![]() 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. 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. If your buffer size is 256 and the sample rate is 96 KHz you will get 256/96,000 = 2.7 ms latency. A common sampling frequency for live use is 44.1 KHz.įor 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. ![]() Sample rate determines how many samples your audio interface will capture every second and do the above-mentioned conversions. It takes any audio input, converts that into digital form (numbers) and then on the output side – converts those numbers back to analog audio. Your audio interface is an analog-to-digital as well as digital-to-analog converter. ![]() The latency here is about 10 ms.īuffer size is basically the number of samples that will be collected before your audio plugins get to process them. Basically, it is a delay.įor 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. ![]() In this section we will introduce some basic terms such as audio latency, buffer size and sample rate and suggest recommendations for live use.Īudio latency is simply the amount of time that passes between the sound being generated and then perceived by your brain. ![]()
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