Wait Your Turn! Conversation by Voice Versus Text Text interactions require taking turns, but voices may interrupt or overlap. Here’s how AI is evolving for voice interactions.

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Diagram of an RQ-Transformer speech system with Helium and Depth Transformers for audio processing.
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Dear friends,

Continuing our discussion on the Voice Stack, I’d like to explore an area that today’s voice-based systems mostly struggle with: Voice Activity Detection (VAD) and the turn-taking paradigm of communication.

When communicating with a text-based chatbot, the turns are clear: You write something, then the bot does, then you do, and so on. The success of text-based chatbots with clear turn-taking has influenced the design of voice-based bots, most of which also use the turn-taking paradigm.

A key part of building such a system is a VAD component to detect when the user is talking. This allows our software to take the parts of the audio stream in which the user is saying something and pass that to the model for the user’s turn. It also supports interruption in a limited way, whereby if a user insistently interrupts the AI system while it is talking, eventually the VAD system will realize the user is talking, shut off the AI’s output, and let the user take a turn. This works reasonably well in quiet environments.

However, VAD systems today struggle with noisy environments, particularly when the background noise is from other human speech. For example, if you are in a noisy cafe speaking with a voice chatbot, VAD — which is usually trained to detect human speech — tends to be inaccurate at figuring out when you, or someone else, is talking. (In comparison, it works much better if you are in a noisy vehicle, since the background noise is more clearly not human speech.) It might think you are interrupting when it was merely someone in the background speaking, or fail to recognize that you’ve stopped talking. This is why today’s speech applications often struggle in noisy environments.

Intriguingly, last year, Kyutai Labs published Moshi, a model (GitHub) that had many technical innovations. An important one was enabling persistent bi-direction audio streams from the user to Moshi and from Moshi to the user.

If you and I were speaking in person or on the phone, we would constantly be streaming audio to each other (through the air or the phone system), and we’d use social cues to know when to listen and how to politely interrupt if one of us felt the need. Thus, the streams would not need to explicitly model turn-taking. Moshi works like this. It’s listening all the time, and it’s up to the model to decide when to stay silent and when to talk. This means an explicit VAD step is no longer necessary. (Moshi also included other innovations, such as an “inner monologue” that simultaneously generates text alongside the audio to improve the quality of responses as well as audio encoding.)

Just as the architecture of text-only transformers has gone through many evolutions (such as encoder-decoder models, decoder-only models, and reasoning models that generate a lot of “reasoning tokens” before the final output), voice models are going through a lot of architecture explorations. Given the importance of foundation models with voice-in and voice-out capabilities, many large companies right now are investing in developing better voice models. I’m confident we’ll see many more good voice models released this year.

It feels like the space of potential innovation for voice remains large. Hard technical problems, like the one of latency that I described last week and VAD errors, remain to be solved. As solutions get better, voice-to-voice will continue to be a promising category to build applications in.

Keep building!

Andrew 

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