Impedance help please!

[bassalisk]

Hello,

I'm starting to understand this, but I still sense there's something I'm not taking into consideration.

I'm trying to run the stereo main outs of a Roland VB-99 into 2 Sansamp DIs, split the stereo signal, send one into a Lexicon MX400 and the other directly into my interface. based on the following info, will everything work out?




The VB-99's Mains output at ?10 dBu.





Here's the Sansamp "Ins and Outs" info:

INPUT: 1/4?, 1megOhm, instrument level. Switches battery power on/off.To avoid battery drain, unplug
when unit is not in use.AND... DON?T FORGET TO MUTE!
PARALLEL OUTPUT: 1/4? unbalanced direct output is ?hard-wired? parallel with input jack.
Instrument signal passes through, UNEFFECTED, to the input of your stage amplification system.
BALANCED XLR OUTPUT: Balanced low Z output. Sends effected or uneffected signal to mixing
console/recorder, depending on the orientation of the Footswitch.
XLR Output Level Switch: -20dB pad to match the output to equipment with different input
level requirements. In the line level position, the output is 0dB. In the instrument level position, the
output is -20dB.
1/4? OUTPUT: Unbalanced low Z output. Sends effected or uneffected signal to amp rig or power
amp, as per the orientation of the Footswitch.
1/4? Output Level Switch: +10dB boost to match the output to equipment with different
input level requirements. In the line level position, the output is 0dB. In the instrument level position,
the output is -10dB.







The Lexicon Input info:


Analog Audio Inputs
Connectors four ?" TRS balanced or unbalanced
Impedance 50k Ohms bal., 25k Ohms unbal.
Input Level +4 dBu nominal, +24 dBu maximum
A/D Conversion 24-bit, 48kHz or 44.1kHz, 128 x oversampling






....So based on all this I'm concluding that the Sansamps need to convert the impedance so the -10 signal is at +4 when it hits the ins of the Lexicon. The answer is probably obvious but I'm still a bit confused and overwhelmed.


Any suggestions?


Thanks so much,

[Gregg Juke]

Break it down one-time... 'Splain it real slow.

What are you trying to do? (And why?)

GJ

[farview]

OK, if you are taking the left output and running it to the DI and then into the mic input of the interface, you have that right.


The right output can just plug right into the Lexicon. There is no impedance issue, just a level issue. The input knob on the Lexicon should be able to take care of that.

If you want to make it even easier, you can just plug the left output into the line input of your interface without the DI.


Those Sansamp DI's are great for plugging a bass into because they tend to thicken the sound up a lot. They are not all-purpose DI's because they cahange the sound way too much.

[Gregg Juke]

That's what I was wondering... Why the SansAmp in this case, instead of a straight-up passive DI?

GJ

[farview]

That's what I was wondering... Why the SansAmp in this case, instead of a straight-up passive DI?

GJ

I was half way through my answer when I realised that he doesn't need any DI boxes.

[bassalisk]

I really appreciate the help.


So I guess i don't need to worry about The mains of the VB-99 outputting at -11 and the Lexicon operating optimally at +4? I was just worried that if I fed the Lexicon directly it wouldn't be the right level and the signal would be too hot. I must have read about this about a dozen times, but I guess I still don't understand.
The VB-99 also has sub-outs that are +4 but those outputs are spoken for.



....I was sort of planning on using 2 sansamps for 3 reasons. first, to adjust the stereo signal level and take it from -11 to closer to Line level (which is +4..am I right?). Second, to thicken up the sound which is...um... a little from thin from the Roland. Third, to split the stereo signal.






Here's a somewhat unrelated question. Is there any advantage to a stereo signal if the two sides are virtually identical?


Thanks again,

Aram

[farview]

So I guess i don't need to worry about The mains of the VB-99 outputting at -11 and the Lexicon operating optimally at +4? I was just worried that if I fed the Lexicon directly it wouldn't be the right level and the signal would be too hot. I must have read about this about a dozen times, but I guess I still don't understand.

well, -11db is much lower than +4db. So the signal won't be too hot, it will be a little quiet. If you add some gain at the roland and at the input of the lexicon, you should be fine.

....I was sort of planning on using 2 sansamps for 3 reasons. first, to adjust the stereo signal level and take it from -11 to closer to Line level (which is +4..am I right?). Second, to thicken up the sound which is...um... a little from thin from the Roland. Third, to split the stereo signal.

From what I undersatand, the sansamp takes a line level signal and brings it down to mic level, you are thinking it does the opposite.

It will thicken the sounds by adding some 100hz and a little fuzz.

If you have left and right outputs, the signal is already split into stereo.

Here's a somewhat unrelated question. Is there any advantage to a stereo signal if the two sides are virtually identical?

No. There is really not much advantage to stereo bass in general. I will use up to 4 tracks for bass, but they will all be panned center.

[bassalisk]

"well, -11db is much lower than +4db. So the signal won't be too hot, it will be a little quiet. If you add some gain at the roland and at the input of the lexicon, you should be fine."

Gotchya... thanks.

"From what I undersatand, the sansamp takes a line level signal and brings it down to mic level, you are thinking it does the opposite."

hmm... Instruments don't output at line level do they? Because the Sansamp DIs are certainly designed for instruments to plug right in.

"If you have left and right outputs, the signal is already split into stereo."

Yes, of course it is. I'm trying to spilt the stereo signal into 2 stereo signals. 2 lefts and 2 rights. Perhaps I'm not using the right terminology.

"No. There is really not much advantage to stereo bass in general. I will use up to 4 tracks for bass, but they will all be panned cent

er."

Yea... I'm trying to decide what would be best. My bass project is out of the ordinary so there only bass signals in origin. It's an unaccompanied solo bass album (my second) so unusual methods have come into play including string separation via an RMC Polydrive system installed into a custom Ritter.


Thank you so much for the suggestions!

[farview]

hmm... Instruments don't output at line level do they? Because the Sansamp DIs are certainly designed for instruments to plug right in.

Some do. Some active basses, keyboards, some effects pedals. The DI is really meant to take a high impeadance signal and change it to a low impedance signal. Most of the time that means that the signal level is dropped in the process. Some active DI's don't drop the signal, some do.

Yes, of course it is. I'm trying to spilt the stereo signal into 2 stereo signals. 2 lefts and 2 rights. Perhaps I'm not using the right terminology.

You are using the right terminology, you are just doing something out of the ordinary. I wasn't visualising what you were trying to do properly, my bad.

Yea... I'm trying to decide what would be best. My bass project is out of the ordinary so there only bass signals in origin. It's an unaccompanied solo bass album (my second) so unusual methods have come into play including string separation via an RMC Polydrive system installed into a custom Ritter.

Well, then all bets are off.

[bassalisk]

I sincerely appreciate the help. I'm attempting to do the tracking for this project at home with decent results. I have an Apogee Ensemble and Logic Pro. I also have an aphex 204, an ART VLA and some Sansamp DIs..... I'm running 1 dry mono signal from the 1/4'' out of my bass, 2 dry stereo "separated string" piezo signals from the Roland and 2 wet stereo signals from the Lexicon M400...
I'm in the process of finding some nice channel strip and extra compression to class up the signals. looking at a UA LA-610 for the 1/4'' signal.

Some do. Some active basses, keyboards, some effects pedals. The DI is really meant to take a high impeadance signal and change it to a low impedance signal. Most of the time that means that the signal level is dropped in the process. Some active DI's don't drop the signal, some do.

I believe I'm getting confused by the difference between
impedance and level. Hi-z, -10dBu, +4dBu, line level, balanced, unbalanced. these terms all seem related and, in some cases, somewhat interchangeable but in other cases totally independent.

Is there some easy way to know which pieces of gear should not be plugged into other pieces of gear?

You are using the right terminology, you are just doing something out of the ordinary. I wasn't visualising what you were trying to do properly, my bad.

No worries... I appreciate you taking the time. I'm not sure if a stereo signal is necessary. both sides of each stereo signal are identical so I'm not sure they need to be stereo... Is the advantage the ability to add stereo effects later?

[The Scum]

I believe I'm getting confused by the difference between
impedance and level. Hi-z, -10dBu, +4dBu, line level, balanced, unbalanced. these terms all seem related and, in some cases, somewhat interchangeable but in other cases totally independent.

Those terms tend to show up in arbitrary combinations. With a little more background, they start to make sense.

Hi-Z is shorthand for High Impedance. This is what you find when interfacing passive electric guitars - there's a coil and some magnets that turn mechanical vibration into a little bit of current. But not much current. If you try to load it very much (like plugging it straight into a speaker), you'll "weigh it down" and it'll disappear...or the lows or highs will disappear. So inputs in this realm are very high impedance - 1 megohm is common. You find this on stuff intended for use with guitars - amplifiers, pedals, DIs (Sansamps) etc.

The level in the Hi-Z world is also pretty tiny...consider that most pedals use 9V batteries - so there's 9V peak-to-peak maximum headroom (usually less than that in practise), but it's not usually a bottleneck. They're almost always unbalanced, too.

High imp stuff is also more subject to some misbehavior - a really long cable can act as a shunt capacitance, robbing you of highs. Because the impedance is high, and not much current is flowing, any little noise induced in the wiring may be comparatively significant.

Active pickups can turn this in several directions - depending on the design of the system, it could simply be a buffer amp, that doesn't add level, but increases the current that the instrument can provide. Or it could be a full-on preamp that boosts the output to line level (aren't there some that take 2 or 4 9V batteries for increased headroom?)

Mic level can be similar - a passive mic is somewhat similar to a guitar pickup - magnets and coils. But a phantom-powered mic probably has some sort of internal amplifier, and it's output might be closer to line level to begin with. That's why mic preamps have a really wide range of input gain - to take signals of almost arbitrary level, and condition them into line level.

Line level covers a bunch of related interfaces. They're almost always driven by amplifiers of some sort. The output impedances are usually low (often 100 Ohms, to preven a short of drawing enough current to kill something), and the input impedances are moderately high (in the 10K to 50K range). There are two common nominal levels -10dBV and +4dBu (notice one is 'V' and the other is 'u'...measured by different standards). -10dBV was the "consumer" standard, for stuff like hifi systems, that trickled into the production world in budget gear like Tascam and Fostex. The +4 standard is more common in the pro world...meaning stuff like radio and TV production...the music studio world is filled with gear on both sides of the fence.

If you really want to be pedantic, there are +4-to-10 interface boxes, commonly called "bump boxes." Henry Engineering make one of the common ones.

However, sometimes, just recognizing that you're interfacing gear built to each standard can be enough to solve the problem. Just use your meters and keep an eye open to headroom, and listen to verify that you're not losing highs/lows or adding noise. If the -10 is driving the +4, you can drive it harder, or add a little gain at the input. If the +4 is driving the -10, you need to lose 12 dB, either turning down the output or the input.

Balanced uses some cleverness to add noise immunity. Keep in mind that balancing is entirely independent of impedance and nominal level. A description of the noise canceling mechanism here:

http://en.wikipedia.org/wiki/Differential_signaling

Is there some easy way to know which pieces of gear should not be plugged into other pieces of gear?

Don't plug anything that's not prepared for it into phantom power.
Always load tube amps properly.
Be careful when mixing balanced and unbalanced - there are a few combinations that just plain don't work.
Use common sense regarding headroom.

both sides of each stereo signal are identical so I'm not sure they need to be stereo... Is the advantage the ability to add stereo effects later?

If both of the signals are truly identical, then it's just dual-mono...and eating up an extra track

I've never had any qualms about running a mono signal into both inputs of a stereo device. Some of my favorite reverbs are mono-in/stereo-out.

I'm in the process of finding some nice channel strip and extra compression to class up the signals.

I humbly submit that you need EQ.

I just checked out your tracks - whoa!

All I can really say is trust your ears, have fun exploring, and ignore us sticks-in-the-mud who say "it's gotta be a p-bass straight into a flip-top, recorded in mono, panned dead center." Our old rules no longer apply.

[bassalisk]

Thanks so much for all the great info and for checking out my tracks Scum!!