Files
GMW/services/frontend/src/shared/hooks/useAudioTransmit.ts
T

124 lines
4.1 KiB
TypeScript

// ─── Audio transmit hook — captures mic, encodes to PCM, sends via WebSocket ──
import { useCallback, useRef, useState } from "react";
import { getAPIURL } from "../api/client.js";
import { createLogger } from "../lib/logger";
const SAMPLE_RATE = 24000;
const logger = createLogger("useAudioTransmit");
async function sendTransmitCommand(command: string): Promise<void> {
// Send via HTTP API (kept as exported function for backward compatibility)
const resp = await fetch(`${getAPIURL()}/api/voice/command`, {
method: "POST",
headers: { "Content-Type": "application/json" },
body: JSON.stringify({ command }),
});
if (!resp.ok) {
logger.warn("HTTP command response", {
status: resp.status,
statusText: resp.statusText,
});
const text = await resp.text().catch(() => resp.statusText);
logger.warn("HTTP command failed", { error: text });
throw new Error(`HTTP ${resp.status}: ${text}`);
}
}
function sendWsCommand(
socketRef: { readonly current: WebSocket | null },
command: string,
): boolean {
if (socketRef.current?.readyState === WebSocket.OPEN) {
socketRef.current.send(
JSON.stringify({
type: "voice_command",
command,
}),
);
return true;
}
return false;
}
export function useAudioTransmit(socketRef: {
readonly current: WebSocket | null;
}) {
const [isStreaming, setIsStreaming] = useState(false);
const streamRef = useRef<MediaStream | null>(null);
const audioContextRef = useRef<AudioContext | null>(null);
const processorRef = useRef<ScriptProcessorNode | null>(null);
const stop = useCallback(() => {
// 6c: Prefer WebSocket round-trip over HTTP for lower latency
if (!sendWsCommand(socketRef, "voice:transmit:stop")) {
sendTransmitCommand("voice:transmit:stop").catch(() => {});
}
setIsStreaming(false);
if (processorRef.current) {
processorRef.current.disconnect();
processorRef.current = null;
}
if (audioContextRef.current) {
audioContextRef.current.close();
audioContextRef.current = null;
}
if (streamRef.current) {
for (const track of streamRef.current.getTracks()) track.stop();
streamRef.current = null;
}
}, [socketRef]);
const start = useCallback(async () => {
// 6c: Prefer WebSocket round-trip over HTTP for lower latency
if (!sendWsCommand(socketRef, "voice:transmit:start")) {
await sendTransmitCommand("voice:transmit:start");
}
const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
streamRef.current = stream;
setIsStreaming(true);
const AudioContextCtor =
window.AudioContext ||
(window as unknown as { webkitAudioContext: typeof AudioContext })
.webkitAudioContext;
const audioContext = new AudioContextCtor({ sampleRate: SAMPLE_RATE });
audioContextRef.current = audioContext;
const source = audioContext.createMediaStreamSource(stream);
const processor = audioContext.createScriptProcessor(1024, 1, 1);
processorRef.current = processor;
source.connect(processor);
processor.connect(audioContext.destination);
processor.onaudioprocess = (event) => {
if (!socketRef.current || socketRef.current.readyState !== WebSocket.OPEN)
return;
const inputData = event.inputBuffer.getChannelData(0);
const pcmData = new Int16Array(inputData.length);
for (let i = 0; i < inputData.length; i++)
pcmData[i] = Math.max(-1, Math.min(1, inputData[i])) * 32767;
// 6b: Safe loop instead of spread operator to avoid call-stack overflow
const bytes = new Uint8Array(pcmData.buffer);
let str = '';
for (let i = 0; i < bytes.length; i++) {
str += String.fromCharCode(bytes[i]);
}
const base64 = btoa(str);
socketRef.current.send(
JSON.stringify({
type: "voice_transmit",
buffer: base64,
}),
);
};
}, [socketRef]);
const toggle = useCallback(async () => {
if (isStreaming) stop();
else await start();
}, [isStreaming, start, stop]);
return { isStreaming, toggle, stop, start };
}