How to Reduce Latency So It Doesn’t Affect the Timing of Your Playing
Arghh! Latency! Don’t worry, it doesn’t have to be that way!
One of the biggest issues we face when recording and mixing on computers is latency.
It takes time for an analogue signal, your voice for example, to pass through a microphone into your audio interface, get converted into digital binary code so it can be recorded and then be converted back into an analogue signal so you can hear it through your speakers.
The result is a delay between the signal going in and the signal coming out. This delay can affect your performance as it can make it difficult to play or sing in time to tracks that you have on the timeline.
The more powerful your computer, the quicker it can process the conversion of analogue signal to digital and digital back to analogue, thereby minimising the delay to a point where it still exists but is barely noticeable.
There are other settings you can adjust to reduce latency too.
The hardware buffer setting in your DAW determines the size of the cache that the computer uses to temporarily store the audio input and output data. Usually the buffer can be set to store 32, 64, 128, 256, 512 or 1024 samples of audio. Some DAWs can store more.
The bigger the buffer size the longer it takes for the computer to process the audio data. This is because the computer has to process a greater number of temporarily stored samples.
A benefit of larger buffer sizes is that you’ll be able to run bigger sessions with more tracks and plugins, which is great for mix-down time but not so great when recording.
So, let’s say you’ve recorded some virtual instrument parts into your DAW, a programmed drum beat, a bass line and some keyboard parts, you now need to record a live guitar part. You’ve record armed your audio track only to find that there is a delay between you striking the guitar strings and hearing it playing back in time with your virtual instrument parts. All of your virtual instrument parts are playing together in time, but the delay you are hearing when you strike a guitar string is making it impossible to play with feeling.
One solution is lowering the buffer setting in your DAW’s playback engine. This can reduce the latency but depending on how powerful your computer is, it may come at the cost of possible sporadic glitches or stutters during recording and playback.
Another solution is to free up computer power temporarily by deactivating tracks that you can do without whilst playing your guitar part. You could try recording your guitar whilst listening to just the drums, bass and lead vocal and deactivate all keyboard and backing vocal tracks. By freeing up computer processing power you may be able to set the buffer size low enough so that latency is not an issue whilst avoiding stutters and glitches.
You can also freeze tracks and their plugins if your DAW has this feature. Freezing is the process of exporting a track – be it an audio or a midi track – with all of its compression, EQ and effects plugins, then deactivating them and automatically placing the exported audio back on the timeline in place of the original track. By freezing several tracks, you can free up a lot of computer processing power.
A preferred solution is to monitor the live input of the guitar through your audio interface mixer software instead of through the record armed track in your DAW. The mixer for your audio interface routes the live guitar input to the record channel of your DAW and simultaneously directly to the speakers with near zero latency so you can play in time with the previously recorded parts.
Be sure to keep the record armed track in your DAW muted so you don’t hear the delayed signal as well as the monitored signal. One downside of this is that you won’t be able to monitor through any plugins you have inserted on your record channel but the upside of near zero latency may well be worth the trade-off.
The buffer size and sample rate setting both have an impact on latency. Higher sample rates offer less audible delay but at the cost of higher CPU usage.
You can calculate exactly how much latency there is by dividing the buffer size by the sample rate. For example, 256 samples ÷ 44100 (44.1kHz) = 0.0058 – 5.8ms of latency.
At high sample rates the number of inputs and outputs on your audio interface may be reduced. For example, you may have 8 inputs and 8 outputs on your interface. At 44.1kHz all 8 will be available but 96kHz only 4 may be available.
Latency can also be caused by inserting certain plugins, due to the time it takes for them to process the audio material. The more intensive the calculation the plugin has to perform, the longer the delay will be. With multiple plugins inserted on lots of tracks this can become a bit of a mess resulting in tracks playing back out of time with each other. Most recording software compensates for this with ADC – automatic delay compensation – so you don’t have to move any of your audio or midi parts on the time line manually…phew!
To summarise – use a powerful computer, adjust the sample rate, and buffer size depending on whether you are recording or mixing, or monitor through your audio interface mixer to reduce latency so that it doesn’t affect the timing of your playing.