This is called balanced input/output. TRS connectors are also quite common, especially for interconnects. I find TRS/XLR combi sockets [0] used for this purpose fascinating :)

Third wire does not collect interference in a balanced line. Instead, voltage signalling balanced line uses one ground and two signal lines of opposite polarity and input is difference between these two signal lines. Induced interference is more or less the same on both signal lines, therefore cancels out. At a quick glance this article [1] seems to explain this without delving into electronics. Some differential lines (e.g. RS-485, popular in industrial automation) use current signalling thus requiring two wires per signal.

[0]: http://www.cliffuk.co.uk/products/combijacks/combijacks.jpg [1]: http://www.aviom.com/blog/balanced-vs-unbalanced/

> Some differential lines (e.g. RS-485, popular in industrial automation) use current signalling thus requiring two wires per signal.

Virtually all high-speed digital communication uses differential signalling.

Agree on that, but all high speed interconnects I know use voltage signalling.

That's orthogonal to differential or single ended, but typically the transmit side uses some form of current steering; it just goes well with it.

E.g. PCIe transmitters are HSCL, which is a form of current steering that gives the receiver exactly the voltages it needs to see at nominal currents and termination values.

It gets even stranger: https://en.wikipedia.org/wiki/Star_quad_cable

No, DSSS is a modulation process rather than a signal-shielding process. The difference is that differential signalling can be converted to single-ended with a simple subtraction (or other passive techniques), but recovering DSSS requires digital signal processing.

In fact, he's got a nice little simulation of generating a "gold code", adding noise and multipath distortion to it, feeding it through an I/F mixer, quantising it, and recovering the original code. http://www.aholme.co.uk/GPS/SRC/2011/C++/Simulation.cpp

XLR is bipolar/differential. It sends +sig and -sig on the two conductors.

The receiver inverts -sig and adds it to +sig.

Any noise appears in both lines and is subtracted to zero, leaving 2 * sig.

Exactly.

That's incorrect, the line driver for an XLR output is simply a polarity inverter that exposes both the original and the inverted signal at the same magnitude. The trick is that both lines are equally affected by the interference, you invert the inverted signal again on the receiving end resulting in 2x the signal and 0x the interference.

That's just one of the possibilities. You can either do the cheapo one-line-driven-by-inverter trick, or use a transformer or a differential line driver (like 8903A output stage); basically the opposite of an instrumentation amp. These behave pretty much the same as a transformer, i.e. they have high common mode output impedance. Available in chip form as well, e.g. DRV135.

Problem is the circuit needs very accurate resistors and needs to be carefully balanced, just like an instrumentation amp.

Sorry, what’s different here? The second line is there so that its signal can be subtracted (inverted + added) from the first line, no? I guess I didn’t realize it also had the the inverted audio signal on it, though, thanks.

Is there any reason that prevents audio from being transmitted digitally with zero quality loss? Why can't the digital-to-analog conversion be performed as late as possible? Inside the actual speaker, for example.

You can; it’s just expensive. With Dante-compatible gear all the way through, for example, you could have ADC in your wireless mics and DAC not until your powered speakers. A high-end tour or auditorium built in the last few years might have that stuff. But analog gear still works fine, is more serviceable, is a lot cheaper, and is usually brought over to digital one system at a time (the biggest gains come from digital mixing boards).

there are pro speakers with digital inputs (not just for a clean signal, but usually to do some digital correction for the speaker response) but then you are stuck with that DAC, amp and speaker combination, and you either need a volume control at the speaker or you are losing bit depth by turning down the digital signal.

The opposite also exists - a “digital stage box” to convert analog signals from mics and instruments to digital as soon as possible.