Low Latency Voice Options See all the Low latency voice options for your most demanding apps.

As machine learning engineers, especially when working with audio, we’re often juggling the demands of high performance and real-time processing. Whether you’re building a voice assistant, designing audio plugins for a DAW (Digital Audio Workstation), or developing real-time communication software, low latency is crucial.

Every millisecond matters when processing or transmitting audio, and even minor delays can result in noticeable glitches, poor audio quality, or unwanted reverb in voice playback.

In this post, I’ll walk you through key concepts and tools to optimize low-latency audio processing on platforms like Windows, Android, Linux, and macOS. I’ll also dive into use cases, technical considerations like buffer size, sample rates, API options, and give you pointers to open-source tools, github repos, and industry-standard frameworks.

Why Does Latency Matter?

Latency is essentially the delay between an input and its corresponding output. In the context of audio, it refers to the time it takes for an audio signal to be processed and played back. For real-time applications like VOIP, gaming audio, or DSP (digital signal processing) systems, low latency is paramount for a good user experience.

Even a small delay of 10-20 milliseconds can degrade the perceived quality, causing misalignment between sound and visual feedback or making it hard to maintain a conversation.

Roundtrip latency is the total time it takes for audio to go from an input (like a microphone), through processing, and back out through the speakers or headphones. Ideally, you’d aim for near zero-latency, but in practice, several factors, such as CPU load, sample rate, and buffer size, introduce delays.

Factors That Impact Latency

1. Buffer Size

The buffer size is one of the biggest factors in determining latency. Smaller buffer sizes reduce latency but increase the risk of glitches and require higher CPU resources. Larger buffers can lead to noticeable delays but provide stability, especially for complex audio tasks like decoding or real-time audio processing.

2. Sample Rate

The sample rate refers to how many samples of audio are processed per second. Higher rates can improve audio quality but also demand more processing power and increase potential latency. Lower sample rates decrease the load on your CPU and GPU but might sacrifice clarity. A balance between quality and performance is necessary.

3. Audio Interface and Drivers

Your audio interface and drivers play a big role in achieving low latency. On Windows, using an ASIO driver (Audio Stream Input/Output) is preferred because it bypasses much of the OS-level audio processing, resulting in lower latencies than standard drivers. Mac and Linux users benefit from Core Audio and ALSA, respectively, which are optimized for professional audio tasks. For developers targeting Android, working directly with the native audio output APIs can help minimize delays.

4. Plugins and DSP

Audio plugins used in DAWs or other audio environments also introduce latency, especially if they rely on complex DSP algorithms like reverb or noise cancellation. Minimizing plugin latency requires optimizing the internal processing and reducing buffer sizes while maintaining high quality.

5. Platform-Specific Considerations

Latency varies depending on the platform:

1. Windows: As mentioned, ASIO drivers offer the lowest latency options, with latencies often as low as a few milliseconds. However, low-latency audio on Windows can be complex, especially if the API layer isn’t optimized.
2. Linux: The combination of ALSA and JACK Audio Connection Kit makes low-latency audio viable for professionals using open-source solutions. Linux’s flexibility allows for extensive tuning, but setup can be tricky.
3. Mac: Core Audio offers one of the best out-of-the-box low-latency experiences. Latencies of under 10 milliseconds are common with the right audio device.
4. Android: Optimizing for low latency on Android is more difficult due to hardware variability. However, Android’s OpenSL ES API or AAudio API (for newer versions) offer developers low-latency paths to the audio output.
5. Bluetooth: Wireless audio solutions, particularly over Bluetooth, typically suffer from high latency due to encoding and decoding delays. Using aptX or LDAC codecs can help, but these still introduce significant latency compared to wired solutions.

6. Network Conditions

For real-time communication applications (think VOIP or voice chat), bandwidth limitations and network variability play a significant role in latency. High latency networks can lead to delayed audio or even dropouts. Proper audio processing can mitigate some of these effects, but only up to a point.

Key Tools for Achieving Low Latency

1. APIs

If you’re looking to build an application that requires extremely low latency without compromising on the quality of the voice, then PlayHT has quickly distanced itself from the pack and stands out. Check out the PlayHT text to speech API, you can sign up for free and test it with your most demanding app.

2. ASIO4ALL (Windows)

If you’re stuck with a consumer-grade audio card, ASIO4ALL is an excellent driver to achieve lower latency. It doesn’t replace high-end interfaces but can reduce latency to tolerable levels for many applications.

3. JACK Audio Connection Kit (Linux, macOS, Windows)

For professional-grade low latency audio on Linux, JACK is the gold standard. It allows for real-time audio routing and is ideal for use in high-performance setups like live audio processing or DSP work.

4. AAudio (Android)

For developers on Android, AAudio is a modern API built to handle real-time audio with minimal latency. While Android isn’t the easiest platform for low-latency work, using AAudio in combination with proper sample rate and buffer size tuning can result in more responsive audio playback.

5. PortAudio (Cross-platform)

PortAudio is an open-source API that provides a uniform interface for low-latency audio across Windows, macOS, and Linux. It’s great for machine learning engineers who want to create cross-platform audio solutions while minimizing delay.

6. GitHub Repositories for Real-Time Audio Processing

Explore repositories like the JUCE Framework on GitHub, which is a popular tool for building audio applications with low latency. JUCE is used for everything from plugin development to standalone audio applications and is known for its flexibility and ease of use.

Use Cases for Low Latency Audio

1. Voice Assistants: Real-time voice processing is crucial for apps like Alexa or Siri, where delay directly impacts the user experience. Fast response times ensure a fluid interaction with the assistant.
2. DAW Plugins: For real-time effects like reverb, zero-latency operation is necessary, especially during live performances or real-time monitoring.
3. VOIP Applications: Low-latency ensures smooth communication with minimal delay between parties, critical for platforms like Skype or Zoom.
4. Machine Learning Audio Models: Models that perform real-time inference for audio tasks (such as voice recognition, noise cancellation, or even music generation) require sub-millisecond processing delays to be effective.

Minimizing Latency: Practical Tips

1. Optimize Buffer Size: Reducing your buffer size can dramatically lower latency, but keep an eye on CPU usage to avoid glitches.
2. Tune Sample Rates: Match your sample rate to the intended audio device. Often, setting the same sample rate across input and output paths can avoid unnecessary conversion delays.
3. Use Dedicated Audio Hardware: While consumer audio interfaces can sometimes suffice, using a professional-grade audio interface can dramatically reduce latency, especially for live audio tasks.
4. Monitor CPU/GPU Utilization: Too much load on the CPU or GPU can bottleneck your audio processing. Offload tasks or optimize your code to make room for real-time tasks.

Latency can be the make-or-break factor in real-time audio systems. Whether you’re working on Android, Windows, Linux, or macOS, optimizing buffer size, sample rate, and utilizing platform-specific tools like ASIO, Core Audio, or AAudio are all crucial to hitting the lowest latency possible.

And don’t forget about the importance of hardware – a dedicated audio interface and well-optimized drivers make all the difference.

As you dive into building or optimizing your next real-time audio system, don’t forget to check out the wealth of open-source tools available on GitHub, and pay close attention to platform-specific docs. Achieving the lowest latency often comes down to knowing how to properly tune your environment.