The curse of latency

Definition of latency:
Latency is the time delay in a signal.

When using digital sound recording and playback systems, latencies inevitably occur, because the analog signal of e.g. the voice or the instrument is first converted into a digital signal. Then it is usually processed digitally for example with digital compressors, chorus, reverb and finally converted back to an analog signal so that it can be heard through speakers or headphones. One speaks here of analog-digital and digital-analog converters, or in short: A/D or D/A converters. The time for the A/D and D/A conversion is typically:

Delay Equivalent distance *)
A/D converter 0.5 ms 0.17 m
D/A converter 0.5 ms 0.17 m
Buffering for digital signal processing **) typically
2 - 30 ms
0.68 m - 10.20 m

*) As we know, the speed at which sound propagates is quite small compared to light, namely about 340 meters per second (340 m/s). This means that the sound needs, for example, about half a millisecond (0.5 ms) to cover a distance of 0.17 meters (0.17 m). In this way one can determine a corresponding equivalent distance for each signal delay.

**) For example, with a sampling rate of 44100 values ​​per second (44100 Hertz) and a buffer size of 1024 values, it takes at least 1024 s / 44100 i.e. 23.2 ms to fill the buffer.

So what?

... you could say. But: The greater the delay, the more difficult it is to play tight to the point, i.e. to the metronome or drums beat. You can only play well together on larger stages if you don't stand further than three meters apart, or if you have a good monitor system.

My recording experiences are:

Delay Equivalent Distance Feeling to hit the metronome
0 ms 0 m Excellent
4.5 ms 1.5 m Very good
6 ms 2.0 m Good
10 ms 3.3 m Still ok
15 ms 5 m Difficult
> 29.4 ms > 10m Awful

When playing acoustic piano or acoustic guitar, the distance between the sound source and the player's ears is usually 1..2 m. With electric guitars, a distance of 3 m from the speaker to the player is quite common. However, if you try to play when you are more than 10m away, for example with a wireless device, latency makes it almost impossible to hit the beat and to keep up with the band.

Singers hear their own voice right in their head due to the sound transmission through the bones. The slightest delay results in a comb filter effect, which means that some frequencies are extinguished and the voice sounds kind of strange. Despite getting used to typical monitoring situations on stage, it is of course a pleasure for singers to have an analog mixer and an in-ear monitoring system or headphones.

Apropos Combe filter:
The formula for the resulting amplitude of 2 overlapping identical signals, one of which is delayed, is:

$$A(f)=2\cdot|\ cos(2\cdot\pi\cdot f\cdot\frac{dt}{2})\ |$$ For example, dt = 2 ms leads to extinction at (A=0) at 250 Hz, 750 Hz, 1250 Hz, 1750 Hz, 2250 Hz,... (250 Hz * 1, 3, 5, ...),

How do you avoid excessive latencies when recording?

There are several ways to avoid excessive delays:

Measure the delay!

This is an easy way to measure the delay:

1st microphone

Head phones and 2nd microphone

Record the metronome with the 1st microphone.

Record the monitor signal from the head phones with a 2nd microphone.

The comparison of the two recordings shows the time delay (latency) of 50 ms here.

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