The Truth about Latencies
Latency is unavoidable when working with digital signal processing:
|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
3.5 - 30 ms
1.16 m - 9.90 m
*) For example as the velocity of sound is approx. 330 m/s, the sound needs e.g. 1 ms to go 0.33 m.
**) for example at a sampling rate of 44.1 kHz and a buffer size of 1024 samples you need at least 1024/44.1 * ms = 23.2 ms to fill the buffer.
... you might think. But: 10 ms is the maximal delay you (I) can handle to exactly hit the metronome beat.
And that also meets my experience on large stages: You have to make sure that no musician is further away from the drums (and bass) than 3 meters to be tight on the groove.
So that´s what I think about acceptable delays *):
|Delay||Equivalent Distance||Feeling to hit the Metronome|
|4.5 ms||1.5 m||Excellent|
|6 ms||2.0 m||Good|
|10 ms||3.3 m||Ok|
|15 ms||5 m||Difficult|
|> 15 ms||> 5m||Awful|
*) I´m, talking about playing guitar so guitar players are used to delays up to 10ms because of the distance from the amp or guitar to the ear. For singers things are quite worse: Because singers hear their own voice directly in their head, the smallest delay result in a comb filter effect which means that some frequencies will be extinguished and their voices sound like being on the loo. So despite of being used to this effect from typical monitoring situations on stage it´s of course a pleasure for singers having an analog mixer and an in-ear-monitoring system or headphones. For those who still think you cannot hear effects resulting from a comb filter effect when the original signal is superimposed by the same but 2ms (that´s what today´s digital mixers are able to provide) delayed signal: The formular for the resulting amplitude of 2 overlapped equal signals where one is delayed (dt=2ms): A(f) = 2 * abs[cos(2*pi*f*dt/2)] And this means extinctions (A=0) at 250Hz, 750Hz, 1250Hz, 1750Hz, 2250Hz,... (250 Hz * 1, 3, 5, ...) So you better go to the acoustician if you don´t hear that. For the older devices (dt=10ms) it´s even worse: Extinctions (A=0) at 50Hz, 150Hz, 250Hz, 350Hz, 450Hz,... (50 Hz * 1, 3, 5, ...) So when shall we trash our good old analog devices? I would say when delay times are lesser than 0.033ms (33µs). This leads to extinctions (A=0) at 15kHz, 45kHz,... that no one will hear.
What to do?
There are several ways to avoid too great delays:
- Hardware monitoring (analog signal will be put through): Most of the sound cards (even the cheaper ones) have some sort of direct analog recording out you can put through to your head phones. Of course your instrument is then "extra dry" (without any effect)
- Better approach: Using an analog mixer between your instrument and the sound card device. Plug your instrument via DI box into the analog mixer and put one output to the digital recording system and one to the monitoring system.
Measure the delay - don´t trust the manual
This is an easy way to measure the delay:
Use a separate metronome and record the metronome beat with the 1st microphone
Send the sound to the head phones and record the output with a 2nd microphone (as near as possible).
|Compare the two records: In this example we got 40ms delay|