perf(voice): reduce transmit latency

- FFmpeg: frame_duration 10ms, compression_level 0, nobuffer/low_delay flags, minimal probe
- Frontend: buffer 4096→1024 samples (~170ms→~42ms), faster base64 with chunking
- Remove processor.connect(destination) to avoid audio feedback loop

Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>

services/discord-gateway/src/modules/voice-recording/transmitter.ts

@@ -50,8 +50,16 @@ export class VoiceTransmitter {
       "-ar", "48000",          // Output sample rate: 48kHz
       "-ac", "2",              // Output channels: stereo
       "-application", "lowdelay", // Low delay mode for real-time
-      "-frame_duration", "20", // 20ms frames
+      "-frame_duration", "10", // 10ms frames (smaller = less latency)
       "-packet_loss", "0",     // No packet loss expected
+      "-compression_level", "0", // Fastest compression (lowest CPU latency)
+      "-cutoff", "20000",      // Audio cutoff frequency
+      "-vsync", "0",           // No video sync
+      "-fflags", "nobuffer",   // No buffering
+      "-flags", "low_delay",   // Low delay processing
+      "-strict", "experimental",
+      "-analyzeduration", "0", // No analysis duration
+      "-probesize", "32",      // Minimal probe size
       "pipe:1",                // Write to stdout
     ], {
       stdio: ["pipe", "pipe", "pipe"],

services/frontend/src/shared/hooks/useAudioTransmit.ts

@@ -58,10 +58,10 @@ export function useAudioTransmit(socketRef: {
     const audioContext = new AudioContextCtor({ sampleRate: SAMPLE_RATE });
     audioContextRef.current = audioContext;
     const source = audioContext.createMediaStreamSource(stream);
-    const processor = audioContext.createScriptProcessor(4096, 1, 1);
+    const processor = audioContext.createScriptProcessor(1024, 1, 1);
     processorRef.current = processor;
     source.connect(processor);
-    processor.connect(audioContext.destination);
+    // Don't connect processor to destination (no monitoring feedback)
     processor.onaudioprocess = (event) => {
       if (!socketRef.current || socketRef.current.readyState !== WebSocket.OPEN)
         return;
@@ -70,15 +70,15 @@ export function useAudioTransmit(socketRef: {
       for (let i = 0; i < inputData.length; i++)
         pcmData[i] = Math.max(-1, Math.min(1, inputData[i])) * 32767;
 
-      // Convert to base64
-      const bytes = new Uint8Array(pcmData.buffer);
-      let binary = '';
-      for (let i = 0; i < bytes.length; i++) {
-        binary += String.fromCharCode(bytes[i]);
+      // Fast base64 using Uint8Array + btoa
+      const uint8 = new Uint8Array(pcmData.buffer);
+      let base64 = '';
+      const chunkSize = 8192;
+      for (let i = 0; i < uint8.length; i += chunkSize) {
+        const chunk = uint8.subarray(i, i + chunkSize);
+        base64 += btoa(String.fromCharCode(...chunk));
       }
-      const base64 = btoa(binary);
 
-      // Send as JSON for backend to forward to Redis
       socketRef.current.send(JSON.stringify({
         type: 'voice_transmit',
         buffer: base64