In this chapter we introduce some basic terms such as audio latency, buffer size and sample rate and suggest recommendations for live use.
Audio latency is simply the amount of time that passes between the sound being generated and then perceived by your brain. Basically, it is a delay.
For example, if you are 10ft away from the speakers, and since the speed of sound is approximately 1,000 ft/s in air it means that it takes 10 ft : 1000 ft/s = 0.1 seconds (or 10 milliseconds) for sound to travel from the speakers to your ears. The latency here is about 10 ms.
Buffer size is basically the number of samples that will be collected before your audio plugins get to process them. Your audio interface is an analog-to-digital as well as digital-to-analog converter. It takes any audio input, converts that into digital form (numbers) and then on the output side – converts those numbers back to analog audio.
Sample rate determines how many samples your audio interface will capture every second and do the above-mentioned conversions. A common sampling frequency for live use is 44.1 KHz.
For example, if your buffer size is 256 and your sampling rate is 44.1 KHz (44,100 times per second, as Hz means cycles per second) then your latency will be 256/44,100 seconds which is 0.0058 seconds or 5.8 ms.
If your buffer size is 256 and the sample rate is 96 KHz you will get 256/96,000 = 2.7 ms latency.
You can experiment with this: If you change the buffer size to 128 and leave the sampling frequency at 44.1 KHz – your latency will be 2.9 ms and so on.
These values directly affect the performance of your PC, as smaller latency values require your computer to respond more quickly to process all those samples in time without producing any glitches.
It has been shown that people can perceive differences between 3 ms – 10 ms, and that our brain cannot distinguish anything below 3 ms.
Therefore, for live performance, many musicians use a buffer size of 256 or 128 and a sample rate of 44.1 KHz. We are not discussing here a recording scenario, which should be done slightly differently.
Generally speaking, using higher buffer sizes or lower sample rates lets your computer run at lower temperatures and allows you to run more intensive plugins in parallel without crackles or pops, but the trade-off will be higher latency or lower quality respectively.
Another factor is the additional latency introduced by your audio interface. Not all audio interfaces are created equal. Some have low internal latency for both A/D and D/A conversions while others add more latency than perhaps you might like, so you have to lower your buffer size to get the overall desired latency.
To find out how much extra latency your interface introduces - you can use Gig Performer's built in Latency Measurement Tool.
Note: Gig Performer 4 also provides you with a feature to see how much latency every new plugin adds to your setup. Simply hover over your plugin block to show a tooltip look for the Latency information. Find out more in the Plugins chapter.
The point is to create sufficient CPU headroom to ensure a glitch-free live performance. You can also set smaller buffer sizes if that does not impact your live performance, but at the cost of less CPU capacity for processing more plugins.
