rich-dusk-37560
05/11/2026, 11:39 AMtall-airline-16756
05/11/2026, 11:39 AMtall-airline-16756
05/11/2026, 11:40 AMsetMicrophoneEnabled(false) (or room.localParticipant.setMicrophoneEnabled(false)) whenever the user is not expected to speak. This stops publishing audio to the room and prevents input billing during that time.
• Use client-side VAD to toggle the mic automatically
◦ Run a local VAD (e.g., WebAudio-based or a lightweight VAD library) in the browser. Enable the mic only when speech is detected and disable it after speech ends or after a short timeout. This avoids sending room tone while preserving hands-free UX.
• Gate before sending to Gemini (server-side VAD or audio gate)
◦ If you need to keep the track published to LiveKit for other reasons, add a server-side or intermediary VAD that forwards audio to Gemini only when voice activity is detected. Note: your current Gemini Live path does not do this by default.
• Reduce unnecessary publish during agent output
◦ Disable user mic while the agent is speaking or during known quiet windows (e.g., during Gemini TTS playback). Re-enable for user turns.
Trade-offs to consider
• UX vs cost: aggressive muting/VAD reduces costs but may add latency or miss short user utterances. Tune VAD sensitivity and timeouts.
• Noise cancellation vs gating: browser and LiveKit noise cancellation reduce what the model hears but do not guarantee zero billing if audio is published while mic is enabled.
• Implementation complexity: client-side VAD + mic toggling is simpler to implement and avoids changes to the Gemini path; server-side gating requires routing audio through additional logic.
Suggested next step
• If you want, I can:
◦ Suggest a lightweight client-side VAD approach and code pattern to toggle setMicrophoneEnabled, or
◦ Outline how to integrate a server-side gate if you prefer that architecture.
Which option would you like help implementing?
Sources: Gemini Live API plugin | LiveKit Documentation | https://livekit.com/pricing.md| | Turns overview | LiveKit Documentation | Understanding "silence detected on local audio track" warning | Text and transcriptions | LiveKit Documentation, with interactive elementsrich-dusk-37560
05/22/2026, 6:59 AMsetMicrophoneEnabled, or
◦tall-airline-16756
05/22/2026, 6:59 AMsetMicrophoneEnabled. This will reduce sending silence to Gemini Live while keeping UX responsive.
Summary
• Use a lightweight VAD running in the browser (WebAssembly or WebAudio + small model) to detect speech locally.
• When VAD reports speech, enable the mic (setMicrophoneEnabled(true)); when VAD reports silence for a short timeout, disable the mic (setMicrophoneEnabled(false)).
• Tune sensitivity and silence timeout to balance missed starts vs. cost.
Recommended options for browser VAD
• Silero VAD compiled to WebAssembly (recommended for accuracy).
• A tiny WebAudio/TensorFlow.js VAD (lower accuracy but simpler).
(Choose one depending on complexity you want; below uses a generic WebAudio + WASM VAD pattern.)
High-level pattern
1. Capture a short raw audio stream locally (AudioWorklet or ScriptProcessor) and feed frames into the VAD.
2. Maintain a speech state machine:
◦ On speech start -> call room.localParticipant.setMicrophoneEnabled(true) (if not already).
◦ On prolonged silence (e.g., 500–800 ms) -> call room.localParticipant.setMicrophoneEnabled(false).
1. Optional: while mic is disabled you can still keep a muted track for playback sync or UI; ensure you’re not publishing audio when disabled.
Example TypeScript/React pattern (concise) 1) Install/prepare VAD (pseudo-install for WASM or TF.js)
• If using a WASM VAD, include the compiled WASM and a small JS wrapper.
2) Hook to microphone and VAD in a React hook:
import { useEffect, useRef } from "react";
// Assume `room` is your LiveKit Room instance available in component scope
// and `createVAD` is a small wrapper that returns {processFrame: Float32Array=>bool, dispose}
export function useClientVAD(room: any, enabled = true) {
const vadRef = useRef<any>(null);
const audioCtxRef = useRef<AudioContext | null>(null);
const sourceRef = useRef<MediaStreamAudioSourceNode | null>(null);
const workletNodeRef = useRef<AudioWorkletNode | null>(null);
const silenceTimerRef = useRef<number | null>(null);
const speakingRef = useRef(false);
useEffect(() => {
if (!enabled) return;
let mounted = true;
async function start() {
audioCtxRef.current = new AudioContext({ sampleRate: 16000 });
// request mic but do NOT publish until we explicitly enable via LiveKit SDK
const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
sourceRef.current = audioCtxRef.current.createMediaStreamSource(stream);
// load VAD WASM / model
vadRef.current = await createVAD(); // implement or import your VAD wrapper
// simple ScriptProcessor for small demo (or use AudioWorklet)
const bufferSize = 2048;
const processor = audioCtxRef.current.createScriptProcessor(bufferSize, 1, 1);
processor.onaudioprocess = (e) => {
const input = e.inputBuffer.getChannelData(0);
const isSpeech = vadRef.current.processFrame(input); // returns bool
if (isSpeech) {
// cancel pending silence timeout
if (silenceTimerRef.current) {
window.clearTimeout(silenceTimerRef.current);
silenceTimerRef.current = null;
}
if (!speakingRef.current) {
speakingRef.current = true;
// enable publishing to LiveKit so audio is sent to Gemini
room.localParticipant.setMicrophoneEnabled(true);
}
} else {
// start or reset silence timeout (e.g., 700ms)
if (!silenceTimerRef.current) {
silenceTimerRef.current = window.setTimeout(() => {
silenceTimerRef.current = null;
if (speakingRef.current) {
speakingRef.current = false;
room.localParticipant.setMicrophoneEnabled(false);
}
}, 700);
}
}
};
sourceRef.current.connect(processor);
processor.connect(audioCtxRef.current.destination);
workletNodeRef.current = processor;
}
start();
return () => {
mounted = false;
// cleanup
if (silenceTimerRef.current) window.clearTimeout(silenceTimerRef.current);
if (workletNodeRef.current) {
workletNodeRef.current.disconnect();
workletNodeRef.current = null;
}
if (sourceRef.current) {
sourceRef.current.disconnect();
sourceRef.current = null;
}
if (audioCtxRef.current) {
audioCtxRef.current.close();
audioCtxRef.current = null;
}
if (vadRef.current) vadRef.current.dispose?.();
// ensure mic publishing disabled on unmount
room.localParticipant.setMicrophoneEnabled(false);
};
}, [room, enabled]);
}
Notes & tuning
• Buffer size & sample rate: convert frames to the rate your VAD expects (common choice 16 kHz). Use an offline converter if needed.
• Silence timeout: 500–800 ms is a common starting point. Lower values risk chopping words; higher values increase billing.
• Warm-up: start VAD in a muted state, enable only when speech is detected to avoid initial billing.
• Edge cases: If you need to detect very short interjections (“uh-huh”), lower timeout but be aware of false positives.
• UX: show a mic LED/state so users know when mic is enabled. Allow manual override (user mute button) that forces mic off regardless of VAD.
If you want, I can:
• Provide a small concrete VAD implementation suggestion (Silero WebAssembly wrapper or a TF.js-compatible tiny model) and example createVAD() implementation, or
• Tailor buffer sizes and timeouts to your existing audio chain in Chicky.
Which would you prefer?
Sources: Silero VAD plugin | LiveKit Documentation, with interactive elements