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I posted this in the old forum months ago,

since many peole found it useful I thought I could re-post it here:

___

 

I really think these are important things not stated (or in some cases not clearly stated) in the manual, that you need to know to start seriously programming your POD HD
 

I ran some serious tests on my Pod HD 500 and here is what I found out (I made my tests on a POD HD500 but this is valid for HD500X and HD Pro and is very similar for the "bean" version too),

I'll try to be as synthetic (but complete and clear) as possible, but this is going to be a long reading so sit down and take your time or just go surfing somewhere else

 

§§§
 

In short: the "famous" [input-1: guitar / input-2: variax] setting gives you different levels of signal depending on the position of the first *mono* effect block you use (amplifiers included), in particular you LOSE 6dB in the "pre" path in comparison to the path A/B or post path

(note that this is not the same as saying that you gain 6dB with input-2 to "same", read on).

 

[ if you don't know what I'm talking about just go and read this thread:

http://line6.com/support/thread/74045?start=0&tstart=0

then come back here to hear a different opinion on the matter]

 

first of all, try it out:

- connect a guitar to the guitar input and the Left output to a full-range linear amp (or use your headphones)

- recall a "new tone" default blank patch

- set input-1 to Guitar and input-2 to Variax

- set mixer channel A fader to unity (0.0dB) and pan to center

- set mixer channel B fader to mute

- setup a noise gate* with the threshold set to 0% in "pre" position

(with this setting this IS a unity gain mono fx block)

- play thru it

- now if you bypass it, you'll hear that it looses 6dB of level when it's active (I initially thought this was noise gate's fault, but it's NOT)

- now re-activate the noise gate and move it in A or "post" path

- now if you try to bypass it you'll hear that it does NOT loose any dB

- try moving the block back and forth between pre and A or post paths and you'll hear more level in path A or post than in pre

 

this was already found out at least by hurghanico here: http://line6.com/support/message/403287#403287

but it's so important that needs a dedicated and more detailed thread.

 

[* you can repeat the experiment with other mono effects instead of a noise gate but keep in mind that, if you want to clearly hear a level difference, you need a mono unity gain (www.music-dictionary.org/unity_gain) effect, for example:

- a tube comp with thresh 100% & level 2% settings will work just as the noise gate above

- an fx loop block with a mono cable connected between send and return will work just the same (but also read point 2 below)

- do it with an amp with medium-low gain and, moving it between pre and A or post paths, you'll hear a significant difference in gain/ovedrive/distortion, not only level difference]

 

§§§
 

OK now that you heard it, let's see it in detail;

 

these are the REAL schemes of the pod and fx blocks routing, yes it's done by hand and I love it ;-)

 

pod_routing_%26_fx_blocks_graphs.png

 

As you can see the pre path is a "dual-path"

while A, B and post are all stereo paths;

at the splitting point, where the path A and B are born,

the signal coming from input-1 is splittted to the Left and Right channels of the path A

and the signal coming from input-2 is splittted to the Left and Right channels of the path B;

 

furthermore all fx blocks have TWO inputs and two outputs and the mono blocks do attenuate by 6dB and sum their inputs, then process the result and then split their mono output to both outputs of the block;

 

for those who don't know, notice that:

- "splitting" means duplicating one mono signal to two "routes"

- and summing those two identical signals means doubling the level of the original signal (which equals to 6dB more)

 

[and some side-notes:

- the "stereo dry & mono wet" effects are for example the pitch effects and the "dry" type delays, I'm not considering this type of effects in this post, but they work as expected from the scheme you see above;

- you can find a list of all the fx blocks divided by type here:

http://line6.com/support/page/kb/_/pod/pod-hd/stereomono-fx-list-for-pod-hd-r567

where "stereo dry & mono wet" blocks are called "Stereo Thru/Mono Effect" which I personally find less clear

- the mixer control named as "pan" is actually a "balance" control because if you move it to one side (e.g.: left) it acts on the stereo or dual mono signal by doing NOTHING on that side (left) and ATTENUATING the opposite side (right)]

 

§§§
 

So, summarizing, if you only activate input-1, in the pre path, the first mono effect is attenuating the input 1 and 2 and summing them, but, since input-2 is actually silence, you loose 6dB;

in A, B and post paths the effects are receiving a doubled signal on L/R, so the mono blocks, attenuating and summing the two signals, receive the right signal level to process

 

so using "same" or "guitar" for input-2 does not mean to gain anything, but having a constant doubled signal wich is compensated by a 6dB attenuation in each mono summing it encounters in his flow

please note that I am NOT saying that using only input-1 is wrong, you just need to know that this can give you different gain results depending on the position of the first mono effect

 

with only input-1 active (Guitar/Variax) and the same parameter values, this:

rout-1.png

 

is giving you more distortion than that:

rout-2.png

 

now, if you use those two setups with "Input-1: Guitar / Input-2: Same", you get EXACTLY the same sound with both

 

and this is something that can not be ignored

...don't know how to be more clear than that

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Good post. The unholy mess that is POD HD signal routing drove me potty for the first year of owning the thing.

 

I gradually worked out how best to use it - for me. As I never bother with dual-amp patches I put all effects and amps in the A path and mute the B path from the mixer. The mixer stays at the end of the chain with all effects before it.

 

As your routing guide shows this is a much more consistent approach. L6 made quite a mess of the signal routing in POD HD and back when the company could be bothered to post on the forums gave the impression it didn't understand what was happening. Which implies the signal routing wasn't exactly planned this way.

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  • 1 month later...

 

unfortunately the link above does not work anymore, but thanks for quoting me ;)

 

yes, they recently moved the old forum and didn't bother to set up some kind of redirection to make the old links work...

(of course when I posted this it was working...)

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i am meambobbo and I approve this post!

 

another main thing to take note of -> every "line" in the onboard display, or in the HD Edit software on a computer, is a STEREO/DUAL signal.  Towards the beginning of the signal, it's connected to the inputs, so it can seem to be more about dual mono tones, or getting the levels right for your needs, as said above.  If you are using a single amp, it's all going to get mixed down to mono.  Past the amp block and where you'd typically use stereo effects, the lines are more like a traditional stereo signal.  But for dual amp tones, or times when you want to do a wet/dry thing with the two channels, it's important to realize EACH channel is also stereo.  So it's 4 distinct signals at that point.  You can take advantage of that by sending one side out via the fx loop send, which is actually a TRS connector, and can be separated out to stereo.  Or do cab/mic on one side and not the other, and send the no cab version to a real power amp and cab.

 

Also, the "pan" control on the mixer, is more accurately called "balance".  full L or R isn't pushing both of the stereo signals into one of the outs; it's muting the other field.  Full L means Mute R.

 

here's my guide on routing:

http://foobazaar.com/podhd/toneGuide/setup#routing

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i am meambobbo and I approve this post!

 

another main thing to take note of -> every "line" in the onboard display, or in the HD Edit software on a computer, is a STEREO/DUAL signal.  Towards the beginning of the signal, it's connected to the inputs, so it can seem to be more about dual mono tones, or getting the levels right for your needs, as said above.  If you are using a single amp, it's all going to get mixed down to mono.  Past the amp block and where you'd typically use stereo effects, the lines are more like a traditional stereo signal.  But for dual amp tones, or times when you want to do a wet/dry thing with the two channels, it's important to realize EACH channel is also stereo.  So it's 4 distinct signals at that point.  You can take advantage of that by sending one side out via the fx loop send, which is actually a TRS connector, and can be separated out to stereo.  Or do cab/mic on one side and not the other, and send the no cab version to a real power amp and cab.

 

Also, the "pan" control on the mixer, is more accurately called "balance".  full L or R isn't pushing both of the stereo signals into one of the outs; it's muting the other field.  Full L means Mute R.

 

here's my guide on routing:

http://foobazaar.com/podhd/toneGuide/setup#routing

 

thanks Bobbo!

 

you're right, in the pod LCD and Editor screen the single lines indicate stereo/dual signals;

 

I'm writing this note just to clarify and help people who are reading my schematics and these posts:

all meambobbo wrote in his post above is written and drawn in my original post/schematics

 

in my schemes above every line is a single signal, so they are a lot more accurate than the Edit software or LCD representation, that's one of the reasons I drawn them.

 

just by understanding and reading them you can really master the routing inside and outside the pod,

in fact I personally look at them everytime I need to do some complex routing

 

following the signal through them you can come to a lot of conclusions that unfortunately are NOT explained in the manuals including the so called "advanced" guide

 

peace

Lore

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yes,brue58ski

correct

if you connect to guitar input and leave the default input setting (everything/same) [or maybe opt for guitar/same to avoid any possible noise from the mic in]

you have nothing to worry

 

yes, if you don't want the 6dB loss you need the inputs to be the same, except when you use a separate path A (or/and B )

 

what hurghanico adds is that if someone wants to use two instruments/sources with two separate paths

they of course have to set different sources in the inputs settings (e.g. guitar/mic),

and they also have to avoid mono effects in the pre-path or the separation is gone...

 

this way they do not have to worry about any signal loss too

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  • 1 month later...

Great post.  I've been running my HD500 into the FX in on my amp and I noticed that an empty patch would get quieter as soon as you added an amp into the signal chain (in the default position).   I want to go back and check how 

my inputs are setup and see if they are "1:Guitar 2:Variax" or "1:Guitar, 2:SAME".

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thanks guys!


it's good to know that I can be helpful to you!


 


and to sjnewbold:


 


check the input settings, but I don't think this is the case:


the bypassed-amp vs active-amp level match is independent from the input settings


and depends on the "channel volume" of the amp model (of course the master volume and drive affect the level too, but they also affect the tone)


 


anyway, in my opinion, it's not something so important to care about: normally you do not "bypass" your real amp ;-)


 


in your "guitar > pod > amp return" configuration the only thing that matters level-wise is that the return gets the right level for the power amp to work correctly (expecially if it's a valve power amp and you want to push it to get it's sound)


 


so you just have to roughly match the level of your amp's preamp with the pod (with an amp active)


 


example:


 


- match a clean pod amp level with your amp's preamp clean channel


(to do so use the channel volume of the amp model and the pod's analog output level and 1/4" out switch)


 


- do the same with a crunch and lead tone (using only the amp model channel volume)


 


- experiment with the output settings (Combo or Stack should be the way to go but only your ears will tell you)


 


- then start building your patches in relation to that level


 


bye


Lore


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Hello "Perapera",
I have been translating almost all Line 6 manuals from English into my native Czech language for local country Line 6 distributor.
I would like to use your schematics for the Czech manuals of POD HD 500/X/Pro family.

I beg for your kind permission for sharing the schematics.

Have a nice time. 
cz-milan

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Hi Milan

wow, I'm really pleased by your offer!

yes of course you can

You just have to mention me as the author

I think you can write <<original schematics by Lorenzo Sempio, "perapera" on the Line 6 forums>>

 

I'm particularly pleased cause I've been many times in czech touring as a sound engineer

and I found many kind people, friends and music lovers

 

ahoy!

Lorenzo

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Thanks very much for posting this very useful information. I am still struggling to get my head around the routing of the HD500, but this thread certainly explains a lot of things and why certain things I have encountered have occurred.

 

I have a question which I can't figure the answer to.   When creating a new patch or when tweaking an factory provided setting, I have noticed that the MIXER is set as 100% LEFT and 100% RIGHT.  As I play live in MONO, I take both the L and R XLR outputs with a Y lead and connect them to a single XLR lead to run to the mixer and then on to a powered speaker.  I always struggled to get my sound to punch through in a band setting and discovered that if I used MIXER values of 0% for both L and R values that I would get a louder fatter sound than using the default of 100% LEFT and 100% RIGHT, even though I was manually summing the signals with my Y lead external to the HD500, the 0% settings produce a fatter sound and drive the speaker better (or so I thought).  I had often wondered why this was the case because I thought both scenarios should effectively be the same but from your diagrams it would seem that using the default values (100% L+R) means you MUTE the other half of the signal from that path, but using % for both means you keep both L+R signals from both paths, which means I am doubling the signal output.   Is that correct? 

 

What is the recommended/best way to use the mixer?  Is it best to use the defaults 100% L+R or to modify to 0% L+R?    Which option will produce the best final output sound?  

I am concerned that what I thought was an obvious way of getting a stronger signal (ie: 0% L+R) is just producing a level that is overpowering the end of the signal chain and therefore creating a harsher tone rather than a fatter warmer tone that I am trying to create, and therefore, maybe I should be going back to the default setting to get a better end sound.  Your advice would be greatly appreciated.  Thanks in advance.

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For a MONO setup - I would also like to understand the difference between placing the AMP before the SPLIT Paths and placing the AMP after SPLIT paths after the mixer?

 

I experimented this morning, trying the same amp before and after the split paths and also with mixer L+R values at 100% and then at 0%.

With amp before and mixer L+R at 0%, I get lots of volume. If I swap mixer L+R to 100% then I get reduced volume.

If I move amp to after mixer and have L+R at 0% I get reduced volume in comparison to having amp before split.

If I move amp to after mixer and have L+R at 100% then volume seems to be similar but the amp gets driven more.

 

Again, I would appreciate recommendations for where is best position to place amp for a mono setup.  Where will it provide most realistic (best) modelled sound?  Default seems to have amp always in front of split path - so is that the ideal position?  Is that scenario where the best sound will be generated?

 

Thanks in advance.

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hi Milan

 

I'd like to check your re-drawings before you publish them

 

and I'd like to use them in this thread (of course with credits to you as the "digital artist" ;-) )

 

Hi Lorenzo,
 
Thank you very much for your prompt response and your kind permission. :)
 
I shall mention you as an author, of course, exactly by your wish.
 
I will redraw your sketches into publishable form.
 
Ahoy
 
Milan

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hi edstar1960


 


I'm sorry I don't have a lot of time to answer you in these days


I'll try to be as clear and synthetic as possible


 


first, you're gonna have to follow me on my schematics while you read here;


second, what I'll write here is referred ONLY to the type of mono setup you described in your posts,


which I think is the most commonly used by guitarists:


guitar > POD HD500 > mono "monitoring system" (whichever you use)


third, since you're using XLR outputs I will not consider the jack outputs


 


let's start with a configuration that uses only one amp (no fx, nothing)


starting from a default "new tone" just choose an amp model of your choice


 


what happes is that at the split point the mono signal coming from the amp is splitted in FOUR identical signals which go to the left and right channels of path A and to the left and right channels of path B


now if you leave the mixer at the default (100% L/R)


and connect for example only the left XLR to a monitoring system


you get EXACTLY the original "single" signal which is coming out of the amp model


 


if you have followed the signal in the routing schematics you should agree


 


the consequence is that, externally summing the two XLR, is giving you a doubled signal


 


(BUT I must say that "Y cables" used for summing are everything but ideal;


see here:


http://www.gearslutz.com/board/geekslutz-forum/453415-wye-not-wye-combining-signal-y-cable.html


and here


http://www.rane.com/note109.html )


 


AND if you turn the pan pots into the pod's mixer to 0% you'll get another doubling of the signal


 


now remember that our hearing system is NOT linear: an increase in level is perceived as a sound with more lows and highs and this is often described as "fuller", "warmer", "more open" but it's just level, my friend


(and of course if it gets too much it can clip and become harsh)


 


http://en.wikipedia.org/wiki/Fletcher%E2%80%93Munson_curves


_________


 


frome here, if you add effects, things can be more complicated...


...but less than one can think


 


for example (always follow me on the schematics):


 


problem: if you leave the mixer at default and put effects on path B they will output only their Right signal to only the Right XLR which in our case is disconnected


solution: turn the B pan-pot to 100% Left too!


 


this "looses" the Right signal but this could only be a problem for some effects (ping pong and stereo delays, auto-panners, etc) which a person who wants a mono signal should't use, should he?


___________


 


sorry there is no "best" way to use the mixer


I can only suggest to start from the default and experiment and listen...


 


...just like you did with the "amp pre-split vs post mixer" experiment:


- you heard that at default settings (L/R 100%), it's the same


so your choice should be based on if and where to take advantage of the parallel paths for special fx combinations


- with the the mixer's pan-pots at center (0%) if A-Left, A-Right, B-Left and B-Right are all carrying the same signal and you put the mixer's pan-pots at center (0%) you'll double the signal,


so moving the amp pre-split gives more level ONLY IF YOU SUM the L/R XLR's but not if you use the left only,


while moving the amp post mixer gives more gain (distortion)


_________


 


I'm sorry if I didn't discuss all possibilities but this should give you a lot of material to think about ;-)


 


bye


Lorenzo


 


p.s.


wow that wasn't exacly "synthetic", was it? :)


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Thank you very much Lorenzo! I really appreciate all your time and effort in giving me such a comprehensive reply.  I have clicked on the links you gave me about Y cables and now understand that I have been doing the wrong thing!  In fact I am now concerned that maybe I have damaged my gear because of this misuse of the Y cable - I hope not.  I have also read your other thread where you and hurghanico describe in great detail the difference between the balanced (XLR) and unbalanced (jack) connectors on the back of the HD500.  And I have also read through the HD500 ADVANCED GUIDE V2.10 specifically looking at detail about signal paths and also the connectors.  I think that finally I understand the signal routing paths of the HD500.   If it was not for your post here, I would never have got to the bottom of it, so thank you very much for sharing and for answering my questions!

 

Today, I will put into practice everything I have learnt, and will hopefully get a great sound out of my HD500 at GIG volumes with any gear.  I will rework my powered PA speaker connections (STOP USING Y CABLES!) and then rework my HD500 patches to work with the correct connections.  I will then repeat the process with my other patches for direct amp connections and for DT25 with L6Link to ensure the signal paths are correct.    UPDATE: Removing the Y cable and taking the L/mono jack unbalanced out from the HD500 into my mixer channel, plus on HD500 setting using Global Option STUDIO/DIRECT and for each patch setting mixer pan controls to 100% for L and 100% for R, made a big difference. Everything sounds much better already!  :)

 

Background:

Why did I start using a Y cable in the first place?

I had a V700 and an X3L.  The X3L never cut through at gigs - always sounded very thin when amplified at volume. I thought it was because I was only using one half of the output signal, so decided to try using a Y cable to capture both L+R outputs and it certainly did seem to improve things, however, it still did not provide sufficient signal level to drive my STUDIOMASTER GX12A powered speaker at sufficient volumes to compete with the band at gigs. So I then introduced a BEHRINGER B105D, and took this combined XLR L+R signal and put it into one of the mic/line inputs. I only had to boost that signal by a few db's, with the trim control on the channel set to between 8 and 9 o'clock. The THRU output from the B105D was at line level and went into the input of the GX12A and bingo, I had the gig volume I needed and depth of tone I was after, although it was still not perfect and compared to a real guitar through an amp did lack depth and punch - in other words still sounded "thinner".  I then got my JTV59 and used the same setup, and everything was improved with the better sounds produced from the JTV59.  Then I got the HD500, and it took me many months to migrate from the X3L to the HD500. Many many hours of tweaking etc., and I am still doing it now as I have just bought 2x BEHRINGER B210D active powered speakers, so needed to get the sound adjusted for them. The HD500 seemed to have an even weaker output than the X3L when using the same PA setup, so I carried forward the idea of using the Y cable to capture both L+R outputs to avoid a thin "live" sound and also found that the HD500 output still needed a bigger boost before heading to the powered PA speakers to generate the correct volumes. So, I started using my XENYX 802 for this purpose. The XLR outputs from HD500 went into a Y cable and then a single XLR went to the XENYX 802 into a mic channel with trim set to about 9 o'clock - and then a little adjustment on the HIGH and MID EQ controls, set the CHANNEL volume to 0db (12 o'clock) and the MASTER mixer output to 0db (12 o'clock) - take both L+R master outs (jack connectors) combine with yet another Y lead (oh no!) and then into a jack to XLR converter and then XLR into the B210D and then link that to the next B210D with the THRU connector and another XLR.  Both B210D's had their trim control's set to 9'oclock.  That seems to produce a nice loud volume that is OK for home usage but will not be loud enough for gigs. However, I still have the ability to turn up the MIXER channel volume, the MIXER MASTER out and the B210D trim controls which should get me there, BUT, I have not had the chance to try that yet.

What now?

After reading all your information I now realise that ALL of what I have done to overcome the "live" thin sound of the POD at GIG volumes is WRONG!  I will have to start again from scratch with the correct knowledge of how the HD500 signal routing works and work at getting a good "full" tone from the POD at gig volumes using other means! 

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hi Milan

 

I'd like to check your re-drawings before you publish them

 

and I'd like to use them in this thread (of course with credits to you as the "digital artist" ;-) )

O.K. Lorenzo,

I will try to redraw the schematics as soon as possible and give them to you at your disposal. 

Milan

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thanks edstar, I really enjoy helping when I can,

reading your story I thought:

1 - since you have many speaker oprtions why not using the pod in "stereo"?

stereo is in quotes because using both XLR outs does not mean you are in stereo

it's the kind of effect you use that create the stereo image

2 - if you are using a mixer as an additional gain stage you can avoid Y cables and connect both pod outputs to the mixer, pan them both to center

or maybe both to left for example and then take the left out of the mixer and connect the two linked speakers

3- a 12" speaker works better on guitar than a 10" speaker even if it's a PA, so if you can try that too

4- the pod XLR outputs carry a balanced microphone level signal, it's a little hot for a microfone level but it's definitely NOT a line level, it could be something like 30dB softer than the jack outputs;

the only benefit of XLR is the balancing which helps with noise interferences ON LONG cables

I mean over 10 meters, you don't need that

AND using a jack out will mix to mono internally

 

 

but, as I see from your edit, you discovered this by yourself ;-)

 

5- many guitarists know that a valve power amp has a way of treating dynamics that gives it more volume and cut-thru even with low wattage, your 200+200W system should be enough to balance maybe a 30-40W valve amp

 

Thank you very much Lorenzo! I really appreciate all your time and effort in giving me such a comprehensive reply.  I have clicked on the links you gave me about Y cables and now understand that I have been doing the wrong thing!  In fact I am now concerned that maybe I have damaged my gear because of this misuse of the Y cable - I hope not.  I have also read your other thread where you and hurghanico describe in great detail the difference between the balanced (XLR) and unbalanced (jack) connectors on the back of the HD500.  And I have also read through the HD500 ADVANCED GUIDE V2.10 specifically looking at detail about signal paths and also the connectors.  I think that finally I understand the signal routing paths of the HD500.   If it was not for your post here, I would never have got to the bottom of it, so thank you very much for sharing and for answering my questions!

 

Today, I will put into practice everything I have learnt, and will hopefully get a great sound out of my HD500 at GIG volumes with any gear.  I will rework my powered PA speaker connections (STOP USING Y CABLES!) and then rework my HD500 patches to work with the correct connections.  I will then repeat the process with my other patches for direct amp connections and for DT25 with L6Link to ensure the signal paths are correct.    UPDATE: Removing the Y cable and taking the L/mono jack unbalanced out from the HD500 into my mixer channel, plus on HD500 setting using Global Option STUDIO/DIRECT and for each patch setting mixer pan controls to 100% for L and 100% for R, made a big difference. Everything sounds much better already!  :)

 

Background:

Why did I start using a Y cable in the first place?

I had a V700 and an X3L.  The X3L never cut through at gigs - always sounded very thin when amplified at volume. I thought it was because I was only using one half of the output signal, so decided to try using a Y cable to capture both L+R outputs and it certainly did seem to improve things, however, it still did not provide sufficient signal level to drive my STUDIOMASTER GX12A powered speaker at sufficient volumes to compete with the band at gigs. So I then introduced a BEHRINGER B105D, and took this combined XLR L+R signal and put it into one of the mic/line inputs. I only had to boost that signal by a few db's, with the trim control on the channel set to between 8 and 9 o'clock. The THRU output from the B105D was at line level and went into the input of the GX12A and bingo, I had the gig volume I needed and depth of tone I was after, although it was still not perfect and compared to a real guitar through an amp did lack depth and punch - in other words still sounded "thinner".  I then got my JTV59 and used the same setup, and everything was improved with the better sounds produced from the JTV59.  Then I got the HD500, and it took me many months to migrate from the X3L to the HD500. Many many hours of tweaking etc., and I am still doing it now as I have just bought 2x BEHRINGER B210D active powered speakers, so needed to get the sound adjusted for them. The HD500 seemed to have an even weaker output than the X3L when using the same PA setup, so I carried forward the idea of using the Y cable to capture both L+R outputs to avoid a thin "live" sound and also found that the HD500 output still needed a bigger boost before heading to the powered PA speakers to generate the correct volumes. So, I started using my XENYX 802 for this purpose. The XLR outputs from HD500 went into a Y cable and then a single XLR went to the XENYX 802 into a mic channel with trim set to about 9 o'clock - and then a little adjustment on the HIGH and MID EQ controls, set the CHANNEL volume to 0db (12 o'clock) and the MASTER mixer output to 0db (12 o'clock) - take both L+R master outs (jack connectors) combine with yet another Y lead (oh no!) and then into a jack to XLR converter and then XLR into the B210D and then link that to the next B210D with the THRU connector and another XLR.  Both B210D's had their trim control's set to 9'oclock.  That seems to produce a nice loud volume that is OK for home usage but will not be loud enough for gigs. However, I still have the ability to turn up the MIXER channel volume, the MIXER MASTER out and the B210D trim controls which should get me there, BUT, I have not had the chance to try that yet.

What now?

After reading all your information I now realise that ALL of what I have done to overcome the "live" thin sound of the POD at GIG volumes is WRONG!  I will have to start again from scratch with the correct knowledge of how the HD500 signal routing works and work at getting a good "full" tone from the POD at gig volumes using other means! 

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Thanks again Lorenzo.  I do have lots of options and I will try some different configurations.  The simplest one is what I have now got working. It allows me to just take one speaker to rehearsal and then move up to two for gigs when more volume is needed.  Having two speakers also allows me to go "stereo" if I want to.  I chose the 10" speakers because of price (2 for less than half the price of l2m), flexibility (stereo or mono) and size (portability) and because I thought 200W+200W would be enough power and headroom.  However, I am now wondering if I should have chosen the B212D instead - twice the power (400W) and the 12" speaker for about 30% more price.  I may yet change up.    I was also considering upgrading to the Stagescape L2m as it has had such great reports and integrates with L6LINK and has 800W of power - even though it would remove any "stereo" option unless I bought two of them, but that is highly unlikely at their price.  I believe the l2m only has a 10" speaker.

 

Thanks once again for all your help and advice!   :)

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yeah the lm2 has a 10" speaker

but don't compare a 10" DSP powered speaker to a 12" one that costs 3 times less...

... well you can compare it, using your ears if you can try out one!

I've never tried one

don't buy before try!

 

I personally use an unusual setup that has worked great for me with normal-tuned, baritone and synth guitar:

pod L-jack out > ART DPSII valve preamp/DI > markbass F1 return (used as a Class D 500W linear power amp with comprehensive eq) > one or two Gallien Krueger neo112 (passive 12"x1 bass cabinets with the tweeter switched off)

maybe it wouldn't work for an acoustic guitar

but for my use sounds great from super clean to super distorted and it's light, modular and loud

 

I think the main positive role has to be given to how that speaker, designed for an instrument, works on transients and dynamics (one of the aspects that the pod DO NOT model) in comparison to a PA speaker

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So, assuming I set inputs this way: 1.Guitar - 2-Guitar/Same: do I have to worry about what meambobbo says in his guide about "some comb filtering" that makes "the tone sound looser and slightly out-of-phase"?  Can anyone quantify this loss of tone quality?

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  • 3 weeks later...

Hi Milan

I'm VERY sorry for my delay

I've been too busy in the last month...

You did a great job, man!

I like your drawings

the only details that I would change are in the pan section:

1. if you want to insert the volume fader, that should be a separate block from the pan and it should be before it

2. I would call the pan "pan (balance)" just because it IS a balance but it's labelled pan in the HD500

3. (this is difficult to explain in words, but it's important: I'll try my best)
the lines which represent the signals that output from the pan should not start from the same point but should be separated

the concept that must be represented is "two signals go into the balance and two signals come out of the balance"

4. if you want you can add a "+" sign (which is not present in my drawings) where the left signals from A and B are summed and another where the right signals from A and B are summed

again I know I'm late but if you can put theese mods into your drawings they would become perfect

thanks again
and merry christmas!!
Lorenzo

p.s.
please, let me know how to quote you as the drawer when I'll put your drawings in my original post in this thread
 

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So, assuming I set inputs this way: 1.Guitar - 2-Guitar/Same: do I have to worry about what meambobbo says in his guide about "some comb filtering" that makes "the tone sound looser and slightly out-of-phase"?  Can anyone quantify this loss of tone quality?

 

my short answer to your question is "if you don't hear the problem yourself, don't worry"

 

meambobbo is definitely "devoted" to precision and other users in this forum reported the same "feeling" that something is wrong when using both inputs, so I do not doubt they hear something,

but I never read more than personal opinions on this matter and no scientific test

(simply switching between one or two inputs in the same patch is not even to take into consideration due to gain differences)

 

If the phase issue between input-1 and 2 was true, then it should be ascribed to a serious bug in the POD HD signal routing

 

I did a quick test (I think more than a year ago) and I posted the results in the old forum which unfortunately is not searchable now...

 

I remember I tried the two inputs whith sine waves and there was no sign of phase issues...

 

anyway you can try it empirically:

 

- recall a default "new tone"

- set-up a fixed volume pedal (de-assign it from any expression pedal) in the pre path

- set the volume to 71% (this is approximately -6dB)

- select an amp

- save the patch and then save a copy of it in a new memory location

- in the new patch bypass the volume pedal and set Input-1 to Guitar and Input-2 to Variax

- save (if you're using the pc editor, also save on the pod itself and disconnect the usb cable)

- now switch between the two patches with the footswitches and listen for differences

 

I did it:

- I did put a looper before the pod (the internal looper IS affected by the input selections)

and tried the above with different riffs and some different amplifiers (twin vib, plexi brt, mesa) and I don't hear ANY difference (with headphones)

- I even tried it with parallel paths putting a mono-summing fx block (the noise gate with 100% decay and 0% threshold) after the volume pedal in the single input patch and two identical amps on the two paths

(by the way, the dual path with two with identical amps and settings, sounds exactly as the single path with only one amp and with identical amp and settings; I used the default amps settings to be sure)

- I then tried a "real" dual path with different amps: same result

 

...I'd like to have the time to run more tests and record the results to permanently confirm or deny the issue...

but sorry, for now I'm done... for me it's not an issue

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Yeah, I don't see how there could be a phase issue, but there could definitely be an internal clipping issue.  I wouldn't be surprised if max input b4 A/D clipping is 6dB lower than the internal clipping of  FX,  I wouldn't use "same" input settings at all.  I wonder if you can drive the amp model inputs more than the FX without clipping?  If this hasn't been covered already, I'd also keep the amp block Channel Outputs down, and avoid boosting the Mixer block levels or centering the pans if running a mono summing FX post amps to ensure against post FX or D/A clipping.  Line 6 should really clear all that up in the documentation.

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- internal clipping is surely possible but not at default settings and only using one amp (being it at default settings too),


 


in the test above, if the guitar doesn't clip the A/D converter (you can hear it with all amp/fx bypassed), then using Guitar/Same will not clip;


 


If engineers at line6 designed a dual path unit which, at its default settings, is unable to receive the same signal on both inputs, without internal clipping, they would be crazy or unqualified (if you start adding eq, comp, etc of course you can clip it)


 


- my understanding of the POD HD500 routing, as explained in my schematics, simply says that the "normal" level to send to the first effect is with both inputs active (Guitar/Same),


this is proved by the fact that putting the same distortion/amp in the pre-path or in one of the dual paths gives the same distortion ONLY if the inputs are on Guitar/Same (see the last part of my original post, which has been edited to be even more clear);


 


as I already wrote, this doesn't mean that using only one input (Guitar/Variax) is wrong,


on the contrary, it is perfectly reasonable if you want cleaner tones:


it gives exactly the same result as adding a fixed volume pedal at -6dB, but it's better, because you don't waste an fx block!


 


in the same way you just can't say that using both inputs is wrong or that it's causing problems on its own (unless you prove it);


 


on the contrary, if you want to use a dual amp setup and you choose only one input (Guitar/Variax) then you have to waste an fx block (noise gate 100%-0%) to send the signal to both paths!


This doesn't make sense to me at all


(unless you need to attenuate the input because the amplifiers' drive pots are already low but you want an even cleaner tone from both amps).


 


- we arrived at the point that many people think to solve the so called "problem" of the two inputs by giving away their ability to use two signal paths and putting every effect (and amp) on path A only!


 


This choice is unfounded for 2 reasons:


1- you have a machine able to do complex routing and you give this possibility away because you think it is flawed (if it's true, return it)


2- worse: by putting everything on Path A, you are actually sending to your effects exactly the same signal * as if you did put everything in the pre path and select Guitar/Same so it just CAN NOT sound different, but, as someone else said that not using input-2 and path-B sounds better, you just do it without properly testing it and against the default settings of the unit


 


I'm aware that all of this confusion comes from the official documentation: I think that the manuals are guiltily incomplete on purpose, to avoid to make you think that it's a complex machine because it has a complex manual.


 


that's why I posted all of this: to explain how this unit really works to people who are interested in using it at it's full potential


 


* look at the last 2 pictures in my original post


if you use those two setups with Guitar/Same you get EXACTLY the same result


if you use those two setups with Guitar/Variax you get less distortion in the second one


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- we arrived at the point that many people think to solve the so called "problem" of the two inputs by giving away their ability to use two signal paths and putting every effect (and amp) on path A only!

 

This choice is unfounded for 2 reasons:

1- you have a machine able to do complex routing and you give this possibility away because you think it is flawed (if it's true, return it)

2- worse: by putting everything on Path A, you are actually sending to your effects exactly the same signal * as if you did put everything in the pre path and select Guitar/Same so it just CAN NOT sound different, but, as someone else said that not using input-2 and path-B sounds better, you just do it without properly testing it and against the default settings of the unit

 

Actually I have another reason for putting everything in path A and using guitar/variax input settings.

The reason is that this way I only need to change the volume setting on one channel in the mixer when I need to tweak the volume. I don't like using the amp volume because sometimes when I want to lower the volume on a patch the knob on the POD is near max. If you then turn the knob the volume will first jump to max. This can result in som scary lollipop (pardon my French).

Using the mixer volume, by turning the little knobs works great because these just adjust the volume frome wherever it is set.

I wish all the knobs on the POD worked like that.

Anyway, using both channels would mean adjusting both channels in the mixer so when I don't need the two paths I put everything in Path A for this reason.

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One way to get unity gain that seems to work for me is to use an SPL meter to check the level of a straight tone with no amp block, then build the tone from there checking the level with each added effect. I'll usually turn off each effect after adding it so I'm only getting the gain of the new effect being added and can adjust to parity with the original dry signal measurement.

 

I'm not sure it matters whether you turn off each effect before you add the next one, but it works for me.

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Hmm...that's a good idea.  Glad I thought of it! :lol: Thanks boyce.  There are SPL meter apps for cell phones.  Unity gain can be checked through an audio system, but internal clipping can not.  Monitoring should be reasonably loud for better meter accuracy.  I don't think you need to turn off each previous effect when checking each successive one.  Once you've checked the previous effect, all that matters is whats after it.

 

I finally read through your original post perapera.  It all makes sense now.  Mono FX sum the 2 same signals to +6dB, and drop the output back -6dB, so they won't initially sum +6dB if one output is set differently.  To achieve unity gain with unique  inputs on either signal path, I guess the first mono FX block output on either pre path should actually be boosted +6dB.

 

I just checked the peaks on some of my guitar recordings.  It's not uncommon for the top ~6dB of the transients to fall within a few ms.  If that were clipped, it may be too brief to be perceived at all if masked under various gain inducing FX.  However, a much longer portion of the signal following the few ms clip is altered as an A/D or D/A circuit recovers.  For that matter, internal digital signal clipping of a few ms before an amp may not be an issue, because there is no real circuitry involved to recover from it.  Still, a few ms clip after the amps could definately cause ear fatigue and irritation in the listener.

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Hello everybody,

 

I'm getting a little bit confused. Can somebody help me with my setup?

 

I usa a JTV59p guitar using the VDI cable.

- What input settings do you suggest?

 

Next, i'm only using one amp. I need to send the same sound to both my stagewedge (XLR or Jack) and to the FOH mixer.

What I do, is centering the Mixer and send XLR left to my wedge en XLR right to the FOH

 

Is this the way to go? Or can you give me some advice?

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Hello everybody,

 

I'm getting a little bit confused. Can somebody help me with my setup?

 

I usa a JTV59p guitar using the VDI cable.

- What input settings do you suggest?

 

Next, i'm only using one amp. I need to send the same sound to both my stagewedge (XLR or Jack) and to the FOH mixer.

What I do, is centering the Mixer and send XLR left to my wedge en XLR right to the FOH

 

Is this the way to go? Or can you give me some advice?

 

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.

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Thanks!

Another question. Do i have to put the effects in path A only between the amp and de mixer or after the mixer?

Path A & B are "combined " at the output of the mixer into a summed stereo feed (L/R from path A + L/R from path B ). So if you want the effects in Path A with a dual amp setup, you'll need to place them before the mixer in Path A. If only using one amp block, you can place them post mixer. In this case you might want to center the pan control on the mixer so you are getting the output from both the L/R channel of Path A and mute Path B.
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Thanks!

 

Another question. Do i have to put the effects in path A only between the amp and de mixer or after the mixer?

 

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.

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We might be hijacking the thread, but there are good suggestions here.  If you run stereo FX after the amp, you would need 2 XLR's to FOH to get the stereo signal. You only get the summed stereo signal when you use 1 1/4" main out jack.

 

I did live sound for a while back in the 80's.  We never ran stereo mixes back then, because you can't get a good mix for various audience positions.  It's hard enough get a decent eq for various positions without having to consider stereo as well.  FI, if you run a stereo delay, the dry signal won't be as loud for people on one side, so the guitar part could sound off time.  It seems like M/S mixes could be made use of live, but I wouldn't worry about all that.  If you run 2 XLR's to FOH, the HE can sum them to mono or whatever is normally done at the venue.

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Actually I have another reason for putting everything in path A and using guitar/variax input settings.

The reason is that this way I only need to change the volume setting on one channel in the mixer when I need to tweak the volume. I don't like using the amp volume because sometimes when I want to lower the volume on a patch the knob on the POD is near max. If you then turn the knob the volume will first jump to max. This can result in som scary lollipop (pardon my French).

Using the mixer volume, by turning the little knobs works great because these just adjust the volume frome wherever it is set.

I wish all the knobs on the POD worked like that.

Anyway, using both channels would mean adjusting both channels in the mixer so when I don't need the two paths I put everything in Path A for this reason.

 

Hi talwilkins

 

yes of course yours can seem a good reason for doing this

and you're not saying that you do it because it sounds better... 'cause it doesn't

 

indeed it's exactly the same, I mean exactly...

 

...follow me:

I imagine that you mute the path B and center the path A pan, right?

well what I'm telling you is that if you move all yor effects in pre and post path and put the amp in its default position keeping the same fx/amp order

(and leave the B fader muted and the path A pan centered)

the sound will remain TOTALLY unchanged as long as the input settings are at their default (or guitar/same)

and you'll still have you single master fader

 

see? no need to go so far from the default, that's all I'm saying here

 

just try it on a copy of your patch, save it and compare them

 

 

peace

Lore

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