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Toju/e2e/helpers/webrtc-helpers.ts
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test: Add playwright main usage test
2026-04-11 16:48:26 +02:00

135 lines
4.3 KiB
TypeScript

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