy so there is no problem for you. Which is great! :) Just for clarity, it is important to note that the illustrated L6 LINK AUDIO screen instructs the patch to send the output from both L and R channels summed to MONO to the DT25, therefore, there is no need to pan both amps paths A and B to L. You can leave the default MIXER settings (which are; path A panned hard L and path B panned hard R) and you will still get both amps output to the DT25. Note: If you use the patch defined with both paths panned hard L, ensure any stereo fx used is summed to MONO by choosing appropriate fx chain order OR change the patch to use default mixer settings (A hard L and B hard R), then you won't be missing any parts from the output signal. EDIT; Just for clarity, I should not really use the word panning although the mixer controls are labelled PAN A and PAN B, as the mixer does not pan the signal from L to R, instead it adjusts the BALANCE or VOLUME of both L and R channels from both A and B paths. 100% L means you only hear the L channel summed from both A+B paths and the R is muted. 100% R means you only hear the R channel summed from both A+B paths and L is muted. A value of 0% will mean you hear BOTH L + R equally summed from paths A+B which means you also get double the output volume! In between 100% to 0% for either L or R means you get a mix of both L+R channels summed from both A+B paths weighted according to the channel and value chosen.

It looks like the screens are showing default settings for L6 LINK AUDIO and CONTROL. If there is only one DT25 amp connected then the settings for L6 LINK AUDIO are indicating that output from both the LEFT and RIGHT channels of the mixer stage will be sent to the DT25 (1: AMP) L6 LINK CONTROL shows that the connected DT25 will set it's TOPOLOGY to follow the amp selected as AMP A on the HD500. The MIXER settings are indicating that only the LEFT channel will be sent as it is set to 100% LEFT thereby muting the RIGHT channel output - so you will only be getting HALF the output signal from the amp. Remember both amps from path A and B will output STEREO L/R signals and these get split at the MIXER to the output L/R. It does not make any sense to me to mute the RIGHT channel for the above patch. From what I can see - the DT25 will use one TOPOLOGY matching AMP A on the HD500 (in this case JTM45) - and you can then flip flop between different pre-amps using the patch by using the VOLUME pedal switch trick to mute one of the pre-amps.

I bought a DT25 cover from ROQSOLID who are based in the UK - they are very good. Quick service and good hard wearing heavy duty covers. You can ask them to make custom made covers just by providing dimensions or you can buy one of their ready made ones for popular amps - and they cover the Line6 range. Their website is here: http://www.roqsolid.co.uk/wp/ But you order your amp covers from here: http://www.amplifiercoversonline.com/zencart/index.php And I did a search for DT50 and they do provide ready made one covers for DT50 range including 212: http://amplifiercoversonline.com/zencart/index.php?main_page=advanced_search_result&search_in_description=1&zenid=14fbup42bt1hicqiuuqs8tadu6&keyword=DT50&x=32&y=6

Thanks for your input. So it could possibly be a bad pickup. Sorry to hear that you have had problems and that the guitar is going back for a 4th time. Well, that is 3 of us, so I am not hitting a completely unknown problem. Time for me to raise a Support Ticket and see what L6 have to say ...

I read the release notes for latest firmware 1.2 for HD500X and it clearly states that if you connect the HD500X to Stagesource speaker that it will default to Studio/Direct output mode, and the Stagesource speaker will default to FRFR mode. However, it does not say what happens when you connect to a DT25 first and then to a Stagesource speaker. If you connect your HD500X direct to L3t does it work as described in the 1.2 release notes? And then when you connect it via DT25 does it behave as you described above? That is, stays on either Elec Guitar mode or whatever you manually swap it to? Are you using standard XLR cables for your L6 link connections or are you using the recommended AES cables? Also, when you have it connected as above, did you try changing hd500 to acoustic patch an then manually swapping l3t to acoustic guitar mode and THEN saving the hd500 patch. Then swap to different patch for elec guitar, see if l3t changes mode, if not then manually change to required mode and THEN save that patch. Then retun to each patch on hd500 using the foot switches as normal and see if that now auto switches the l3t made with patch change.

Thank you once again gckelloch. Here is a useful link for hearing safety and appropriate db levels: http://www.gcaudio.com/resources/howtos/loudness.html And this: http://www.noisehelp.com/noise-dose.html

I thought that the GLOBAL STRING level was just a way to get a base level for all models in one place and that its there to allow you to control any discrepancies between the output volumes of each piezo on each string. I know someone mentioned in some post that piezo outputs can vary by as much as 25% so there is obviously a need to adjust your string volume globally for each individual JTV. Then after that, if you want to change the characteristics of any particular model, you can then adjust the string volumes for that model only, and those adjustments would be relative to the base level set by the GLOBAL STRING volumes. So I think they apply the global settings first and then apply any adjustments that may have been made to string volume for each model. That's what makes sense to me, but I don't know for sure if that is exactly how it is implemented in the JTV.

Thanks for the reply - glad to know that I am not on my own with this problem. Have you raised a support ticket with Line6 to see what they say? Thanks.

Thanks for the advice! I will give that a try!