What should my audio latency be?

What is Audio Latency?

Audio latency refers to the time delay between when an audio signal enters a system and when it emerges. It is measured in milliseconds (ms). Latency occurs because it takes time for the audio interface, computer processor, and Digital Audio Workstation (DAW) software to process and output the audio (Digital Audio Latency Explained).

The causes of latency include analog-to-digital conversion, buffering, digital signal processing, transmission time, and the speed at which the audio interface and computer system can process the audio data (What Audio Latency Is And How To Reduce It). Higher latency results in a noticeable delay between playing a note and hearing it come out of the speakers.

For musicians recording into a DAW, high latency can make it very difficult to play in time and with a good “feel.” Even small amounts of latency can negatively impact the timing of rhythm parts or a musician’s ability to properly double track a part (Latency (audio)). Reducing latency is crucial for getting the best performance during recording and monitoring.

Acceptable Latency Levels

In the audio production world, there are general guidelines for acceptable latency levels depending on the application and use case:

For tracking and recording live instruments, latency below 10ms is ideal. Anything above 10ms can cause timing and synchronization issues that throw off musicians during performance. Experts recommend keeping latency below 5ms when tracking vocals or instruments.

During mixing and mastering, latency up to 30ms is generally not noticeable and considered acceptable. Mix engineers can comfortably work with buffer sizes of 128 to 256 samples at standard sample rates.

For DJing and electronic music production, latencies below 20ms are recommended for rhythmically tight performances. DJs using software like Traktor or Serato need latencies between 5-10ms for perfect sync between decks.

Gaming audio requires very low latencies below 20ms, since timing is critical for gameplay and immersion. Gamers notice and complain about audio lag during gameplay.

Video post-production can handle latencies around 30-40ms without issue. However, when recording voiceovers or foley, it’s best to enable low-latency monitoring.

The key is using the lowest latency possible for the task at hand without creating performance issues or glitches. Understanding these general standards helps optimize systems for different use cases.

Setting Buffer Size

The buffer size has a direct impact on audio latency. The buffer is a short interval of time that the audio interface uses to process incoming and outgoing audio signals. A larger buffer size leads to more latency, while a smaller buffer reduces latency. However, if the buffer is too small, you may get audio dropouts, clicks, and pops (Source).

To find the ideal buffer size, start with a large buffer like 1024 samples and slowly reduce it until you experience performance issues. The lowest usable setting without audio problems is optimal. For live playing, aim for 10-20ms latency. For recording, up to 40ms may be acceptable. Use a latency calculator to estimate your latency based on sample rate and buffer size.

Lower buffer sizes require a faster CPU and hard drive to handle the reduced processing time. Upgrading computer hardware can help achieve lower latencies. Some audio interfaces and DAWs have a “Low Latency Mode” to automatically optimize settings.

Driver Optimization

One of the most important things you can do to reduce audio latency is to make sure you have the latest audio drivers installed. Audio drivers act as the interface between your audio software and the computer’s hardware. Outdated drivers can result in higher latency and glitches during recording and playback.

First, you’ll want to find out what audio interface or onboard sound chip you are using. Then visit the manufacturer’s website and download the latest drivers. Make sure to completely uninstall the old drivers first before installing new ones.

After updating to the newest drivers, look in the control panel or settings for your specific audio interface. There may be a “Performance” tab or settings to reduce buffer size or tweak other parameters that affect latency. For example, lowering the buffer size is one way to reduce latency but can lead to glitches if set too low.

Additionally, disable any extra processing like effects in the driver settings as these can add extra latency. Turning off CPU throttling and power savings in the control panel may also help.

According to this article, opening the Windows audio settings and disabling unused audio interfaces can improve performance of the primary interface. Optimizing drivers to reduce overhead will lower latency allowing for more real-time monitoring and playback.

Using Low-Latency ASIO Drivers

One of the most effective ways to reduce audio latency is to use a dedicated low-latency ASIO (Audio Stream Input/Output) driver. ASIO drivers allow audio interfaces and DAWs to communicate more efficiently by bypassing the default Windows audio subsystem and reducing buffer sizes.

Steinberg is a leading creator of ASIO driver technology and their drivers are included with many professional audio interfaces like those from RME and Focusrite [1]. The key benefits of using ASIO drivers include:

Much lower latency versus standard Windows drivers

More stable performance and reduced risk of glitches/dropouts
Lower CPU usage and ability to use smaller buffer sizes
Support for higher sample rates up to 192kHz

To take advantage of ASIO, you need to configure your DAW and audio interface to use the ASIO driver instead of alternatives like DirectSound or WDM. Exact setup steps vary by DAW, but generally involve selecting the specific ASIO driver for your interface in the audio settings. This unleashes the full low-latency performance potential of your hardware.

For those without an audio interface, the free ASIO4All driver can provide some benefits over standard Windows audio [2]. However, dedicated ASIO drivers written specifically for your interface will provide lower latency and better performance.

Optimizing the DAW

One of the most important things you can do to reduce audio latency is optimize the settings in your DAW (digital audio workstation). This primarily involves adjusting the audio buffer size and tweaking other latency-related preferences.

The buffer size determines how much audio data is processed at once. A smaller buffer means audio is processed in smaller chunks, leading to reduced latency. But this places a heavier load on your CPU. As explained on edmprod.com, “A low buffer size will reduce latency, but increase the strain on your CPU. On the other hand, increasing the buffer size will lighten the CPU’s load but also increase latency.”^[1]

So you’ll need to find the right balance based on your system capabilities. Try starting with a buffer of 128 or 256 samples, and go lower if you can without hearing pops, clicks or distortion. Refer to your DAW’s manual for specific instructions on changing the buffer size.

Next, disable any latency-inducing plugins or effects during recording and live playing. As Puget Systems recommends, “Disable track monitoring effects while recording audio tracks. Effects like reverb and delay add a significant amount of latency.”^[2] You can enable them again during mixing.

Also try increasing the audio playback priority in your DAW’s settings. This allocates more CPU resources to audio processing to help minimize latency. Disable power saving options as well.

Adjusting these DAW settings is key for establishing a low-latency workflow before even considering new audio interfaces or equipment.

[1] https://www.edmprod.com/daw-latency-and-cpu/

[2] https://www.pugetsystems.com/support/guides/daw-driver-latency/

Using External Audio Interfaces

One of the most effective ways to reduce audio latency is by using an external audio interface instead of your computer’s built-in soundcard. External interfaces connect to your computer via USB, Thunderbolt or FireWire and feature high-quality AD/DA converters and audio drivers optimized for low latency.

Benefits of using an external interface include:

Much lower latency compared to built-in soundcards
Higher-quality audio conversion
ASIO/Core Audio drivers designed for pro audio

Multiple inputs and outputs
Audio metering and monitoring capabilities

Some reputable brands of external interfaces include Focusrite, Presonus, RME, Apogee, MOTU, and Universal Audio. Popular low latency models are the Focusrite Scarlett Solo, Presonus AudioBox, and RME Babyface Pro.

To use an external interface, simply install the drivers and connect it to your computer via the included USB/Thunderbolt cable. Open your DAW and select the interface as your audio device. The lower latency and improved audio quality makes mixing and recording much easier.

Upgrading Computer Hardware

One of the most effective ways to reduce audio latency is by upgrading key computer hardware components like the CPU, RAM, and storage drives.

Investing in a faster multi-core CPU can significantly improve real-time audio performance. Modern CPUs with higher clock speeds, more cores, and hyperthreading will allow your DAW to process more audio streams simultaneously with lower latency. For intensive projects, an Intel Core i7 or i9 CPU or AMD Ryzen 7 or 9 CPU is recommended.

Adding more RAM allows your computer to load samples, plugins, and sessions faster into active memory instead of relying on virtual memory swapping to a hard drive. For pro audio production, 16-32GB of RAM is recommended depending on your typical project size.

Using fast SSD drives instead of traditional hard disks for your operating system, sample libraries, and project files greatly reduces load times and streaming latency when accessing samples. Top-tier NVMe SSDs provide the fastest read/write speeds for optimal real-time streaming performance.

Upgrading to cutting-edge hardware components like a 12th Gen Intel CPU, DDR5 RAM, and PCIe 4.0 NVMe SSDs can provide a significant reduction in system latency compared to dated hardware.

Tweaking Plugin Settings

One common source of latency comes from plugins and virtual instruments. Each plugin will add a small amount of processing latency, which can quickly add up when using multiple plugins on a track. There are a few ways to reduce plugin latency:

Adjust the latency settings in plugin interfaces – Many plugins have a “Low Latency” mode or settings to reduce latency. For example, virtual synthesizers often have a control for reducing the sample buffer size. Setting plugins to their minimum latency mode can greatly reduce delay.

Freeze resource intensive plugins – Freezing or rendering plugins like reverbs and virtual instruments to audio will prevent them from processing in real-time and adding latency. This essentially “prints” the plugin’s output as an audio file played back with zero latency.

Avoid oversampling – Some plugins use oversampling techniques to improve audio quality, but this requires additional processing and adds latency. When tracking, try disabling oversampling in plugins.

Route plugins in series – Adding plugins in series means each one only adds a small amount of latency, instead of multiplying the latency if they were in parallel. Series routing helps minimize monitoring delay.

Use DSP-efficient plugins – Certain plugins are optimized for low latency by reducing the amount of math required in the background. Using efficient plugins, especially during recording, can help achieve minimal latency.

By adjusting plugin settings, using efficient routing, and freezing tracks, you can keep plugin latency to a minimum for responsive monitoring and tracking.

Summary

When it comes to audio latency, the ideal levels can vary considerably depending on your use case and equipment setup. For live monitoring while recording, a latency below 10 ms is generally recommended to avoid distracting delay. During mixing and mastering, levels below 25 ms are preferred. For MIDI sequencing, composers and producers often aim for latencies under 7-8 ms. However, for certain plugins and virtual instruments, higher buffer sizes and increased latency may be unavoidable.

There are several key strategies you can use to optimize your audio interface and digital audio workstation (DAW) for lower latency performance:

Adjust your audio interface’s buffer size. Lower sizes reduce latency but increase processing demands.
Enable ASIO drivers for your interface to bypass default system audio.
Minimize plugins on tracks and use low-latency plugins when possible.

Toggle on “Low Latency Monitoring” in your DAW if available.
Bounce MIDI tracks to audio to reduce plugin latency.
Upgrade to higher-spec computer hardware like more RAM, faster CPU.

With the right optimizations, you can achieve latency levels appropriate for most studio production and live performance needs.