refactor(voice): optimize recording pipeline and improve performance

Refactor the voice recording and playback systems to improve efficiency, reduce latency, and enhance Docker build performance.

- **Infrastructure**: Optimize Dockerfiles using build mounts for pnpm cache and reorder layers for better caching of dependencies and build tools.
- **Backend/Gateway**:
  - Refactor `broadcast` module to use a generic event-based system instead of hardcoded functions.
  - Simplify voice recording logic by merging metadata and segment management into a unified `segment.ts`.
  - Optimize audio downsampling in `streamSetup.ts` using `Int16Array` views for better performance.
  - Implement `withFallback` utility for more robust Redis/Database command execution.
- **Frontend**:
  - Optimize audio playback visualization using pre-computed level shapes and efficient RMS calculation.
  - Reduce latency in voice commands by prioritizing WebSocket communication over HTTP.
  - Improve base64 encoding efficiency in audio transmission.
- **General**:
  - Add default value for `ADMIN_PASSWORD` in shared config.
  - Fix Docker healthcheck to use `127.0.0.1` instead of `localhost`.
This commit is contained in:
MythEclipse
2026-06-09 22:04:05 +07:00
parent 823b342c2c
commit f04b0f0b42
20 changed files with 447 additions and 633 deletions
@@ -1,13 +1,13 @@
// ─── Audio transmit hook — captures mic, encodes to PCM, sends via WebSocket ──
import { useCallback, useRef, useState } from "react";
import { getAPIURL } from "../api/client";
import { createChildLogger } from "../logger";
import { getAPIURL } from "../api/client.js";
import { createChildLogger } from "../logger.js";
const SAMPLE_RATE = 24000;
const logger = createChildLogger("useAudioTransmit");
async function sendTransmitCommand(command: string): Promise<void> {
// Send via HTTP API
// 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" },
@@ -24,6 +24,22 @@ async function sendTransmitCommand(command: string): Promise<void> {
}
}
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;
}) {
@@ -33,7 +49,10 @@ export function useAudioTransmit(socketRef: {
const processorRef = useRef<ScriptProcessorNode | null>(null);
const stop = useCallback(() => {
sendTransmitCommand("voice:transmit:stop").catch(() => {});
// 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) {
@@ -48,10 +67,13 @@ export function useAudioTransmit(socketRef: {
for (const track of streamRef.current.getTracks()) track.stop();
streamRef.current = null;
}
}, []);
}, [socketRef]);
const start = useCallback(async () => {
await sendTransmitCommand("voice:transmit:start");
// 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;
@@ -75,13 +97,10 @@ export function useAudioTransmit(socketRef: {
for (let i = 0; i < inputData.length; i++)
pcmData[i] = Math.max(-1, Math.min(1, inputData[i])) * 32767;
// Base64 encode
// 6b: Replace string-concatenation loop with single call
// 1024 samples → 2048 bytes — well within call-stack limits
const bytes = new Uint8Array(pcmData.buffer);
let binary = "";
for (let i = 0; i < bytes.length; i++) {
binary += String.fromCharCode(bytes[i]);
}
const base64 = btoa(binary);
const base64 = btoa(String.fromCharCode(...bytes));
socketRef.current.send(
JSON.stringify({