How fast do sound signals travel through the various parts of the sound system? Do sound signals travel faster in analog snake cables or fiber optic cables?

Also, what about transmission through the air with wireless mics? And where do the true and relevant sources of time lag exist in a audio system?

What I find really interesting is that sound signal travels at differing speeds depending on whether it’s in the form of electricity, or sound waves in air, or via wires, or passing through various pieces of commonly used audio equipment.

Consider that in 1.2 milliseconds of time:

- Sound travels about 16 inches in air before going into a microphone or after coming out of a loudspeaker.

- But if we did rock shows under water, we would find that sound cruises about 70 inches, nearly five times faster.

- It just barely makes it from the input XLR to the output XLR of the faster popular digital consoles.

- Yet it travels about 650,000 feet (124 miles) down a regular copper wire or snake, because in copper, electric signals travel about two-thirds the speed of light.

- It travels about a million feet (186 miles) down fiber optic cable; however, it can take about 630 microseconds (over half of our 1.2 milliseconds) to be converted from analog audio to digital light (and vice versa) at each end.

- But if the analog audio signal is already digitized prior to the fiber optics system, the conversion time can be cut to about 10 microseconds.

- You could probably string together every piece of analog gear you and everyone you know have ever owned plus the entire analog inventory in the Rat Sound warehouse, and experience virtually no time lag. (Analog does not delay much at all.)

- It takes more time for sound to travel one inch from a singer’s lips to the mic than it takes for a wireless transmitter to convert sound to RF, transmit it 75 feet to a wireless receiver, and then send it through an analog console, compressors and EQ, and then convert it back to RF via an in-ear monitoring system and get it all the way to the little speakers in the singer’s ears.

As noted above, sound moves through analog gear so fast that it can be considered instantaneous for most purposes, while digital is a bit more confusing because sound makes several stops for conversion between analog and digital.

Further, once signal is inside a piece of digital gear, it can slide behind in time further as processing is added the signal path, unless the manufacturer has implemented compensating delays that lock the delay time at the maximum processing time.

What does this all mean?

For the most part, the time delays we’re talking about are so short that they can be discarded as irrelevant.

But not so fast! If the signals are being electrically recombined, then it is absolutely critical that they are not shifted in time - especially when dealing with different versions of the same signal.

For example, let’s say you put two mics on a guitar rig, and you’re mixing on one of those sexy new digital consoles.

Being a creative engineer, you pan the mics hard left and right, and then insert a cool tube compressor on one of the mics to get a fatter sound on the left side, while using the more dynamic sound from the right side.

All good. The stereo imaging just got wider, and you’ve made that boring guitar rig sound awesome.

But you notice a strange thing happening - when you cue up one mic or the other, each sounds good, but when you cue up both mics together, it sounds nasally and edgy.

Also, the mono feed to the center cluster sounds strange, as do the mono press feeds that are going on tonight’s news broadcast.

Pondering this annoying gremlin, you realize that by using the analog insert on the digital console, the digital to analog conversion and back has introduced a 1.2 millisecond delay on one of the guitar mics.

Normally, no big deal, as it’s too short to be noticeable timing issue - but when the two time-shifted identical or nearly identical signals are electronically recombined for the mono feeds, they cause audible cancellations.

Another common yet overlooked phenomenon is the effect that the 1.2 millisecond delay (and thus a 16-inch delay) of digital consoles has on the sound of in-ear monitor systems.

When singing, a performer hears the sound of his/her voice from two primary sources:
A) The voice itself and natural resonance of their body.
B) The ear buds jammed into their ears.

With analog, the sound of one’s own voice/body and the ear buds are pretty much perfectly timed. So perfectly, in fact, that you can actually and easily determine whether the mic is in polarity or not by singing and listening while flipping the console phase switch.

With digital, things get more complex. The absolute minimum time delay to the ear buds now becomes 16 inches, so perfect time alignment of the natural voice/body sound with the in-ear sound is no longer possible.

The result is that the singer hears his or her own voice sounding “farther away” than with an analog console.

About now, I expect that some of you are saying, “This is silly. There’s no way that running in-ear mixes through a digital board causes this problem, and even if it does, no one even notices.”

But how do you think I discovered it?

First, by hearing it and then investigating why my voice sounded so far away from myself.

Second, by running one in-ear beltpack from an analog console and another beltpack from a digital console, and then with ear buds in my ears, plugging in back and forth between the two belt packs, using my own voice.

Every engineer I have demonstrated this to has heard it clearly.

Whether these types of time delay issues are a big deal or irrelevant depends on the expectations of the performers, the quality of gear, and several other factors.

Regardless, I’m confident that someday, somewhere, that by being aware of these time issues and having a clear understanding of what’s actually happening, you will save yourself some grief - and maybe even a gig.

Until next time! (Pun intended…)