What is low latency mode audio?

What is Low Latency Audio?

Low latency audio refers to audio that has a very short delay between the time a sound is produced and when it is heard through speakers or headphones. Normal audio latency is typically around 100-200 milliseconds, while low latency audio aims for latency under 10 milliseconds (ms) (Source A).

Low latency is critical for applications like live sound, recording, and communication that require near real-time monitoring. Musicians need to hear their vocals or instruments coming through speakers or headphones with minimal delay in order to perform properly and avoid disorientation. Recording engineers require low latency monitoring when tracking audio. Applications like gaming, phone calls, video conferencing, and live streaming also rely on low latency audio for natural timing of voice, video, and audio elements.

Achieving low latency requires optimizing audio drivers, operating systems, audio interfaces, digital audio workstations, and other components to process and deliver audio signals faster. Advanced buffering techniques like ASIO (Audio Stream Input/Output) and WDM (Windows Driver Model) KS (Kernel Streaming) reduce the delay introduced at each step of the digital audio pathway (Source B).

Measuring Audio Latency

Several metrics are used to measure audio latency. An important one is roundtrip latency, defined as the time it takes for an audio signal to go into a system, get processed, and emerge from the output 1. This accounts for the total delay incurred as the signal passes through analog-to-digital converters, digital processing, buffering, and digital-to-analog conversion back to analog. The lower the roundtrip latency, the closer the output is in time to the original input signal.

Another common metric is buffer size. This refers to the amount of temporary audio data storage required as the signal is processed. Larger buffer sizes can accommodate more processing but also introduce more latency. A buffer size of 128 samples may equate to around 3 ms of latency at a 44.1 kHz sample rate 2. For real-time applications like live sound and recording, the goal is to minimize buffer sizes to reduce roundtrip latency while still allowing reliable processing.

For professional audio applications, a roundtrip latency under 10 ms is generally desired. Under 3 ms is ideal for live monitoring while recording and playing virtual instruments. However, average latencies around 25-100 ms are more common in consumer devices and can still be acceptable for casual listening.

Causes of High Audio Latency

There are several main causes that can lead to high latency in audio systems:

  • Sound buffers – Audio interfaces and DAWs use buffers to help process audio, but large buffer sizes lead to increased latency.
  • Audio drivers – Outdated, generic, or badly coded audio drivers can introduce latency issues.
  • Operating system – Unoptimized OS settings like power management can add small amounts of latency.

Other factors like plugins, effects, MIDI connections, and hardware limitations can also degrade audio response times. But the key variables are sound buffers, audio drivers, and OS optimization. Using low buffer sizes, efficient ASIO/Core Audio drivers, and configuring systems for performance are essential for reducing latency.

As explained in the Vintage Vinyl News guide, addressing these main problem areas allows musicians and audio engineers to achieve much lower latencies.

Reducing Latency in Audio Drivers

One of the main causes of high audio latency is improperly configured audio drivers. Most audio interfaces and sound cards come with drivers that allow you to adjust settings to optimize for low latency performance. Here are some key settings to configure in your audio drivers to reduce latency:

Set the audio buffer size or block size to 128 samples or lower. This setting determines how many samples are processed at once. Lower values reduce latency but can increase risk of audio dropouts if your system can’t handle the smaller buffer size. Start with 128 samples and adjust as needed.

Enable audio buffer optimizations if available. Some drivers have an “optimize for low latency” checkbox or mode that tweaks how the buffer behaves. This can help achieve lower latency.

Disable unnecessary processing like EQs, enhancements, and effects. Any extra DSP done in your driver adds latency. Turn off anything you don’t need.

Lower the sample rate if possible. 44.1-48 kHz provides the lowest latency for most applications. Higher rates like 96 kHz can potentially double your latency.

Use ASIO drivers for pro audio interfaces. ASIO provides lower latency performance compared to standard Windows audio drivers.

Keep your drivers up to date. Newer drivers often include latency optimizations not found in older versions.

Consult your audio interface manual for any special low latency modes or additional tweaks to reduce buffer sizes as much as possible without audio dropouts.

With careful configuration of buffer sizes and driver settings, audio latency can be reduced significantly without compromising system stability.

Optimizing Operating Systems

The settings and configuration of your operating system can have a significant impact on achieving low latency audio performance. Both Windows and macOS provide ways to optimize the OS for real-time audio processing. Some key settings to adjust include:

  • Setting the CPU priority for audio applications to high or real-time can allocate more processing resources and reduce latency. This is done through the Task Manager in Windows or Activity Monitor on macOS.
  • Disabling unnecessary background processes and visual effects frees up RAM and CPU cycles. Turning off things like animations, transparent windows, desktop gadgets can help.
  • Increasing the audio buffer size allocates more memory for the OS and audio interfaces to process audio data efficiently.
  • Reducing the number of software interrupts and setting the thread priority higher for audio applications minimizes task switching and improves timing.

Tools like ASIO4ALL, WASAPI, or Core Audio allow audio applications to bypass default Windows or macOS audio pipelines and directly access hardware for lower latency. Optimizing your OS settings compliments these audio driver frameworks for the best performance.

Audio Interface Considerations

The design and drivers of your audio interface play a key role in achieving low latency audio. Most basic computer sound cards have too much built-in latency for real-time monitoring and tracking. Upgrading to a dedicated external audio interface with optimized drivers is essential.

Key factors that impact audio interface latency include:

  • Driver architecture – ASIO drivers provide lower latency than standard Windows audio drivers. macOS Core Audio drivers are optimized for low latency.
  • Bit rate/sample rate – Higher sample rates allow faster data transfer but increase buffer sizes. Consider 96-192 kHz for latency-critical tasks.
  • Converter quality – High-end converters add less conversion latency between analog and digital domains.
  • Direct monitoring – Interfaces with `zero-latency’ direct monitoring avoid going through computer buffers entirely.
  • USB/Thunderbolt connectivity – Provides more bandwidth than older Firewire or MIDI interfaces.
  • Buffer size – Lower buffer sizes reduce latency but increase risk of pops/clicks and crashes.

Leading low-latency audio interfaces include RME’s Fireface series (https://www.electronicshub.org/best-low-latency-audio-interface/), Presonus Audiobox (https://www.reddit.com/r/edmproduction/comments/zi99az/recommended_audio_interface_for_no_latency/), and models from Focusrite, Apogee, Universal Audio, and Avid.

Digital Audio Workstations

Digital audio workstations (DAWs) play a critical role in achieving low latency audio production. The way a DAW is designed and optimized can determine how capable it is of low latency performance.

Certain DAWs like Ableton Live and FL Studio are specifically engineered for low latency audio, often sacrificing some recording and editing features in favor of real-time performance. They utilize audio engines and drivers built for efficiency and reducing delay.

However, even full-featured DAWs like Pro Tools, Logic, and Cubase are capable of achieving low latency with the proper settings and optimizations. This usually involves reducing the audio buffer size, sometimes as low as 32-64 samples. Lowering sample rate can also help, albeit at the expense of fidelity.

Multi-core processors and optimization of CPU loads are critical DAW factors. Poorly optimized DAWs may overload a single core and introduce latency spikes, while efficient DAWs spread real-time tasks smoothly across cores and threads. Some find better results disabling hyperthreading to reduce thread contention.

Choosing the proper audio interface drivers for your DAW setup is also important, as is adjusting the project sample rate to align with the hardware sample rate. Too great a discrepancy can require sample rate conversion and introduce latency.

Careful track and plugin routing to avoid latency-inducing serial processing chains can help too. Overall, settings like buffer size along with DAW and system optimization offer powerful ways to achieve low latency workflows.

As an example, one forum user reduced latency from 1000+ samples to 128 samples stable on a Dual Xeon system by optimizing CPU Affinity in their DAW after struggling with poor real-time performance initially (source).

Plugins and Effects

When using plugins and effects, it’s important to be mindful of latency. Many plugins, especially those that apply complex processing like reverb, can introduce significant latency. There are a few strategies to maintain low latency when using plugins:

Use native plugins or plugins specifically designed for low latency whenever possible. Some plugins advertise “zero latency” modes that optimize the processing for real-time use (Zero-latency plugins – list – Plugins).

When using plugins that induce latency, place them on aux sends/buses rather than directly on tracks. This prevents the latency from stacking up across multiple plugins.

Use clip delay compensation features in your DAW to automatically align tracks and compensate for any plugin latency.

Be selective with plugins – sometimes less is more. Limit use of lush reverbs and other intensive effects while recording/tracking. Save those for mixing.

Consider using analog outboard effects while recording instead of plugins to avoid latency entirely.

Hardware Synths and MIDI

When using external MIDI hardware synthesizers and controllers, latency can become an issue. The time it takes for MIDI messages to be sent from the computer to the hardware and back creates latency. There are a few things you can do to optimize MIDI and reduce latency when using external gear:

Use a fast, dedicated MIDI interface rather than relying on MIDI over USB. Devices like the MOTU MIDI Express or Roland UM-One provide very low latency MIDI connectivity. According to users on forums like Reddit, dedicated MIDI interfaces have latency in the sub-millisecond range.

If possible, connect synths digitally using MIDI over USB or MIDI DIN. This avoids the need to convert from digital to analog and back, which adds latency. Using MIDI cables to create a direct digital connection between devices results in minimal latency.

Optimize your operating system’s and audio interface driver latency. This will ensure MIDI timing is accurate. You may need to tweak your audio buffer settings or use a lower sample rate to achieve ideal latency.

Use MIDI clock sync to keep everything locked together. Clock signals sent via MIDI or DIN sync will keep hardware and software in time.

Overall, reducing reliance on analog conversions and optimizing drivers provides the best results for integrating latency-sensitive hardware into a hybrid studio.

Key Takeaways

When producing and recording music, low latency audio is essential for getting the best performance and timing. Latency refers to the delay between an audio input and output, which can negatively impact musicians’ timing and feel. By optimizing your audio interface, DAW, plugins, and computer system, you can achieve latencies under 10 milliseconds, which is imperceptible.

There are many benefits to optimizing audio latency:

  • Musicians can record takes more accurately by hearing audio playback immediately.
  • Software instruments and MIDI drum pads feel more responsive.
  • Monitoring live audio has no noticeable delay.
  • Effects like reverb and compression feel more “real time”.
  • Collaborators can jam together remotely with no lag.

While specialized low latency audio hardware and drivers used to be required, modern computers, audio interfaces, DAWs, and drivers can now achieve professional levels of latency. By understanding the various components in your audio chain and optimizing each one, you can take your music production to the next level.

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