A Comprehensive Guide to Audio Latency in Cubase 13

A comprehensive guide to audio latency and audio system settings in Cubase 13 for recording, mixing, and mastering.

15 min readNov 19, 2023

Introduction

Hi! Recently, a new version of Cubase was released. Therefore, I think writing a new story about DAW is a good time. Today’s topic will be Cubase 13 audio settings — how to understand and configure them for the best performance and usability. I think many of you want the best performance for recording audio and responsiveness of the application while working on music within your DAW. I know that sometimes, some audio latency settings may be confusing. So here, I’ll give you a complete and detailed overview of these settings. This article will help you choose the best monitoring, mixing, or mastering workflow settings.

The Aspects of Audio Latency

Let’s take a look at the general concepts of audio performance.

Audio latency is a short period of delay (several milliseconds) between when an audio signal enters a system and when it emerges on the output [1, 2]. All software and hardware devices in the chain, for example, audio interfaces, OS, audio drivers, DAW, and VST plugins, produce some latency. Latency is not an issue for analog systems because it is near zero. Analog-to-digital (ADC) and digital-to-analog (DAC) conversion provide very little latency in nanoseconds, which is also minor and can be ignored. However, it is an issue for digital processing [1, 2]. Where audible audio latency occurs [1, 2]:

Audio interface connection type, protocol, or bus to the computer or laptop
Audio interface drivers inside the operating system
Inside the operating system (which may require additional time to process and mix streams from other applications before passing them out to the speakers)
Inside DAW (Digital Audio Workstations) (e.g., monitored tracks, native effects, plug-ins, etc.)
Inside AU, AAX, VST Plugins

An analog-to-digital (ADC) signal converter transforms an analog signal, such as a sound picked up by a microphone, from an electric guitar, bass guitar, or synth into a digital signal representation [1, 2]. Digital representation consists of numbers within some buffer (a buffer is an array with a defined size containing numbers). Digital-to-analog (DAC) converters reversely convert data to an analog audio signal sent to the speakers or other devices.

In digital recording, audio is represented by a series of numbers called “samples.” Each “sample” is a digital number representing an analog voltage value from the analog input. The higher sample rate allows for capturing more snapshots of the audio signal within one second. The audio sample rate is measured in hertz (Hz) or kilohertz (kHz). The most used sample rates are 44100 Hz, 48000 Hz, 96000 Hz, and 192000 Hz. Higher sample rates will result in slightly lower global latency but create more demand on the CPU [2, 3].

Buffer is needed to do a conversion. A buffer is a region of the computer memory storage used to store chunks of these samples. System or software often uses two or three buffers when converting the signal to prevent audio glitches. When one buffer is filled with the data, the second one is sent to the output, and this process repeats. As such, the minimum latency is equivalent to the time required for a single audio buffer to be processed within a given rate of samples per second [2, 3].

Buffer Size (number of samples) ÷ Sample Rate (Hz) = Expected Latency (ms)

For example, while running with a Buffer Size of 128 samples and a Sample Rate of 44100 Hz (44.1 kHz), an audio interface will convert the incoming signal with 2.9 milliseconds of expected latency before it appears inside the DAW [3]. Then, the same latency is needed to convert the signal from DAW and send it to the audio interface to listen to speakers or headphones. We need to add the latency from the DAW and all processing units within the DAW, like VST Plugins [2].

The formula above applies to a single signal conversion stage — for example, only on input or output. Also, the term “Overall Latency” exists. It means the sum of both latencies on input and output. All additional layers of processing may add some more latency delay. Sometimes, audio interfaces report inaccurate latency values, which can be corrected manually by a driver error compensation value [2]. Overall latency:

Input Latency + DAW/Plugins Processing Latency + Output Latency = Expected Latency (ms)

Different Delays on a Tracks

Each track within your DAW may have some delays created by the MIDI instrument or plugins. Each of these tracks will have a different number of delays. To work correctly, DAW has a compensation mechanism, which takes the longer latency delay and uses it as a reference for other tracks. Then, other tracks with the more minor delay are correctly aligned to the longer delay. The delay applied to a track’s output will be interpreted as latency when the signal is monitored in real-time [2].

Cubase provides the Channel Latency Overview window that shows the latency for the current channel [4]. In the image above, we can see that each channel has a different amount of latency. The maximum latency is 64 ms (3072 samples) for the last channel where FabFilter Pro-Q exists. Therefore, the overall latency is the latency on the channel that has the maximum value among all channels.

The Channel Latency Overview displays the latencies caused by insert effects, channel strip modules, or panners for audio-related channels in the MixConsole [4]. The Channel Latency Overview is only available if latencies are present.

Name — the name of the plug-in that causes the latency [4].
Type — indicates if the latency is caused by an insert effect, a channel strip module, or a panner [4].
Latency (ms) — Shows the latency in milliseconds. If the latency value is marked with (*), the corresponding plug-in features a Live button or a low latency mode. If you activate Constrain Delay Compensation, this mode is automatically activated. If the latency value is marked with (**), the corresponding plug-in does not feature a Live button or a low latency mode. If you activate Constrain Delay Compensation and the plug-in latency is higher than the Constrain Delay Compensation threshold, this plug-in is automatically deactivated [4].

Also, there is an additional feature called “lookahead.” This applies a slight delay on the incoming signals but provides a better processing quality. For example, some compressors or dynamic processors may have this feature [2]. A medium or large buffer is needed to convert a signal from the time domain to the frequency domain for processing or display. Therefore, spectral analyzers or equalizers may create some latency [2]. Huge sample buggers also may be needed for intelligent plugins like iZotope Ozone [2].

Constrain Delay Compensation

There are cases when you play a VST or real instrument in real-time with many plugins that give some latency. This makes playing the instrument unplayable and distracting. Constrain Delay Compensation may be activated to fix these issues [5].

If you activate Constrain Delay Compensation for instrument channels, record-enabled audio track channels, group channels, and output channels, the following happens:

For VST 3 plug-ins with a Live button and third-party VST 3 plug-ins with a low latency mode, activating Constrain Delay Compensation activates the Live button or the low latency mode for that particular plug-in [5].
For VST plug-ins with no low latency mode, activating Constrain Delay Compensation turns off that plug-in [5].

VST plug-ins that are activated for FX channels are disregarded. After recording or using a VST instrument, deactivate Constrain Delay Compensation again to restore total delay compensation [5].

You can set your threshold value for the Constrain Delay Compensation value inside Cubase settings. Just open Cubase → Settings → VST window. This means all plugins that give the latency more than this value will be turned off.

Does UI Graphics impact audio delay?

No, graphics in DAW should not impact audio delay. The UI and graphics are processed in different process threads. Therefore, UI and graphics may lag, but it is expected not to impact the audio latency. The UI may lag when the UI shows the data based on the audio processing results or, for example, Fourier transformations [2].

How to find a suitable audio latency balance

As I said, latency is the time needed to process the sound. For daily usage of your DAW, you need to take into account your audio interface model, audio interface drivers, DAW, audio settings, plugin settings, and the count of plugins, OS, CPU, memory, and all additional processing. The faster your computer, the more tracks, effects, and EQ you can play. The human ear can recognize any delays. The latest research from Aalto University says that human ears are sensitive even to half of millisecond delays [6]. But generic studies say:

Delays up to 4 ms are great for monitoring and recording.
Delays from 4 ms to 7 ms are acceptable for monitoring and recording.
Delays of more than 8 ms are distracting for monitoring and recording.
Delays of more than 8 ms are completely allowable for mixing and mastering but not for recording or monitoring.

Please consider that each person has its characteristics and hearing nuances. Therefore, these values may vary for each person. More information is represented within Head Related Transfer Function (HRTF) research [7].

Cubase 13 Audio Settings

Let’s take a closer look at Cubase 13 audio settings. We’ll start with opening the main window with the settings. This information is taken from the official Cubase 13 operational manual because this is the first source of truth [8]. Please check official references for more details. You can open these settings by clicking on Studio → Studio Setup → Audio System.

Studio Setup

ASIO Driver — selected audio driver for the connected audio interface[8].
Release Driver when Application is in Background — releases the driver and allows other applications to playback via your audio hardware even though Cubase is running [8]. Cubase will not use this audio driver in this mode while in the background mode. This mode may be usable for Windows users when switching between multiple software applications using the same audio driver.
Input Latency — shows the input latency of the audio hardware [8].
Output Latency —shows the output latency of the audio hardware [8].
ASIO-Guard Latency — shows the ASIO-Guard latency [8].
HW Sample Rate — shows the sample rate of your audio hardware [8].
HW Pull Up/Down — shows the pull up/down status of the audio hardware [8].

Advanced Options

Processing Precision — allows you to set the audio processing precision to 32-bit or 64-bit float. Depending on this setting, all channels are processed and mixed in a 32-bit floating-point or 64-bit floating-point format. A processing precision of 64-bit float can increase CPU load and memory consumption. To show all plug-ins that support 64-bit float processing, open the VST Plug-in Manager from the Studio menu and activate Show Plug-ins That Support 64-bit Float Processing in the Display Options pop-up menu. VST 2 plug-ins and instruments are always processed with 32-bit precision [8].
Activate Multi Processing — you can distribute the processing load evenly to all available CPUs. This way, Cubase can fully use the combined power of multiple processors [8]. I recommend keeping this option turned on.
Activate ASIO-Guard — activates the ASIO-Guard. This is only available if Activate Multi Processing is activated, too [8].
ASIO-Guard Level — allows you to set the ASIO-Guard level. The higher the level, the higher the processing stability and audio processing performance. However, higher levels also lead to an increased ASIO-Guard latency and memory usage [8].
Audio Priority (Windows only) —this setting should be set to Normal, if you work with audio and MIDI. If you do not use MIDI at all, you can set this to Boost [8].
Activate Steinberg Audio Power Scheme (Windows only) —if this option is activated, all power safe modes that have an impact on real-time processing are deactivated. Note that this is only effective for very low latencies, and that it increases the power consumption [8].
Disk Preload — allows you to specify how many seconds of audio are preloaded into RAM prior to starting playback. This allows for smooth playback [8].
Adjust for Record Latency — if this is activated, the plug-in latencies are taken to account during recording [8].
Record Shift — allows you to shift the recordings by the specified value [8].

Audio System ASIO Driver Setup

When you click on the name of your audio interface — in my case, Scarlett 4i4 USB you’ll see additional options. Here, you can set up your audio interface driver. Input and Output latency is the same as we saw before. Also, it is possible to turn on the external clock, direct monitoring, and configure ports [9].

Audio System → Scarlett 4i4 USB Settings

Control Panel — opens the control panel for the audio hardware [9].
Input Latency — shows the input latency of the audio driver [9].
Output Latency — shows the output latency of the audio driver [9].
Clock Source — allows you to select a clock source [9].
Externally Clocked — activate this option if you use an external clock source [9].
Direct Monitoring — activate this option to monitor via your audio hardware and to control it from Cubase [9].

In the Ports section, the following options are available:

Reset — allows you to restore the default port names and to enable the visibility for all ports [9].
I/O — the port input/output status [9].
Port System Name — the system name of the port [9].
Show As — allows you to rename the port. This name is used in the Input Routing and Output Routing pop-up menus [9].
Visible — allows you to activate/deactivate audio ports [9].
State — the state of the audio port [9].

At the bottom of the page, the following options are available:

Reset — sends a reset signal to the active ASIO device and restarts the audio processing. This can solve problems with audio playback. This leads to a short interruption of the playback [9].

CoreAudio Device Settings

Clicking on the Control Panel opens a new dialog with the Buffer Size setting. Select the buffer size you want. Remember that a smaller buffer needs a more powerful CPU and provides more negligible audio latency. The bigger buffer size can work with less powerful CPUs but provides more latency and is unsuitable for near real-time monitoring or recording. Therefore, experiment with these settings to determine which Buffer Size value is better for your setup. I prefer not to select Set Device Attenuation To 0 dB. So you can also leave it as it is.

ASIO Guard

The ASIO-Guard allows you to shift as much processing as possible from the ASIO real-time path to the ASIO-Guard processing path. This results in a more stable system. The ASIO-Guard allows you to preprocess all channels and VST instruments that do not need to be calculated in real-time. This leads to fewer dropouts, the ability to process more tracks or plug-ins, and the use of smaller buffer sizes. High ASIO-Guard levels lead to an increased ASIO-Guard latency. For example, when you adjust a volume fader, you will hear the parameter changes with a slight delay. The ASIO-Guard latency, in contrast to the latency of the audio hardware, is independent of live input. The ASIO-Guard cannot be used for real-time-dependent signals and external effects and instruments. If you activate the monitoring for an input channel, a MIDI instrument, or a VST instrument channel, the audio channel and all dependent channels are automatically switched from ASIO-Guard to real-time processing and vice versa. This results in a gentle fade out and fade in of the audio channel [10].

Audio Performance Window

The Audio Performance panel shows the audio processing load and the hard disk transfer rate. This allows you to ensure you do not run into performance problems when adding effects or plug-ins, for example [11].

To open the Audio Performance window, select Studio → Audio Performance.

Real-Time — shows the average load of all audio realtime processes [11].
ASIO-Guard — shows the average load of processes that can be preprocessed [11]. Preprocessing only takes effect if you activate Activate ASIO-Guard on the Audio System page of the Studio Setup dialog [11].
Peak — shows the processing load in the real-time path of the audio engine. The higher this value, the higher the risk that dropouts occur [11].
Processing Overload — the overload indicator on the top right indicates dropouts. Dropouts occur if the processing load exceeds 100% or if the audio engine is restarted due to the internal detection of excessive processing delay. For example, this can occur if the preprocessing buffer runs empty as a result of the real-time load exceeding the limits. If the overload indicator lights up, decrease the number of EQ modules, active effects, and audio channels that play back simultaneously. You can also activate the ASIO-Guard [11].
Disk Cache — shows the hard disk transfer load [11].
Disk Cache Overload — the overload indicator to the right of the disk indicator lights up if the hard disk does not supply data fast enough. If it lights up, use Disable Selected Tracks to reduce the number of tracks playing back. If this does not help, you need a faster hard disk. To reset the overload indicator, click its display. In the Audio Performance category of the Key Commands, you can also assign a key command for this. You can show a simple view of the performance meter on the Transport panel and on the Project window toolbar. These meters only feature the average and the disk indicator [11].

VST Plug-in Manager

You can manage VST effects and VST instruments in the VST Plug-in Manager window. To open the VST Plug-in Manager window, select Studio → VST Plug-in Manager [12]. You can find more information about the selected plugin by clicking the small icon “i” on the bottom left. The Plug-in Information panel shows the ASIO-Guard setting, the count of ts, I/O configuration, and the Latency in samples and other settings. ASIO-Guard was described in this article in the chapter above. You can enable or disable this setting for a specific plugin.

Summary

In this article, we reviewed the many aspects of audio latency. Completed a detailed review of all audio settings for Cubase 13, audio drivers, audio interface, audio performance window, and VST Plug-in Manager. Technically, there are many details that we need to know to fine-tune the DAW. But overall, the rule is still the same:

Small buffer size — faster response time between the input and output, but needs a powerful CPU. If the CPU does not have enough processing power, the sound clicks, pops, or is corrupted. This mode is suitable for monitoring and recording.
Big buffer size — slower response time between the input and output. Can be used for less powerful CPUs. A slower response time leads to audible sound delay between sound input and output, which is unsuitable for monitoring and recording. But this mode is suitable for mixing and mastering.

I hope this article helped you to have a better understanding of how audio latency and Cubase settings work.