fix: audio pipeline — 16 kHz AudioContext, 4096-sample buffering, SERVER_READY handshake

Root causes of disconnection and slow transcription:
- AudioWorklet was firing every 128 native samples (~48 kHz), sending
  ~375 tiny WebSocket messages/sec. Server flooded with tiny frames
  during silence → keepalive ping timed out → connection dropped.
- JS resampling 48 kHz → 16 kHz added CPU overhead on every chunk.
- Audio started on ws.onopen before server sent SERVER_READY, so early
  frames were dropped.

Fixes:
- audioWorklet.js: accumulate 4096 samples before posting (256 ms/chunk
  at 16 kHz, ~4 messages/sec), transfer ArrayBuffer zero-copy.
- transcriptionService: AudioContext({ sampleRate: 16000 }) — browser
  handles native resampling, no JS resampling needed. Remove
  resampleTo16kHZ entirely.
- Wait for SERVER_READY message before calling setupAudioProcessing().
- Send 'END_OF_AUDIO' string on stop so server can finalise last segment.

Co-Authored-By: Claude Sonnet 4.6 <[email protected]>

public/audioWorklet.js

@@ -1,12 +1,26 @@
 class AudioProcessor extends AudioWorkletProcessor {
+  constructor() {
+    super();
+    this._buffer = new Float32Array(4096);
+    this._bufferIndex = 0;
+  }
+
   process(inputs) {
     const input = inputs[0];
     if (input.length > 0) {
-      const audioData = input[0];
-      this.port.postMessage(audioData);
+      const samples = input[0];
+      for (let i = 0; i < samples.length; i++) {
+        this._buffer[this._bufferIndex++] = samples[i];
+        if (this._bufferIndex >= 4096) {
+          // Transfer ownership (zero-copy) to main thread
+          this.port.postMessage(this._buffer.buffer, [this._buffer.buffer]);
+          this._buffer = new Float32Array(4096);
+          this._bufferIndex = 0;
+        }
+      }
     }
     return true;
   }
 }
 
-registerProcessor('audio-processor', AudioProcessor); 
+registerProcessor('audio-processor', AudioProcessor);