What Is Audio Latency?

What is Audio Latency?

Audio latency refers to a short delay between when an audio signal enters a system and when it emerges. This delay is an unavoidable byproduct of the steps involved in digitizing, processing, and outputting an analog audio signal.

Latency occurs because it takes time for audio interfaces and digital audio workstations (DAWs) to convert analog signals to digital data, process that data, and then convert it back to analog for playback. The greater the latency, the more delayed or out-of-sync playback will be compared to the original input signal.

Musicians rely on having near instantaneous monitoring of their vocals, guitar, electronic instruments, etc. to properly perform and record tracks. High latency can cause a distracting echo or delay between playing a note and hearing it, throwing off timing and groove.

The ideal is latency low enough that it is imperceptible to the musician, generally considered to be under 10-20 milliseconds. More latency than this can make it challenging to play in time and get a natural sounding performance.

Audio latency occurs due to the processing demands of audio interfaces, computer CPU load, audio drivers, DAW settings, plugin effects, and other factors. Managing latency involves optimizing hardware, software, and system resources to lower delay during recording and monitoring.

Measuring Audio Latency

Audio latency refers to the time delay between an audio signal entering a system and that signal being audible from the output. Latency is measured in milliseconds (ms). The lower the latency, the more responsive a system will feel.

There are several tools available for measuring audio latency. For example, the open-source application Audio Latency Benchmark can be used to test latency on Windows machines (Source 1). This involves connecting a loopback cable from the audio interface output to the input, then measuring the delay as audio passes through. Another simple approach is to record a sharp handclap sound and measure the latency between the visual clap and audible clap on the recording (Source 2). There are also online tools that can estimate Bluetooth latency.

Causes of Audio Latency

There are several key factors that contribute to audio latency during recording and playback:

Audio Interface

The audio interface is responsible for converting analog signals to digital audio data and vice versa. The speed and quality of the analog to digital converters impacts latency. Higher quality converters generally have lower latency ¹.

CPU Load

The computer’s CPU processes the digital audio, adding buffers and effects. High CPU loads from other programs can cause processing delays and increased latency ².

Driver Issues

Outdated, incompatible, or faulty drivers for audio components like the interface can introduce latency during processing. Keeping drivers updated improves performance ³.

Buffer Settings

Audio buffers help prevent glitches but add latency. Lower buffer sizes reduce latency but increase the risk of audio dropouts if the system can’t keep up.

Reducing Latency in Audio Interfaces

One of the main ways musicians and audio engineers can reduce latency is by choosing a low-latency audio interface. Many modern interfaces advertise very low latency, but the actual amount can vary substantially depending on the drivers and your computer system.

Optimizing the drivers for your specific audio interface is important for achieving the lowest latency. Updating to the latest drivers can help improve performance. Some interfaces have control panel settings to adjust the buffer size and sample rate, which can reduce latency at the cost of increased CPU usage.

Utilizing an audio interface’s direct monitoring feature routes the input signal directly to the headphone output before going to the computer. This avoids the latency induced while recording and monitoring through the DAW software.

Choosing an optimized audio interface that provides the right balance of latency and performance for your needs and system capabilities is key for musicians who want to minimize lag and play in time.

Optimizing CPU and Drivers

Updating your audio interface and system drivers to the latest stable versions can help reduce latency. According to this article, outdated drivers may contain bugs or inefficient code that negatively impacts performance. Check your hardware manufacturers’ websites regularly for driver updates.

Adjusting your audio interface’s buffer size, sometimes called the latency buffer, can also decrease latency. Lower buffer sizes reduce the time audio data sits in the buffer before being processed, but can increase the CPU workload. Find the lowest buffer size you can without hearing pops, clicks, or distortion in your audio. Refer to your interface’s user manual for changing buffer size.

Try to avoid taxing your CPU with other intensive applications when recording and mixing audio. According to this Sound on Sound article, high CPU loads can cause audio dropouts. Consider upgrading your CPU if latency persists after closing other programs.

Latency Introduced During Recording

Latency can be introduced during the recording process due to the analog-to-digital (A/D) and digital-to-analog (D/A) conversion that takes place. When an analog audio signal enters an audio interface, it must be converted into digital data through A/D conversion before it can be recorded into a DAW. This conversion process takes a small amount of time and introduces a bit of latency.

Similarly, many recording setups utilize direct monitoring, where the input signal is routed back out to the headphones with minimal latency. This allows performers to hear themselves while recording. However, the signal must go through D/A conversion on the way to the headphones, which again adds a small amount of latency (Sweetwater). Reducing buffer sizes can minimize conversion latency, but very small buffers can create performance issues.

The monitoring path is another source of recording latency. Even using direct monitoring, there is some latency as the signal travels through the mic preamp, A/D conversion, mixing, D/A conversion, and output. Using low-latency audio drivers and interfaces optimized for real-time audio can reduce monitoring path delays.

Latency During Playback

Playback latency refers to the delay between when audio is sent from the audio interface to when it is actually heard through the speakers or headphones. This occurs in the output path as the digital audio data is converted back to an analog signal and then amplified to drive the transducers that produce the sound waves.

The digital-to-analog conversion process takes a small amount of time, as the audio interface’s DAC chip converts the binary digital audio data to a continuous analog waveform. Higher sample rates and bit depths require more complex DAC circuitry and algorithms, which can increase latency.

Analog audio signals also take time to propagate through cables and amplification stages before reaching the speakers/headphones. The transient response of the transducers themselves can introduce a small delay as the diaphragm physically moves to produce sound waves.

Monitoring adds additional latency, as there is a feedback loop from the analog inputs to the analog outputs. The roundtrip conversion, processing, and routing of the monitored signal increases overall latency. Direct monitoring modes reduce this by routing input signals directly to the outputs, bypassing digital conversion and processing.

In summary, the output path from DAC to speakers/headphones, analog propagation delays, and monitoring architectures all contribute small amounts of playback latency.

Balancing Latency and Performance

The ideal audio latency level depends on the use case and desired performance. According to Practical Music Production, “A latency time of around 10ms or less usually means that it won’t affect the recording process. Above 10 ms, the effect starts to become noticeable.” source

Very low latency matters most for live performance situations where monitoring is critical, like playing guitar through an amp simulator. Musicians need to hear what they play as soon as possible to properly play in time. Ask.Audio recommends 8-12 ms latency for live performance monitoring. source

Higher latency in the 20-50 ms range may be perfectly acceptable for recording and mixing scenarios. While noticeable, a slight delay usually does not impede the creative process in the studio. Higher buffer sizes that increase latency can also optimize the system for enhanced processing power and stability.

In summary, ultra-low latency under 10 ms is ideal for live monitoring but not always necessary. Moderate latency between 10-20 ms provides a good blend of performance for both live playing and studio recording. Higher latency above 20 ms can be tolerated in the studio if needed to improve system capabilities.

Solutions for Musicians

There are a few key ways that musicians and producers can reduce latency when recording and monitoring:

Direct Monitoring

Direct monitoring routes the input signal directly to the headphone output before it goes through the computer. This avoids any latency added by the DAW or audio interface. Most modern audio interfaces have a direct monitoring option.

Low-Latency Audio Interfaces

Using an audio interface designed specifically for low latency can greatly reduce monitoring lag. For example, the RME Babyface interface has <1ms latency, which is imperceptible.

Reducing Plugin Use

Each plugin on a track adds latency as the signal is processed. Limiting use of plugins while recording and using low-latency plugins where possible can help. Some DAWs also allow delay compensation to align tracks.

With proper setup using the above techniques, musicians can monitor themselves in real-time without audible latency. This results in a much better recording experience and performance.

Summary

The key takeaways about audio latency are:

• Latency refers to a short delay between an audio signal entering a system and being outputted, measured in milliseconds.
• Causes include audio interfaces, drivers, CPU load, and recording and playback buffers.
• Reducing latency requires optimizing hardware/software and finding the right balance between latency and performance.
• Solutions like low-latency audio interfaces and drivers, optimized DAW settings, and dedicated systems help.
• Managing latency is crucial for musicians, audio engineers and anyone doing real-time audio work so monitoring is in sync.

In summary, audio latency is an inevitable but manageable part of digital audio systems. While zero latency monitoring is impossible, latency can be minimized through optimized software and hardware choices. Finding the right balance is necessary so latency is low enough to not disrupt real-time workflows. This allows musicians, producers and audio engineers to monitor with confidence during recording and playback. Keeping latency under control makes all the difference in efficient, professional audio work.