Is there an app to adjust audio latency?

What is Audio Latency?

Audio latency refers to a short time delay between an audio input and output. For example, when recording audio on a computer, there will be a small delay between the sound being recorded by a microphone and that audio being played back through speakers or headphones.

This latency is caused by the processing time required for the audio interface and computer to convert the analog audio input to digital data, process that data in software like a Digital Audio Workstation (DAW), and then convert it back to analog audio output. Additional factors like audio buffer settings, driver performance, and overall system capabilities also impact the total latency.

Too much latency can make recording and monitoring difficult for performers and engineers. Even small delays can cause a jarring experience if the performer hears their vocals or instrument delayed back in the headphones mix. This can negatively impact timing and pitch. Audio engineers also rely on low latency monitoring to make precise mixing and editing decisions.

Why Manage Audio Latency?

Audio latency can ruin the listening experience as it causes a perceived delay or “lag” between when audio is played and when it is heard. This poor synchronization or “sync” between playing audio and hearing it impacts applications like music production, video editing, games, and communications.

Musicians and audio engineers are especially bothered by latency when recording and monitoring their performances. Even small delays make it harder to play in time and tune instruments. For remote band practices or online collaboration, latency degrades critical timing between players.

Latency disrupts the natural pace of conversation for applications like video conferencing, voice chat, remote presentations, podcast interviews, and phone calls. Participants talk over one another if they don’t compensate for the lag by briefly pausing. This unnatural pace strains communication and impacts the quality of remote interactions.

Checking Your Audio Interface

Your audio interface settings significantly impact the amount of latency you experience. The buffer size determines how much latency is introduced – a lower buffer size decreases latency but also increases the change of audio glitches and dropouts. Choosing the right audio driver (ASIO on Windows, Core Audio on Mac) also reduces latency compared to default drivers.

According to this article, some recommendations for low latency audio interface settings are:

• Install the recommended driver for your interface (usually ASIO or Core Audio)
• Use a smaller buffer size when recording (64-128 samples)
• Increase the buffer size when mixing (256-512 samples)
• Match the sample rate between your interface and DAW (44.1kHz or 48kHz)

With the right settings, you can achieve roundtrip latency under 10ms. However, you need capable hardware and optimized software to maintain glitch-free performance at very low latencies.

Using a DAW

A DAW, or digital audio workstation, is a software platform used to record, edit, and produce audio materials such as podcasts, songs, and audiobooks. DAWs have a number of settings that can influence how much latency your system is experiencing:

• DAW buffer settings impact latency – The audio buffer is like a short term memory for incoming audio data. A lower buffer size reduces latency, but places higher demands on your CPU. Try adjusting the buffer size between 128 and 512 samples and see what works best without introduced performance issues.
• Plug-in delay compensation – DAWs can automatically adjust for any extra processing delay caused by plug-ins and virtual instruments. Some older plug-ins may not report their latency, requiring further manual adjustment in the DAW.
• Driver error correction modes – Some audio interfaces and their corresponding drivers offer settings like ASIO Guard 2 or Error Correction mode. Turning these on stabilizes performance but increases latency slightly.

Balancing these DAW settings requires some trial and error, but can yield noticeable latency improvements. See the PreSonus guide on understanding and managing latency for additional tuning advice.

Apps to Adjust Latency

There are several apps available for Android and iOS devices that can help adjust audio latency. These apps work by analyzing audio signals and allowing users to apply an offset to align the timing between audio and video sources.

One example for Android is the AV Sync app. This app generates a synchronized flash and beep test signal. By analyzing the timing difference between the flash and beep, it calculates the required offset and can adjust system latency accordingly. Pros of this app include the simple and automated test and adjustment process. Cons are that it requires specialized test signals and only works for audio coming from or through the device itself.

For iOS, the Stereo Lag Time app takes a similar approach. Pros of this app are that it can measure latency from external sources using the device’s microphone. Cons are that it lacks the ability to make system-wide audio adjustments, so is limited to latency within the app itself.

While useful for Aligning audio sources from mobile devices, these apps have limited ability to adjust the end-to-end latency common in production environments with external mixers, audio interfaces, and monitoring systems. They also require specific test signals and access to both ends of the audio chain.

Audio Driver Settings

Adjusting the settings of your audio drivers can help reduce audio latency in many cases. First, make sure you have the latest version of your drivers by checking the manufacturer’s website or your computer model’s support page. For example, if you have a Focusrite audio interface, check their website for the latest Windows drivers.

Next, choose audio drivers designed for low latency audio performance, like ASIO drivers. These driver protocols communicate more directly with the audio hardware, bypassing some of the latency-inducing layers in Windows. Once installed, open the driver’s control panel app to access latency-related settings.

Try settings like increasing the buffer size to reduce dropouts at the cost of more latency. Lower sample rates like 44.1 kHz also tend to have less latency than higher rates like 96 kHz. Enable any “low latency mode” options in the control panel as well. Experiment with different buffer sizes and sample rates to find the best balance for your system and audio needs.

Computer Optimization

Key factors that impact audio latency on your computer include the CPU speed, the amount of RAM, hard drive speed, and operating system settings. A faster CPU will directly improve latency due to less time spent processing audio data (1). This is why upgrading your CPU is often the most impactful change you can make. Increasing RAM lets CPU and DAW resources run more smoothly and access more audio data at once, also helping reduce latency. Finally, having an SSD or fast hard drive minimizes disk access times when streaming and saving audio.

There are also several tweaks you can make in your operating system settings to optimize for lower audio latency:

• Switch to a high-performance power plan in Windows, to prevent CPU throttling (2)
• Disable unnecessary visual effects in Windows
• Give audio processing threads higher priority (3)

You may also consider overclocking your CPU if additional performance is still needed, but make sure to stay within comfortable temperature and voltage ranges.

Using Dedicated Hardware

Dedicated external hardware can help reduce audio latency significantly.

Popular options include external audio interfaces like the Focusrite Scarlett 2i2 3rd Gen, which features:

• Optimized drivers for lower latency
• Quality analog to digital conversion
• Direct hardware monitoring

Amplifiers with built-in USB connections and onboard DSP like the Presonus StudioLive mixers are another option. These mixers combine both digital mixing software and analog hardware, offering the best of both worlds.

Hybrid solutions that merge analog technology with digital conversion and control provide lower latency monitoring while recording into a DAW. Brands like Universal Audio (Thunderbolt interfaces), Avid (HDX), and Waves SoundGrid provide these hybrid workflows.

Latency Troubleshooting

Diagnosing audio latency issues can be challenging, but following a step-by-step troubleshooting guide can help isolate the source. Here are some tips:

Test your audio interface

First, rule out problems with your audio interface by connecting headphones directly to the interface outputs. If latency persists, the issue lies in your computer’s audio configuration or software (Audio Modeling).

Check buffer size settings

In your DAW or audio settings, try adjusting the buffer size – lower values mean less latency but place more demand on your CPU. Finding the right balance for your system can help minimize delay (Practical Music Production).

Test different drivers

If changing the buffer doesn’t help, try alternate ASIO/audio drivers for your interface and compare latency. Device drivers and computer hardware can impact delay.

Optimize background tasks

Close other apps and disable unnecessary background services in your operating system. These can drain CPU resources needed for real-time audio. Check for updated drivers/firmwares too.

Living with Latency

Even after trying all the optimization methods, some amount of audio latency may still persist (How to deal with audio latency). The most common causes of irremediable latency are old or underpowered computers, low-quality audio interfaces, certain plugins, wireless connections, and inherent delays in analog to digital conversion (What Audio Latency Is And How To Reduce It). Sometimes, latency may simply be an unavoidable side effect of your workflow and tools. In these situations, it’s important to understand when latency issues can be worked around or accepted.

Acceptable latency depends heavily on context. For tracking and overdubbing, musicians require near instant monitoring to properly perform and stay in time. Even small amounts of latency can throw off a performer. However, during mixing and mastering latency has less impact. Since tracks are already recorded, short delays on a mixer or master bus generally don’t cause significant problems (Dealing With Computer Audio Latency). The most latency-sensitive use case is live sound, where any delay between mics, instruments, and speakers can cause disastrous feedback and timing issues. Under 20ms roundtrip latency is a good target for live sound.

If you’ve optimized your system fully and still have objectionable latency, consider working around it. Use hardware monitoring while tracking, render tracks before mixing, and print instrument parts early when producing. Delay compensation features in DAWs can also help align late signals automatically. For live use, keep cable runs short, use the fewest plugins possible, and budget for optimized audio interfaces and computers. With care, unavoidable latency can be mitigated in most situations.