test: Add playwright main usage test
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134
e2e/helpers/webrtc-helpers.ts
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134
e2e/helpers/webrtc-helpers.ts
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/* eslint-disable @typescript-eslint/no-explicit-any */
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import { type Page } from '@playwright/test';
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/**
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* Install RTCPeerConnection monkey-patch on a page BEFORE navigating.
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* Tracks all created peer connections and their remote tracks so tests
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* can inspect WebRTC state via `page.evaluate()`.
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*
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* Call immediately after page creation, before any `goto()`.
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*/
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export async function installWebRTCTracking(page: Page): Promise<void> {
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await page.addInitScript(() => {
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const connections: RTCPeerConnection[] = [];
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(window as any).__rtcConnections = connections;
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(window as any).__rtcRemoteTracks = [] as { kind: string; id: string; readyState: string }[];
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const OriginalRTCPeerConnection = window.RTCPeerConnection;
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(window as any).RTCPeerConnection = function(this: RTCPeerConnection, ...args: any[]) {
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const pc: RTCPeerConnection = new OriginalRTCPeerConnection(...args);
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connections.push(pc);
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pc.addEventListener('connectionstatechange', () => {
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(window as any).__lastRtcState = pc.connectionState;
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});
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pc.addEventListener('track', (event: RTCTrackEvent) => {
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(window as any).__rtcRemoteTracks.push({
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kind: event.track.kind,
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id: event.track.id,
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readyState: event.track.readyState
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});
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});
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return pc;
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} as any;
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(window as any).RTCPeerConnection.prototype = OriginalRTCPeerConnection.prototype;
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Object.setPrototypeOf((window as any).RTCPeerConnection, OriginalRTCPeerConnection);
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});
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}
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/**
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* Wait until at least one RTCPeerConnection reaches the 'connected' state.
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*/
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export async function waitForPeerConnected(page: Page, timeout = 30_000): Promise<void> {
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await page.waitForFunction(
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() => (window as any).__rtcConnections?.some(
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(pc: RTCPeerConnection) => pc.connectionState === 'connected'
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) ?? false,
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{ timeout }
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);
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}
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/**
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* Check that a peer connection is still in 'connected' state (not failed/disconnected).
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*/
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export async function isPeerStillConnected(page: Page): Promise<boolean> {
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return page.evaluate(
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() => (window as any).__rtcConnections?.some(
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(pc: RTCPeerConnection) => pc.connectionState === 'connected'
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) ?? false
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);
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}
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/**
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* Get outbound and inbound audio RTP stats from the first peer connection.
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*/
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export async function getAudioStats(page: Page): Promise<{
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outbound: { bytesSent: number; packetsSent: number } | null;
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inbound: { bytesReceived: number; packetsReceived: number } | null;
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}> {
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return page.evaluate(async () => {
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const connections = (window as any).__rtcConnections as RTCPeerConnection[] | undefined;
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if (!connections?.length)
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return { outbound: null, inbound: null };
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let outbound: { bytesSent: number; packetsSent: number } | null = null;
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let inbound: { bytesReceived: number; packetsReceived: number } | null = null;
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for (const pc of connections) {
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if (pc.connectionState !== 'connected')
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continue;
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const stats = await pc.getStats();
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stats.forEach((report: any) => {
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const reportMediaType = report.kind ?? report.mediaType;
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if (report.type === 'outbound-rtp' && reportMediaType === 'audio' && !outbound) {
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outbound = {
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bytesSent: report.bytesSent ?? 0,
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packetsSent: report.packetsSent ?? 0
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};
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}
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if (report.type === 'inbound-rtp' && reportMediaType === 'audio' && !inbound) {
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inbound = {
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bytesReceived: report.bytesReceived ?? 0,
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packetsReceived: report.packetsReceived ?? 0
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};
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}
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});
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if (outbound && inbound)
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break;
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}
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return { outbound, inbound };
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});
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}
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/**
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* Snapshot audio stats, wait `durationMs`, snapshot again, and return the delta.
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* Useful for verifying audio is actively flowing (bytes increasing).
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*/
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export async function getAudioStatsDelta(page: Page, durationMs = 3_000): Promise<{
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outboundBytesDelta: number;
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inboundBytesDelta: number;
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}> {
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const before = await getAudioStats(page);
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await page.waitForTimeout(durationMs);
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const after = await getAudioStats(page);
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return {
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outboundBytesDelta: (after.outbound?.bytesSent ?? 0) - (before.outbound?.bytesSent ?? 0),
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inboundBytesDelta: (after.inbound?.bytesReceived ?? 0) - (before.inbound?.bytesReceived ?? 0)
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};
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}
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