One of the most common issues in audio production is latency. In all my years as a music producer, latency is an issue that I’ve dealt with many times.
It can make your life difficult when processing, performing or recording audio live.
For example, a musician that is doing a live performance will depend on live instrumentation that has to play in time with their vocals.
If the latency is too high, the performance would be hard to carry out because the instrumentation and the vocals of the musician would be out of sync.
In this post ill walk you through the acceptable amount of latency in audio.
With that said, let’s get right into it.
What is considered good audio latency?
What is considered good audio latency is different for vocals and instruments. Most engineers will strive to get latency below 5 milliseconds for vocals and 10 to 20 milliseconds for instrumentation such as keyboards and synths.
Therefore, a generally acceptable audio latency is 10 milliseconds and anything lower.
There’s no real ideal latency, but the average human ear can detect about half a millisecond of latency therefore, keeping latency as low as possible should be the target for the best audio experience.
With that in mind, let’s go on to discuss what exactly audio latency is.
What is Audio Latency?
Latency simply refers to a short period of delay between when an audio signal enters or is sent through a system and when it emerges.
In recording audio, it is the delay between when a voice is captured by the microphone and when it is recorded inside a Digital Audio Workstation. (Analog to Digital Conversion) A/D.
In audio playback, it is the delay between when you press play and when the audio is heard from your speakers. (Digital to Analog Conversion) D/A.
Potential Causes of Audio Latency
Analog to Digital Conversion can be a potential cause of audio latency however its usually never the case because this kind of conversion is known for incredibly little latency.
The minimum latency that can be detected by the human ear is about 11ms. This means that It would essentially take between 10 and 20 stages of A/D and D/A before we noticed any latency in a signal.
When considering latency in any system, Analog-to-digital and Digital-to-analog stages are negligible it is often times more wiser to focus on the audio buffers for your audio interface and sampling frequency.
Buffering can sometimes be a source of latency.
Most Digital Audio Workstations utilize buffers to help handle the incoming and/or outgoing audio data.
The size of those buffers can have an impact on the amount of latency the system has. Lower buffer settings tend to produce lower latency and also require more resources from the host computer.
Digital signal processing
The role of a Digital Signal Processor (DSP) is to take an incoming signal, manipulate it based on mathematics, and then output a result.
A DSP is a essentially a very fast calculator that performs complex equations in very quick time frames.
A common example of DSPs in the audio realm is Universal Audio’s range of Audio interfaces that utilise an in-built DSP to add on some compression and other effects onto your input signal at zero latency without using precious processing power within your computer.
A slow DSP within your system can lead to higher latency. However, most DSPs being manufactured today are pretty highend. So, if you do suspect poor signal processing, it could be time for you to get a better system.
Latency is also greatly affected by the transmission time and propagation delay.
Depending on the distance that a signal is travelling and it’s speed, latency may or may not be an issue because;
In theory, the formula for latency is:
Latency = Transmission time + Propagation delay
The transmission time is simply the amount of time from the beginning until the end of a message transmission. In the case of a digital message, it is the time from the first bit until the last bit of a message has left the transmitting node.
While the propagation delay is measured in minutes per second and is the ratio of distance to propagation speed.
As you may have guessed, Digital to Analog conversion is simply the opposite of Analog to Digital Conversion. It also affects latency.
If you’re using a slow or poor digital to analog converter while running heavy programs and software that demand a lot of juice from your system.
You’ll most likely run into issues of latency.
For example, most inbuilt soundcards that come with most consumer PCs or laptops are not designed to handle heavy conversion work like that of audio production. Therefore, you’re bound to run into issues of latency if you use them to carry out your audio production work.
You need a special well optimized soundcard that can handle Digital-to-analog and Analog-to-digital conversion with ease. These days it means using a specialized audio interface.
You can check out these 5 Audio interfaces.
How to Reduce audio latency
Audio Latency can be an issue that is why it’s good to know how it can be handled.
This part of the post will discuss some things you can do in order to reduce audio latency.
One way to combat latency is by using an Audio Interface that has the “Direct Monitoring” feature.
Some audio interfaces have a direct monitoring switch or a blending knob that allows you to hear what you’re recording in real time with no latency.
Direct monitoring achieves this by summing up the input signal and routing it directly to your main outputs or headphone or speaker outputs.
However, there’s one point to note, direct monitoring does not process the post effects signal, which means that only the dry signal will be monitored.
For example, if you’re recording guitar and have a reverb effect or echo, you won’t be able to hear those effects.
You’ll simply hear the dry signal.
Evaluate system specs/ Optimize your computer
I also recommend taking the time to evaluate your system specifications.
Granted, this necessarily doesn’t help you fix latency per se, however it can certainly help you determine and figure out what your system can essentially handle, specifically the CPU and the RAM.
If your system doesn’t meet at the very least, the minimum required specs for your audio processing, the amount of latency can be greatly increased because this directly ties in with your buffer size and sample rate.
Most consumer computers that are on the market these days aren’t made with heavy audio production in mind.
With that said, many Digital Audio Workstations and plug-ins require certain and sufficient amounts of resources from your computer to run efficiently.
These resources as you may well know include hard drive space, RAM, operating system, and processor.
If these resources are being used elsewhere it can affect performance and result in latency when recording and processing your audio.
Therefore a one time fix is to simply optimize your computer for recording and audio production.
It’s also helpful to utilize your available resources well.
As you may well understand, smaller buffer sizes produce lower latency but higher CPU usage, and larger buffers mean lower CPU usage but a lot more latency.
Your needs will differ at various stages of the audio production process.
When tracking live, low latency is essential to get the most out of your system.
With mixing however, recording latency is irrelevant because all you need is maximum CPU power for audio processing such as the use of digital and analog effects.
Therefore it’s essential that you remember to adjust your audio buffer size to suit the actual task at hand.
Update Audio and Midi Drivers
Also check your system and confirm that all your drivers are up to date.
Out-of-date drivers can also contribute to recording latency, so check your audio or MIDI interface manufacturer’s website to see if they released new audio drivers recently.
Sometimes, driver updates can easily solve your latency problems therefore you should essentially check for drivers first before you take more drastic measures.
Maximize your Sample Rate & Minimize your Buffer
One of the simplest ways to improve latency is to simply adjust the buffer size and sample rate within your Digital Audio Workstation.
These two things can directly affect how the latency is set and how fast samples are being recorded.
To get a bit nerdy, here’s an equation that will give you a general idea of latency in a typical ASIO configuration.
Buffer Size (number of samples) ÷ Sample Rate (kHz) = Expected Latency (ms)
Just divide the buffer size by the number of samples per second (sample rate).
For example, if you have your buffer size set to 256 and your sample rate at 48 kHz, divide those two (256/48000) and you’ll get 5.3 ms.
If you change the settings to 512 and 48 kHz, however, it’ll average out to around 10.6 ms.
A good practice is to make your buffer as small as possible and your sample rate high.
Disable unused audio interface inputs and outputs
You can also disable some of the inputs and outputs that you’re not utilizing on your audio interface as this can greatly reduce latency.
In certain situations you may find yourself having to add tracks late in a project.
In such situations your computer is already under strain from running a heavy project. To avoid latency, you can use your DAW’s “freeze tracks” function to render down energy-draining tracks as Wave audio.
After doing this, you can easily then switch to a lower buffer size for recording.
Afterwards, you can unfreeze the tracks and restore a higher buffer setting.
Therefore, do take time to have a deep look into your DAW and see how you can freeze tracks.
Audio Latency – Vocals
When dealing with vocals you want as little latency as possible, with the preferable option being: using hardware monitoring and the least preferable being wearing only one side of the headphones.
The problem here is one of consciousness and the brain being able to hear our voices even if we aren’t actually vocalizing.
Our minds anticipate our voice coming out, and if it doesn’t line up with what our ears hear perfectly, it’s going to create problems for the vocalist.
Therefore, the preferable latency with vocals is 5, preferably 2 milliseconds or less, if not zero.
Audio Latency- Electric guitars
If you’re an electric guitar player then you’re obviously used to playing several feet away from amplifiers.
This basically introduces a natural amount of delay because the speed of sound is relatively slow.
Therefore it’s not a crime to generally take the average distance in feet that a player typically stands or is seated from his or her amplifier stack and let that be the acceptable latency in milliseconds.
Which means that if you’re used to sitting on your stool, a few feet or maybe five feet from your amp, then 5 ms of delay will sound normal and not distract or disrupt your playing.
However, If you’re miking an acoustic guitar player or any other stringed or brass instrument, you should rely more on monitoring to achieve zero latency.
Audio Latency – Drums
With drums it’s best to keep latency at around 5 ms or under.
Drums and percussion instruments can be easily perceived as sounding off when the latency is too high.
Therefore it’s best to keep latency low to avoid any mistakes in playing.
Audio Latency – Keyboard & Piano
The process of keyboards and pianos producing a note is relatively longer.
With pianos there’s always the delay between hitting a key, the hammer hitting the string, and the string’s vibration traveling to your ear.
Therefore, a 5 to 6 millisecond latency should be fine.
With keyboards, the amount should be the same unless they’re playing pads, in which case you can get as high as 20 milliseconds without them clashing or the crowd noticing the latency.
Practical Music Production – What Audio Latency Is And How To Reduce It
Sweetwater – Better Latency Than Never
Ledger Note – How to Achieve Low Audio Latency While Recording
Sweetwater – Which Buffer Size Setting Should I Use in My DAW?