How to Decrease Latency

In this guide, we’ll be focusing on the issue of audio latency and how to decrease latency in your day to day recording and production activities. We assume that you’re at least somewhat familiar with the topic and that you’re already using a digital audio workstation (DAW) for creating and recording your own music. Let’s get to it.

What is Audio Latency?

For an enthusiastic young musician, there’s no better feeling in the world than making your own music at home. All you need is your instrument, a mic, some cables, a computer, and some recording software, and you’re good to go. If you want to be really fancy, you can even use a bunch of additional effects and plugins.

Except things are far from being that simple. You quickly realize that there’s a major issue – whatever you’re recording comes out of your headphones or monitors with a slight delay, making it difficult or impossible for you to record normally. You’ve got yourself a case of audio latency.

When the audio latency is too damn high.

Even if you have the best possible computer and all of the most expensive hardware and equipment, you’ll always have some degree of latency. The point is to reduce latency to an operational minimum, to a point where you won’t be able to notice it.

Depending on what you’re recording, anything below 12 to 15 milliseconds isn’t noticeable to an average musician. Go above this range, however, and it becomes more and more noticeable and more difficult to to deal with. Eventually, it will be almost impossible for you to record and perform in real-time due to the delay. So what are we going to do about it?

Upgrade your Audio Interface

The first thing to think about if you’re doing recording work is a dedicated audio interface. In most cases, audio interfaces are external to your computer and can be connected via USB, Firewire, or Thunderbolt ports. Technically speaking, most computers already have one. They’re commonly referred to as the PC’s sound card. But in order to make it worthwhile, decrease latency, and use all the features your DAW provides, you’ll need to use this specialized hardware.

Analog-to-Digital & Digital-to-Analog

For your computer to understand it, the analog signal from your input (mic, guitar, etc.) needs to be converted into digital information first. Then, it’s processed by your DAW and your computer’s processor. This is where the effects get added in. Finally, it gets turned back into analog before going out to your monitors or headphones. The faster these analog-to-digital and digital-to-analog conversions happen, the lower your overall latency will be.

The converters on a PC’s stock sound card are simply not powerful enough to handle this, and they can be a major source of latency. A dedicated audio interface provides you with more processing firepower for recording. They contain faster digital-to-analog (DAC) and analog-to-digital (ADC) converters that are better suited to quickly convert analog information to digital, and vice versa.

In addition to decreased latency times, an audio interface will also provide you with improved audio recording quality in the form of higher sample rates and bit depths. There are plenty of great budget audio interfaces you can buy today. It’s not that hard to find a decent one at a reasonable price. You’ll be able to record and reproduce your own quality music with considerably less latency.

Change the Buffer Size & Sample Rate

OK, so you have an audio interface. But there are some parameters you should change in order to further decrease latency. Now that we have the issue of sound cards and audio interfaces squared away, the next thing you need to do is to set the buffer size in your DAW.

If you don’t have a DAW, most audio interfaces intended for recording have their own free software you can install after a purchase. This includes all the drivers and other programs that allow you to operate the device’s settings, including buffer size.

What’s a Buffer?

In computing, a buffer is like a bin where data collects while moving between two places in a computer. After adding effects and processing the signal in the DAW, the data collects in the buffer. The sound card/audio interface pulls the data from the buffer and converts it back to an analog signal with the effects added in. The buffer size has a direct correlation to latency, with larger buffer sizes adding more latency. Great, so just reduce the buffer size to zero, right?

Well, no. Smaller is better only to a point, limited by your processor. Although the reduced buffer size can 100% help with latency, it can also have an impact on your recording quality if it’s set too low due to buffer underruns. A buffer underrun is when the interface drains the buffer of data faster than your processor can refill it. So the key here is to set the buffer as small as possible without causing buffer underruns.

Changing the buffer size in the ASIO4ALL driver.

All in all, there is no one standard buffer size for everyone. It’s a balancing act that depends on your computer’s hardware components, like the CPU, and how much data you need to process. Every audio interface driver comes with this adjustable setting, and it’s crucial to get it right.

Sample Rate Settings

You should also think about changing the sample rate. Raising the sample rate can help you decrease latency times if the buffer size remains constant. This is because a faster sample rate means less time between samples. It definitely sounds counter intuitive, but it does help. Here’s why.

Let’s say your buffer is set to 512 samples like above. A sample rate of 44.1 kHz yields a buffer latency of 512 × (1 ÷ 44,100) = 11.6 ms. Now let’s say your sample rate is 96 kHz. If your buffer size remains the same at 512 samples, then your buffer latency is 512 × (1 ÷ 96,000) = 5.3 ms. So why not just just sample at 192 kHz all the time? This should minimize latency, right? Well, there’s a catch.

Keep in mind that higher sample rates also require more processing power. If you have a weak processor, it’s going to have a rough time handling audio at higher sample rates. Which brings us to the next remedy for latency… your processor.

Upgrade your Processor

So in the name of conquering latency, you’ve bought an audio interface and you’ve reduced the buffer size to the smallest possible setting without causing underruns. But even so, you still have a noticeable amount of latency that’s sabotaging your workflow. The last weapon in your arsenal is going to be a processor upgrade.

The processor is the brain of your computer, and it’s responsible for everything that happens inside your DAW. Adding effects like delay, reverb, and chorus are all things your processor takes care of. The processor also enables you to use soft synths like Sytrus, Serum, Massive, and Spire.

Before you jump the gun on a processor upgrade, or even a new PC build, try decreasing your CPU load by closing out some background programs. Common PC processor-hogs are antivirus software, internet browsers (especially Chrome), media players, streaming videos, PC games, and porn viruses your younger brother accidentally downloaded while “doing his homework.”

Close unnecessary processes in Task Manager to reduce CPU load.If that’s not working out, the best option is to upgrade your computer with a more suitable processor. It might not be the cheapest solution, but it will get the job done, and it will do it well. When buying a new processor, you’ll need to look at the processor speed, number of cores, number of threads, and cache memory. AMD’s newer Ryzen CPUs are great for music production, definitely consider them if you’re going this route.

Use Direct Monitoring

The final solution (sort of) to decreasing latency is to use direct monitoring. This is a solution for those who are looking to just play an instrument and use the computer, hardware, and software for monitoring purposes only.

Certain audio interfaces out there support the feature. When it’s selected, the signal from your instrument or a microphone goes through the audio interface and gets routed directly to the output – your headphones or the speakers. This way, the latency is at its minimum. There’s no processing involved and you can use your speakers or headphones as direct monitors.

The downfall of this though is that you can’t hear what’s coming from your DAW You can use monitor mixing to mix in the live signal with what’s coming from your DAW, latency included. But this isn’t ideal for recording situations.

Summary of How to Decrease Latency

We’ve covered a lot of information here. Hopefully you’re more versed in audio latency and how to decrease it, and not scratching your head. Let’s recap everything we learned.

Latency is the time delay in your audio signal. Lower audio latency is always better. Latency is added at each stage of the signal processing chain when recording audio through a DAW. These stages are:

1. Analog-to-digital conversion (ADC)
2. Processing
3. Buffering
4. Digital-to-analog conversion (DAC)

We can control the performance at each these stages to decrease the round-trip latency time.

• For stages 1 and 4, we can buy an audio interface with converters that are designed to perform much faster than our stock sound card in our PC.
• At stage 2, we can upgrade our processor so it can add effects and process the input signal with less delay.
• For stage 3, we can lower the buffer sizes in our DAW and for our audio interface. Larger buffer sizes can contribute significantly to latency.
• And additionally, we can close out unneeded background processes on our computer, and increase the sample rate of our audio interface.
• Direct monitoring brings latency to zero, but it bypasses your DAW.