Latency in music production is basically the delay between when someone plays or sings something and when they hear it back through their headphones or speakers. It’s measured in milliseconds and happens because digital audio systems need time to convert, process, and output sound. While some latency is just part of digital recording, understanding and managing it properly makes the difference between a smooth recording session and a frustrating one where timing feels off.
Latency is the time delay between an audio signal entering the system and coming back out. In digital recording, this delay happens because computers need to convert analog signals (like voice or guitar) into digital data, process that data, then convert it back to analog for speakers. Every step takes a tiny amount of time, and those milliseconds add up.
The main culprit is the audio interface’s buffer size. Think of the buffer as a waiting room where audio data sits before processing. A larger buffer gives computers more time to process audio smoothly, but it also means longer delays. Sample rate also affects latency – higher sample rates mean more data to process, which can increase delay times.
Modern computers are fast, but they’re still doing millions of calculations per second when recording audio. Add in effects processing, multiple tracks, and real-time monitoring, and it starts to make sense why latency is a fundamental part of digital audio production that needs to be managed rather than eliminated completely.
Most people start noticing audio delay when it exceeds 10-15 milliseconds. At this point, playing feels slightly behind the beat, which can throw off timing and make recording uncomfortable. Professional studios typically aim for latency under 10ms during tracking, though some musicians are more sensitive to delays than others.
Different instruments have different tolerance levels for latency. Drummers and percussionists are usually the most sensitive – even 5-7ms can feel sluggish when trying to lay down a tight groove. Vocalists also struggle with higher latency because it affects how they hear their own voice, potentially causing pitch and timing issues. Guitarists and keyboard players can often tolerate slightly higher latency, especially when playing sustained notes or chords.
The recording situation also matters. If someone’s overdubbing a solo part while listening to a backing track, even 20ms might be workable. But if multiple musicians are trying to record together in real-time, latency as low as possible is needed to maintain that natural feel and tight timing between players.
Input latency is the delay experienced while recording – the time between playing a note and hearing it in headphones. Plugin latency happens during mixing when effects processors add their own delays to the signal. While input latency affects performance in real-time, plugin latency is something DAWs can usually compensate for automatically.
Some plugins add minimal latency – simple EQs or compressors might only add a sample or two of delay. But complex processors like convolution reverbs, linear-phase EQs, or lookahead limiters can add significant latency, sometimes 100ms or more. When multiple plugins get stacked on a track, these delays accumulate, creating what’s called cumulative latency.
The good news is that modern DAWs handle plugin latency compensation automatically. They calculate the total delay for each track and adjust timing so everything stays in sync during playback. However, this compensation doesn’t help during recording – if someone’s monitoring through plugins with high latency, they’ll still experience that delay in their headphones. That’s why many engineers use separate monitoring chains or “low-latency monitoring” modes when tracking.
The quickest way to reduce latency is adjusting buffer size. During recording, setting the buffer to 64 or 128 samples gives minimal delay. There might be some clicks or pops if the computer struggles, but modern machines usually handle these settings fine. When done tracking and moving to mixing, increasing the buffer to 512 or 1024 samples works well – the latency won’t be noticeable during playback, and the computer will run more efficiently.
Direct monitoring is another powerful tool. Many audio interfaces let people hear their input signal before it goes through the computer, giving zero-latency monitoring. The trade-off is not hearing any plugins or effects while recording, but those can be added later. Some interfaces offer DSP-powered effects for monitoring, giving reverb or compression without adding latency.
For music production courses, students often learn to freeze or bounce tracks they’re not actively working on. This renders the audio with all effects applied, reducing the processing load on the system. More efficient plugins can also be used during tracking – saving those CPU-hungry vintage emulations for mixing when latency isn’t as important. Finding the right balance means understanding when low latency matters most and adjusting the workflow accordingly.
As projects grow, computers have more work to do. Every track needs processing power, every plugin needs calculations, and all that audio data needs to flow through the system in real-time. When there are 50 tracks with multiple plugins each, the CPU is juggling thousands of processes simultaneously, and latency increases as the system struggles to keep up.
Complex routing makes things worse. Send effects, bus processing, and parallel chains all add to the processing load. If there are multiple reverb sends, several group buses, and a master chain running, each routing point adds a tiny bit of delay. These delays might be imperceptible individually, but together they create noticeable latency that affects workflow.
The solution is strategic project management. Freezing tracks that are finished editing, especially those with heavy plugin chains, helps. Committing to effects when possible – printing that reverb rather than keeping it live – also works well. Using track templates and presets that work efficiently with the system makes a difference. Many professionals working in music production courses learn to work in stages: tracking with minimal processing, then gradually building up the mix as they freeze and consolidate tracks. This keeps projects responsive even as they grow complex.
Understanding latency helps with working more efficiently and avoiding the frustration of timing issues during recording. While latency can’t be eliminated entirely in digital systems, knowing how to manage it means focusing on making music instead of fighting technical problems. The key is finding the right balance for each stage of production and using the tools available to keep latency under control.
At Wisseloord, artists and producers get help mastering these technical aspects while developing their creative skills. The programs cover everything from basic recording techniques to advanced production workflows, ensuring people have the knowledge to handle any recording situation confidently.
For those ready to learn more, contact the experts today.