refactor: Clean lint errors and organise files

e2e/helpers/webrtc-helpers.ts

@@ -11,9 +11,15 @@ import { type Page } from '@playwright/test';
 export async function installWebRTCTracking(page: Page): Promise<void> {
   await page.addInitScript(() => {
     const connections: RTCPeerConnection[] = [];
+    const syntheticMediaResources: {
+      audioCtx: AudioContext;
+      source?: AudioScheduledSourceNode;
+      drawIntervalId?: number;
+    }[] = [];
 
     (window as any).__rtcConnections = connections;
     (window as any).__rtcRemoteTracks = [] as { kind: string; id: string; readyState: string }[];
+    (window as any).__rtcSyntheticMediaResources = syntheticMediaResources;
 
     const OriginalRTCPeerConnection = window.RTCPeerConnection;
 
@@ -55,18 +61,37 @@ export async function installWebRTCTracking(page: Page): Promise<void> {
       // Get the original stream (may include video)
       const originalStream = await origGetUserMedia(constraints);
       const audioCtx = new AudioContext();
-      const oscillator = audioCtx.createOscillator();
+      const noiseBuffer = audioCtx.createBuffer(1, audioCtx.sampleRate * 2, audioCtx.sampleRate);
+      const noiseData = noiseBuffer.getChannelData(0);
 
-      oscillator.frequency.value = 440;
+      for (let sampleIndex = 0; sampleIndex < noiseData.length; sampleIndex++) {
+        noiseData[sampleIndex] = (Math.random() * 2 - 1) * 0.18;
+      }
+
+      const source = audioCtx.createBufferSource();
+      const gain = audioCtx.createGain();
+
+      source.buffer = noiseBuffer;
+      source.loop = true;
+      gain.gain.value = 0.12;
 
       const dest = audioCtx.createMediaStreamDestination();
 
-      oscillator.connect(dest);
-      oscillator.start();
+      source.connect(gain);
+      gain.connect(dest);
+      source.start();
+
+      if (audioCtx.state === 'suspended') {
+        try {
+          await audioCtx.resume();
+        } catch {}
+      }
 
       const synthAudioTrack = dest.stream.getAudioTracks()[0];
       const resultStream = new MediaStream();
 
+      syntheticMediaResources.push({ audioCtx, source });
+
       resultStream.addTrack(synthAudioTrack);
 
       // Keep any video tracks from the original stream
@@ -79,6 +104,14 @@ export async function installWebRTCTracking(page: Page): Promise<void> {
         track.stop();
       }
 
+      synthAudioTrack.addEventListener('ended', () => {
+        try {
+          source.stop();
+        } catch {}
+
+        void audioCtx.close().catch(() => {});
+      }, { once: true });
+
       return resultStream;
     };
 
@@ -128,10 +161,32 @@ export async function installWebRTCTracking(page: Page): Promise<void> {
       osc.connect(dest);
       osc.start();
 
+      if (audioCtx.state === 'suspended') {
+        try {
+          await audioCtx.resume();
+        } catch {}
+      }
+
       const audioTrack = dest.stream.getAudioTracks()[0];
       // Combine video + audio into one stream
       const resultStream = new MediaStream([videoTrack, audioTrack]);
 
+      syntheticMediaResources.push({
+        audioCtx,
+        source: osc,
+        drawIntervalId: drawInterval as unknown as number
+      });
+
+      audioTrack.addEventListener('ended', () => {
+        clearInterval(drawInterval);
+
+        try {
+          osc.stop();
+        } catch {}
+
+        void audioCtx.close().catch(() => {});
+      }, { once: true });
+
       // Tag the stream so tests can identify it
       (resultStream as any).__isScreenShare = true;