Hey gang,
there has been a lot of good info in this thread, but there appears to be some misunderstanding. It will take far more than a short post to explain all the issues involved, so I will sit down and try to write a clear, understandable explanation about interfacing audio devices.
In the meantime, to address the specific issues here:
dB is a power ratio! That's it, that's all it is, that's all it can ever be. (We'll ignore, for the moment, the fact that is the logarithm of the ratio of two powers.)
When the world switched from 'matched impedance, maximum power' transfer to voltage transfer engineers figured out a way to use a power ration to describe a level ratio. Because of the relationship between level and power you can use the dB to describe voltage ratios.
When the dB is used to describe a signal level it must always be with respect to a standard reference level.
There are two generally accepted reference voltage levels:
0 dBu (the 'u' means unreferenced, save the math for later) uses 0.7746Vrms (saving where that came from for later too)as the reference. A positive dBu means that the level is greater than 0.7746V. A negative dBu means it is less.
+4 dBu is the nominal operating level for 'professional' audio inputs and outputs.
+8 dBu is the nominal operating level for broadcasters.
0 dBV (the 'V' stands for volts, and like everything else surrounding this standard it makes sense... oh well!) uses 1.0Vrms as the reference level. Positive and negative dBV numbers represent levels above or below 1V.
Because the two standards use different references we can not simply add or subtract them. The difference between +4 dBu and -10 dBV is about 11 dB - note that in this case we are talking about a difference in levels expressed as a power ratio.
As if that wasn't enough, input stages, output stages, and the connection between them can be balanced or single-ended (aka unbalanced). Balance refers to impedance only - it has nothing at all to do with levels.
A balanced output presents an equal impedance from each signal conductor to ground. AND, ground is not used as a reference.
A balanced input looks at the impedance from each signal pin to ground, and uses that to cancel out any signal that is common to both conductors, while retaining the difference between the two pins. Again there is no requirement for a ground reference.
A balanced cable has two signal conductors. It can also have a ground conductor (which has problems) or a shield that connects to ground at one or both ends.
Most balanced cables twist the signal conductors to make them more immune to magnetic interference. The shield has little impact on magnetic fields. The shield, on the other hand, protects from RF fields, something twisting the pairs does little to prevent.
A single-ended source has only one signal conductor, and uses ground as a reference.
A single-ended input responds to the difference between the signal conductor and ground.
A single ended cable is usually made up of a single signal conductor and a shield, but it can be two signal conductors.
You can drive a balanced input from a single-ended source easily. The noise immunity is dependent on the design of the input stage, a simple op-amp input stage will suffer from the impedance imbalance, an instrumentation amplifier, a transformer, or the InGenius chip can tolerate huge imbalances.
A quick word about symmetry. A signal is called symmetrical if the level on each conductor is equal in amplitude, but opposite in polarity (not phase!) This provides a bit more headroom, and some improvement in S/N ratio, but it is not the feature that makes a balanced interface work.
The best way to connect a single-ended source to a balanced input is

- connect the pin on the RCA connector (or tip of the TS connector) to pin 2 of the XLR (or the tip of the TRS connector) through the first signal conductor.

- connect the second signal conductor from the sleeve of the RCA or TS connector to the ring of the TRS connector or pin 3 of the XLR.

- connect the shield to the sleeve of the RCA or TS connector and the sleeve of the TRS or pin 1 of the XLR (or even better, to the chassis at the receive end!)
This will work almost as well as a balanced to balanced connection if the input stage is properly designed. It's really pretty remarkable how well it works.
It is usually safe to send a -10 dBV signal to a +4 dBu input, you will lose a little bit of S/N ratio, but the preamplifier should be able to provide sufficient makeup gain. And you'll have TONS of headroom<G>!
It is a bad idea to send a +4 dBu signal to an input designed for -10 dBV. You'll need an pad or attenuator in front of the input, and you'll lose both headroom and S/N ratio.
Phew - I still haven't addressed impedance!

Impedance means the opposition to current flow. It can be resistive, which means it applies to energy at any frequency equally. It can be reactive, which means that it is dependent on frequency. It can also be the result of the relationship between wavelength and conductor length.

What matters for now is that in order for an interface to be good at transferring voltage the input impedance needs to be much higher than the source impedance. The rule of thumb range most often quoted is 10:1. And that certainly works<G>!

Typical source impedances for audio devices are usually very low, a couple of ohms is not uncommon. In order to prevent damage to the output stage most manufacturers add a build-out resistor, anywhere from 10 ohms to as high as 50 ohms.

Typical input impedances range from 100K ohms on up. The actual input impedance for a typical op-amp is in the Meg ohms, but additional circuitry is added to protect the inputs, reduce RFI, etc, and so it is made lower.
That's a lot of detail, I know, but it's the shortest thing I could write without resorting to pictures and equations.
I hope it helps, but if you have questions just ask!

Not sure if this is relevant and I did not read all of the above (and my apologies if I am stating the obvious) ... just kinda' skimmed it but if you have not done so, you may want to check your mixer levels in the M-Audio control panel. 

If I recall, when I initially installed my Delta, the M-Audio software installed with all channels set at a default of -6 dB ... and if you are going to use the M-Audio software mixer ... it doesn't really matter.

But, if you are using Sonar, the M-Audio mixer has to be set-up correctly and all of the levels have to be set to 0 dB.

quantumeffect




If I recall, when I initially installed my Delta, the M-Audio software installed with all channels set at a default of -6 dB ... and if you are going to use the M-Audio software mixer ... it doesn't really matter.

Dave - M-Audio's faders are all set at 0db if that is what you mean. ?  I do mix through Sonar and use the control panel mainly for switching sample sizes and muting signals. On that subject, the way the control is presented could be a lot better! Thanks!

@Bill

Thank you for taking the time to present this information is a understandable way. Reading through this I realized how easy it is to be a little confused with how the correlations work. I need to read this about three more times.  :)


Thank You! 

Musicroom

I went thru something like this a while back. Although you are enjoying a lot more technical help here I did have "several" people lend me a lot of good advice.

Here is the thread...(is is quite long!)
http://forum.cakewalk.com/tm.aspx?high=&m=2477795&mpage=2#2518211

I was having an issue with my monitor management device the "Atty 2 d" holding back my sound delivered to my monitors......

so I went and dug out everyone of my manuals for "Everything" and followed the Train...
was it +4 or was it -10
 
Also all my cables are "Balanced" cables and this is a factor as well.

Everything was built with the +4 reference
Except my Atty which I bought a Mackie Big Knob to replace it

so I went thru my signal chain and set everything in the +4 configuration, inserted the Mackie and what a difference....huge difference.....very happy right now......

I then grabbed my SPL meter and set my levels, got out ARC and took measurements and it was set....

But my advice would be to do the same....get your manuals and go thru your signal chain and set your levels....

Best I can do cause all the other technical data already provided will take me days to read and re read and figure out....

Clifford

You're spot on Clifford - the people here have been extremely helpful as I sort thru the gain staging. I read your thread and it looks like a lot of people jumped in there as well. The Atty looks impressive and it's pricey, but ended up being the problem I see. 

I think here it boils down to finding a way to get the +4 signals from the preamp to the delta to sound clear as they sound when I use the (over sensitive) -10 settings. I've recorded many tests tracks over the weekend and it's noticeable. 

I tried a different input slot on the delta yesterday and also made a change in the control to "not" invert the inputs. Progress - I could now use the +4 now with minimum distortion, but there is still some there and some of the sparkle on the high end is missing - not much though. If you did happen to read thru some of this thread, I have been bypassing the transformers on the preamp and using a unbalanced output to the delta. New cables are in transit and I will try all of this using the preamp's balanced outs to the balanced inputs of the delta. I hoping...

If that doesn't work, I would lean towards equipment failure and could be the delta that needs inspection. Some web digging reads that the delta's capacitors sometimes need replaced. If that is the case, I may just replace. Depends on the cost. I've had both pieces of gears for over 10 years, so could be either. I do use updated custom cabling so I don't think that's the problem.

Thanks!

Dave I don't think you are going to have issues making the balanced connection. You might just get enough extra gain that way so you can back off the input gain again in order to get a good level going into your DAW. (and hopefully get a cleaner sound) The distortion you are hearing might just be your preamp going into distortion mode that is all. I would hope not as you should have the option of a totally clean signal at +4dB if you want it. When you hear this distortion does backing off the input gain control clean it up? And if it does is this signal just low at the moment going into your DAW. What reference level and how high a level are also you trying to achieve in your DAW.  Remember rms levels that average -20dB are very acceptable. You need to test and measure rms levels when calibrating any system. (K system sorts all that out!)

This is where I believe recording everything you do through something like this is not necessarily a good idea. And where clean transparent transformerless solid state preamps come into their own. Get a fantastic clean version of what it is you are recording no matter what and use plug-ins later to add the warmth and sound of analog later. The more I work with great sounding analog plug-ins the more I believe they can do it. I love them! More control over the sound now than ever before. 

Bill brings up the question of how to connect an unbalanced signal output to a balanced input. There is another version (not sure if Bill meant this version) where a 2 wire connection with just a hot wire wire and shielded cable can be used to connect to a balanced input. This is where the tip or hot signal of the unbalanced output feeds either the tip of the TRS or pin 2 of the XLR in. The shield of the unbalanced output simply connects to pin 3 only of the XLR or the ring of the TRS. The sleeve of the TRS or pin 1 of the XLR is simply left un connected. This still works because the shield of the cable is still grounded from the unbalanced end and protects the inner hot wire from hum and other noise. It also avoids any possible ground loop hums etc. or noise. Dave you won't get this cable as a commercial product, you have to wire this one up for yourself. Only do this if you feel there is some noise or possible earth hum between the two. 

Your Peavey may just not be well either. How old is it? How much have you used it? Replacing all the valves might just be the ticket and after a decent service it springs back into a louder sounding device with a much cleaner sound.

Jeff Evans

Bill brings up the question of how to connect an unbalanced signal output to a balanced input. There is another version (not sure if Bill meant this version) where a 2 wire connection with just a hot wire wire and shielded cable can be used to connect to a balanced input. This is where the tip or hot signal of the unbalanced output feeds either the tip of the TRS or pin 2 of the XLR in. The shield of the unbalanced output simply connects to pin 3 only of the XLR or the ring of the TRS. The sleeve of the TRS or pin 1 of the XLR is simply left un connected. This still works because the shield of the cable is still grounded from the unbalanced end and protects the inner hot wire from hum and other noise. It also avoids any possible ground loop hums etc. or noise. Dave you won't get this cable as a commercial product, you have to wire this one up for yourself. Only do this if you feel there is some noise or possible earth hum between the two. 

Not trying to pick nits, but the two wire solution described has some pitfalls...
 
First, a shield does nothing to minimize hum, if by hum you are talking about 60 Hz hum. That's a magnetic field, and it doesn't even see the shield. Shielding works to reject RF interference, and twisting the pair protects against magentic fields.
 
Second, if you connect the two grounds, no matter how you do so, you have the potential for a ground loop. A ground loop results from a potential difference between two different grounds, and when there is a potential difference there is current flow. And that current can get into your audio on an single-ended input. It is a little less likely to be a problem when everything is balanced.
 
The two wire arrangement described may work, but it won't work as well as using a twisted pair as follows (here goes my attempt at text graphics<G>!):
 
Never mind - it was unintellible.
 
So I'll try this - please read Jensen App Notes 3 & 4:
http://www.jensen-transformers.com/an/an003.pdf
http://www.jensen-transformers.com/an/an004.pdf
 
They really are not too heavy on the theory, and Bill Whitlock is considered by most in the pro audio community to be the go-to guy on this topic. Sure, he'd love to sell you a handful of transformers, but his app notes are some of the best, and he shares them generously! (and his graphics are WAY better than mine!!!)

I think it is important to try various ways when connecting unbalanced outputs to balanced inputs. One approach may simply work better than the other. The two wire approach does work and worked every time I have tried it. It is particularly good when connecting laptops and IPhone devices etc which may bring noise (all sorts) into the system under a normal connection but can be totally quiet with the connection I have recommended.

There are good reasons for the approach Bill has suggested as well. I would try Bill's approach to this before mine. Those pdfs are an interesting read. I think taking the balanced outs to the balanced ins is the obvious way to go first and see how it goes from there. There is every reason why a clean undistorted signal is possible at +4dBu from that preamp.

I hate to say this out loud but I don't know how to test or calibrate with any sense of confidence. With the problems I'm having it could be that I'm the problem, seriously.

You asked about the +4 signal distortion when operating at a lower gain on the preamp... Yes - I still hear the distortion and when I amplify the signal - it is there with little or no change. The other thing I notice is the waveforms at the +4 signal are less defined and more box like. The -10 waveforms have definition, separation in appearance. I don't know what you would say about that observation. It may not mean anything.

I bought the peavey new around 1995 and it has never been serviced. The delta is old too, I bought it new around 2000. I have nothing to complain about if they both needed work.

The only other preamp I have at my disposal is a small soundcraft notepad that has clean, but kind of thin sounding preamps. I think they are marketed as gb30 preamps. I found them to be just okay. But maybe using the techniques you spoke of with plugins, I may be able thicken and warm the sound up more.

While on that subject, I have been offered a decent price for my vmp2 and I've been thinking of changing to something different - I've researched the preamp jungle and it's confusing. 

I've also thought about changing out the delta 1010 to one of the many choices out there as well (presonus keeps popping up as the first choice). My main reason for stalling is I have a nice sound going from an akg414 to vmp2 to delta.