AUDIO LATENCYAudio latency is the time delay between inputting a sound and hearing play back from your soundcard or audio interface. It also affects the overall responsiveness of your DAW. On a given track within your DAW, the more processing which must occur within your signal chain, the longer it will take for the sound to make its way to your ears. Although we are talking about delays of merely milliseconds, they tend to add up quickly and the result of all these delays can cause significant problems when working with audio. Operating systems on most computers have complex methods of executing commands both within your DAW as well as outside of it. Unless we instruct the operating system to prioritize our DAW over all else, there is really no way to control how much of your system resources are actually going to be at your disposal when you need them most.
BUFFER SIZEProduction within a DAW requires a continuous stream of data to be fed from your hard drive or RAM to your soundcard's digital-to-analog converter. No PC can do everything at once, so an operating system such as Windows or Mac OS works by running various tasks in turns, in order make the most of your CPU's processing power. Playback in a DAW is achieved by breaking up audio into chunks and storing them using small amounts of system RAM, known as buffers. In order to maintain a continuous audio stream, small amounts of system RAM (buffers) are used to temporarily store a chunk of audio at a time. The soundcard continues accessing the data within these buffers while the operating system performs its other tasks. If the amount of available working memory is set too low and the data runs out before the operating system can get back to empty and refill its buffers, you'll get a tiny gap in the audio stream which results in a click or pop. The smaller the buffer, the more often these audible interruptions will occur.
The obvious way to avoid these problems would be to make buffer sizes larger. However, the bigger the buffer the longer it takes for your DAW to respond to changes to parameters and input, due to increased latency. With high latency values, the producer may be playing a MIDI instrument in real time, but an audible delay occurs between the time a note is played and the time it is actually registered by your DAW. That said, latency is always present, and can never be completely avoided. Finding the balance between buffer size and resultant latency, however, is the key here. Buffer sizes should be set as small as possible without introducing audible pops and clicks, in order to achieve the best performance from your DAW.
PLUGINSVST/AU plugins can also add their own processing latency, particularly dynamics processors and convolution-based reverb effects which employ complicated algorithms and require extensive computation. The more effects plugins present on a given track, the greater the latency will be when compared to other tracks in your DAW. In order to compensate for this, DAWs usually employ compensation throughout the entire audio path.
AUDIO INTERFACES AND ASIOTo reduce the number of errors which occur when attempting to record and process audio, the soundcard driver software has to be designed to offload as much of the processing and scheduling from the CPU as possible. High quality audio interfaces are capable of bypassing the normal audio path from a user application in order to connect your DAW directly to the soundcard itself, eliminating any intermediaries. Audio Stream Input/Output, better known as ASIO, is a computer protocol for digital audio developed by Steinberg, which can provide a low-latency and high fidelity transmission of data between a software application and an audio interface. In other words, ASIO allows producers and sound engineers to access external audio hardware directly, which in turn reduces latency.
Audio interfaces with ASIO drivers feature latency values are much lower than integrated audio. It is not unusual for integrated soundcards to cause latency of over 750ms, while ASIO capable soundcards can easily reduce that latency to 50ms or even less. As a general rule, latency under 100ms will cause software instruments to respond to MIDI input much like a hardware synth would. For monitoring purposes, latency needs to be even lower than that though, ideally between 25ms-50ms. For groove-centric genres such as minimal and tech house, we would suggest getting ahold of an audio interface that is capable of latency under 10ms, as even the slightest delays will affect the overall feel of a track.