test: Add playwright main usage test

e2e/helpers/webrtc-helpers.ts

@@ -0,0 +1,717 @@
+/* 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);
+
+    // Patch getUserMedia to use an AudioContext oscillator for audio
+    // instead of the hardware capture device.  Chromium's fake audio
+    // device intermittently fails to produce frames after renegotiation.
+    const origGetUserMedia = navigator.mediaDevices.getUserMedia.bind(navigator.mediaDevices);
+
+    navigator.mediaDevices.getUserMedia = async (constraints?: MediaStreamConstraints) => {
+      const wantsAudio = !!constraints?.audio;
+
+      if (!wantsAudio) {
+        return origGetUserMedia(constraints);
+      }
+
+      // Get the original stream (may include video)
+      const originalStream = await origGetUserMedia(constraints);
+      const audioCtx = new AudioContext();
+      const oscillator = audioCtx.createOscillator();
+
+      oscillator.frequency.value = 440;
+
+      const dest = audioCtx.createMediaStreamDestination();
+
+      oscillator.connect(dest);
+      oscillator.start();
+
+      const synthAudioTrack = dest.stream.getAudioTracks()[0];
+      const resultStream = new MediaStream();
+
+      resultStream.addTrack(synthAudioTrack);
+
+      // Keep any video tracks from the original stream
+      for (const videoTrack of originalStream.getVideoTracks()) {
+        resultStream.addTrack(videoTrack);
+      }
+
+      // Stop original audio tracks since we're not using them
+      for (const track of originalStream.getAudioTracks()) {
+        track.stop();
+      }
+
+      return resultStream;
+    };
+
+    // Patch getDisplayMedia to return a synthetic screen share stream
+    // (canvas-based video + 880Hz oscillator audio) so the browser
+    // picker dialog is never shown.
+    navigator.mediaDevices.getDisplayMedia = async (_constraints?: DisplayMediaStreamOptions) => {
+      const canvas = document.createElement('canvas');
+
+      canvas.width = 640;
+      canvas.height = 480;
+
+      const ctx = canvas.getContext('2d');
+
+      if (!ctx) {
+        throw new Error('Canvas 2D context unavailable');
+      }
+
+      let frameCount = 0;
+
+      // Draw animated frames so video stats show increasing bytes
+      const drawFrame = () => {
+        frameCount++;
+        ctx.fillStyle = `hsl(${frameCount % 360}, 70%, 50%)`;
+        ctx.fillRect(0, 0, canvas.width, canvas.height);
+        ctx.fillStyle = '#fff';
+        ctx.font = '24px monospace';
+        ctx.fillText(`Screen Share Frame ${frameCount}`, 40, 60);
+      };
+
+      drawFrame();
+      const drawInterval = setInterval(drawFrame, 100);
+      const videoStream = canvas.captureStream(10); // 10 fps
+      const videoTrack = videoStream.getVideoTracks()[0];
+
+      // Stop drawing when the track ends
+      videoTrack.addEventListener('ended', () => clearInterval(drawInterval));
+
+      // Create 880Hz oscillator for screen share audio (distinct from 440Hz voice)
+      const audioCtx = new AudioContext();
+      const osc = audioCtx.createOscillator();
+
+      osc.frequency.value = 880;
+
+      const dest = audioCtx.createMediaStreamDestination();
+
+      osc.connect(dest);
+      osc.start();
+
+      const audioTrack = dest.stream.getAudioTracks()[0];
+      // Combine video + audio into one stream
+      const resultStream = new MediaStream([videoTrack, audioTrack]);
+
+      // Tag the stream so tests can identify it
+      (resultStream as any).__isScreenShare = true;
+
+      return resultStream;
+    };
+  });
+}
+
+/**
+ * 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 aggregated across all peer
+ * connections.  Uses a per-connection high water mark stored on `window` so
+ * that connections that close mid-measurement still contribute their last
+ * known counters, preventing the aggregate from going backwards.
+ */
+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 };
+
+    interface HWMEntry {
+      outBytesSent: number;
+      outPacketsSent: number;
+      inBytesReceived: number;
+      inPacketsReceived: number;
+      hasOutbound: boolean;
+      hasInbound: boolean;
+    };
+
+    const hwm: Record<number, HWMEntry> = (window as any).__rtcStatsHWM =
+      ((window as any).__rtcStatsHWM as Record<number, HWMEntry> | undefined) ?? {};
+
+    for (let idx = 0; idx < connections.length; idx++) {
+      let stats: RTCStatsReport;
+
+      try {
+        stats = await connections[idx].getStats();
+      } catch {
+        continue; // closed connection - keep its last HWM
+      }
+
+      let obytes = 0;
+      let opackets = 0;
+      let ibytes = 0;
+      let ipackets = 0;
+      let hasOut = false;
+      let hasIn = false;
+
+      stats.forEach((report: any) => {
+        const kind = report.kind ?? report.mediaType;
+
+        if (report.type === 'outbound-rtp' && kind === 'audio') {
+          hasOut = true;
+          obytes += report.bytesSent ?? 0;
+          opackets += report.packetsSent ?? 0;
+        }
+
+        if (report.type === 'inbound-rtp' && kind === 'audio') {
+          hasIn = true;
+          ibytes += report.bytesReceived ?? 0;
+          ipackets += report.packetsReceived ?? 0;
+        }
+      });
+
+      if (hasOut || hasIn) {
+        hwm[idx] = {
+          outBytesSent: obytes,
+          outPacketsSent: opackets,
+          inBytesReceived: ibytes,
+          inPacketsReceived: ipackets,
+          hasOutbound: hasOut,
+          hasInbound: hasIn
+        };
+      }
+    }
+
+    let totalOutBytes = 0;
+    let totalOutPackets = 0;
+    let totalInBytes = 0;
+    let totalInPackets = 0;
+    let anyOutbound = false;
+    let anyInbound = false;
+
+    for (const entry of Object.values(hwm)) {
+      totalOutBytes += entry.outBytesSent;
+      totalOutPackets += entry.outPacketsSent;
+      totalInBytes += entry.inBytesReceived;
+      totalInPackets += entry.inPacketsReceived;
+
+      if (entry.hasOutbound)
+        anyOutbound = true;
+
+      if (entry.hasInbound)
+        anyInbound = true;
+    }
+
+    return {
+      outbound: anyOutbound
+        ? { bytesSent: totalOutBytes, packetsSent: totalOutPackets }
+        : null,
+      inbound: anyInbound
+        ? { bytesReceived: totalInBytes, packetsReceived: totalInPackets }
+        : null
+    };
+  });
+}
+
+/**
+ * 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;
+  outboundPacketsDelta: number;
+  inboundPacketsDelta: 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),
+    outboundPacketsDelta: (after.outbound?.packetsSent ?? 0) - (before.outbound?.packetsSent ?? 0),
+    inboundPacketsDelta: (after.inbound?.packetsReceived ?? 0) - (before.inbound?.packetsReceived ?? 0)
+  };
+}
+
+/**
+ * Wait until at least one connection has both outbound-rtp and inbound-rtp
+ * audio reports.  Call after `waitForPeerConnected` to ensure the audio
+ * pipeline is ready before measuring deltas.
+ */
+export async function waitForAudioStatsPresent(page: Page, timeout = 15_000): Promise<void> {
+  await page.waitForFunction(
+    async () => {
+      const connections = (window as any).__rtcConnections as RTCPeerConnection[] | undefined;
+
+      if (!connections?.length)
+        return false;
+
+      for (const pc of connections) {
+        let stats: RTCStatsReport;
+
+        try {
+          stats = await pc.getStats();
+        } catch {
+          continue;
+        }
+
+        let hasOut = false;
+        let hasIn = false;
+
+        stats.forEach((report: any) => {
+          const kind = report.kind ?? report.mediaType;
+
+          if (report.type === 'outbound-rtp' && kind === 'audio')
+            hasOut = true;
+
+          if (report.type === 'inbound-rtp' && kind === 'audio')
+            hasIn = true;
+        });
+
+        if (hasOut && hasIn)
+          return true;
+      }
+
+      return false;
+    },
+    { timeout }
+  );
+}
+
+interface AudioFlowDelta {
+  outboundBytesDelta: number;
+  inboundBytesDelta: number;
+  outboundPacketsDelta: number;
+  inboundPacketsDelta: number;
+}
+
+function snapshotToDelta(
+  curr: Awaited<ReturnType<typeof getAudioStats>>,
+  prev: Awaited<ReturnType<typeof getAudioStats>>
+): AudioFlowDelta {
+  return {
+    outboundBytesDelta: (curr.outbound?.bytesSent ?? 0) - (prev.outbound?.bytesSent ?? 0),
+    inboundBytesDelta: (curr.inbound?.bytesReceived ?? 0) - (prev.inbound?.bytesReceived ?? 0),
+    outboundPacketsDelta: (curr.outbound?.packetsSent ?? 0) - (prev.outbound?.packetsSent ?? 0),
+    inboundPacketsDelta: (curr.inbound?.packetsReceived ?? 0) - (prev.inbound?.packetsReceived ?? 0)
+  };
+}
+
+function isDeltaFlowing(delta: AudioFlowDelta): boolean {
+  const outFlowing = delta.outboundBytesDelta > 0 || delta.outboundPacketsDelta > 0;
+  const inFlowing = delta.inboundBytesDelta > 0 || delta.inboundPacketsDelta > 0;
+
+  return outFlowing && inFlowing;
+}
+
+/**
+ * Poll until two consecutive HWM-based reads show both outbound and inbound
+ * audio byte counts increasing.  Combines per-connection high-water marks
+ * (which prevent totals from going backwards after connection churn) with
+ * consecutive comparison (which avoids a stale single baseline).
+ */
+export async function waitForAudioFlow(
+  page: Page,
+  timeoutMs = 30_000,
+  pollIntervalMs = 1_000
+): Promise<AudioFlowDelta> {
+  const deadline = Date.now() + timeoutMs;
+
+  let prev = await getAudioStats(page);
+
+  while (Date.now() < deadline) {
+    await page.waitForTimeout(pollIntervalMs);
+    const curr = await getAudioStats(page);
+    const delta = snapshotToDelta(curr, prev);
+
+    if (isDeltaFlowing(delta)) {
+      return delta;
+    }
+
+    prev = curr;
+  }
+
+  // Timeout - return zero deltas so the caller's assertion reports the failure.
+  return {
+    outboundBytesDelta: 0,
+    inboundBytesDelta: 0,
+    outboundPacketsDelta: 0,
+    inboundPacketsDelta: 0
+  };
+}
+
+/**
+ * Get outbound and inbound video RTP stats aggregated across all peer
+ * connections.  Uses the same HWM pattern as {@link getAudioStats}.
+ */
+export async function getVideoStats(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 };
+
+    interface VHWM {
+      outBytesSent: number;
+      outPacketsSent: number;
+      inBytesReceived: number;
+      inPacketsReceived: number;
+      hasOutbound: boolean;
+      hasInbound: boolean;
+    }
+
+    const hwm: Record<number, VHWM> = (window as any).__rtcVideoStatsHWM =
+      ((window as any).__rtcVideoStatsHWM as Record<number, VHWM> | undefined) ?? {};
+
+    for (let idx = 0; idx < connections.length; idx++) {
+      let stats: RTCStatsReport;
+
+      try {
+        stats = await connections[idx].getStats();
+      } catch {
+        continue;
+      }
+
+      let obytes = 0;
+      let opackets = 0;
+      let ibytes = 0;
+      let ipackets = 0;
+      let hasOut = false;
+      let hasIn = false;
+
+      stats.forEach((report: any) => {
+        const kind = report.kind ?? report.mediaType;
+
+        if (report.type === 'outbound-rtp' && kind === 'video') {
+          hasOut = true;
+          obytes += report.bytesSent ?? 0;
+          opackets += report.packetsSent ?? 0;
+        }
+
+        if (report.type === 'inbound-rtp' && kind === 'video') {
+          hasIn = true;
+          ibytes += report.bytesReceived ?? 0;
+          ipackets += report.packetsReceived ?? 0;
+        }
+      });
+
+      if (hasOut || hasIn) {
+        hwm[idx] = {
+          outBytesSent: obytes,
+          outPacketsSent: opackets,
+          inBytesReceived: ibytes,
+          inPacketsReceived: ipackets,
+          hasOutbound: hasOut,
+          hasInbound: hasIn
+        };
+      }
+    }
+
+    let totalOutBytes = 0;
+    let totalOutPackets = 0;
+    let totalInBytes = 0;
+    let totalInPackets = 0;
+    let anyOutbound = false;
+    let anyInbound = false;
+
+    for (const entry of Object.values(hwm)) {
+      totalOutBytes += entry.outBytesSent;
+      totalOutPackets += entry.outPacketsSent;
+      totalInBytes += entry.inBytesReceived;
+      totalInPackets += entry.inPacketsReceived;
+
+      if (entry.hasOutbound)
+        anyOutbound = true;
+
+      if (entry.hasInbound)
+        anyInbound = true;
+    }
+
+    return {
+      outbound: anyOutbound
+        ? { bytesSent: totalOutBytes, packetsSent: totalOutPackets }
+        : null,
+      inbound: anyInbound
+        ? { bytesReceived: totalInBytes, packetsReceived: totalInPackets }
+        : null
+    };
+  });
+}
+
+/**
+ * Wait until at least one connection has both outbound-rtp and inbound-rtp
+ * video reports.
+ */
+export async function waitForVideoStatsPresent(page: Page, timeout = 15_000): Promise<void> {
+  await page.waitForFunction(
+    async () => {
+      const connections = (window as any).__rtcConnections as RTCPeerConnection[] | undefined;
+
+      if (!connections?.length)
+        return false;
+
+      for (const pc of connections) {
+        let stats: RTCStatsReport;
+
+        try {
+          stats = await pc.getStats();
+        } catch {
+          continue;
+        }
+
+        let hasOut = false;
+        let hasIn = false;
+
+        stats.forEach((report: any) => {
+          const kind = report.kind ?? report.mediaType;
+
+          if (report.type === 'outbound-rtp' && kind === 'video')
+            hasOut = true;
+
+          if (report.type === 'inbound-rtp' && kind === 'video')
+            hasIn = true;
+        });
+
+        if (hasOut && hasIn)
+          return true;
+      }
+
+      return false;
+    },
+    { timeout }
+  );
+}
+
+interface VideoFlowDelta {
+  outboundBytesDelta: number;
+  inboundBytesDelta: number;
+  outboundPacketsDelta: number;
+  inboundPacketsDelta: number;
+}
+
+function videoSnapshotToDelta(
+  curr: Awaited<ReturnType<typeof getVideoStats>>,
+  prev: Awaited<ReturnType<typeof getVideoStats>>
+): VideoFlowDelta {
+  return {
+    outboundBytesDelta: (curr.outbound?.bytesSent ?? 0) - (prev.outbound?.bytesSent ?? 0),
+    inboundBytesDelta: (curr.inbound?.bytesReceived ?? 0) - (prev.inbound?.bytesReceived ?? 0),
+    outboundPacketsDelta: (curr.outbound?.packetsSent ?? 0) - (prev.outbound?.packetsSent ?? 0),
+    inboundPacketsDelta: (curr.inbound?.packetsReceived ?? 0) - (prev.inbound?.packetsReceived ?? 0)
+  };
+}
+
+function isVideoDeltaFlowing(delta: VideoFlowDelta): boolean {
+  const outFlowing = delta.outboundBytesDelta > 0 || delta.outboundPacketsDelta > 0;
+  const inFlowing = delta.inboundBytesDelta > 0 || delta.inboundPacketsDelta > 0;
+
+  return outFlowing && inFlowing;
+}
+
+/**
+ * Poll until two consecutive HWM-based reads show both outbound and inbound
+ * video byte counts increasing - proving screen share video is flowing.
+ */
+export async function waitForVideoFlow(
+  page: Page,
+  timeoutMs = 30_000,
+  pollIntervalMs = 1_000
+): Promise<VideoFlowDelta> {
+  const deadline = Date.now() + timeoutMs;
+
+  let prev = await getVideoStats(page);
+
+  while (Date.now() < deadline) {
+    await page.waitForTimeout(pollIntervalMs);
+    const curr = await getVideoStats(page);
+    const delta = videoSnapshotToDelta(curr, prev);
+
+    if (isVideoDeltaFlowing(delta)) {
+      return delta;
+    }
+
+    prev = curr;
+  }
+
+  return {
+    outboundBytesDelta: 0,
+    inboundBytesDelta: 0,
+    outboundPacketsDelta: 0,
+    inboundPacketsDelta: 0
+  };
+}
+
+/**
+ * Wait until outbound video bytes are increasing (sender side).
+ * Use on the page that is sharing its screen.
+ */
+export async function waitForOutboundVideoFlow(
+  page: Page,
+  timeoutMs = 30_000,
+  pollIntervalMs = 1_000
+): Promise<VideoFlowDelta> {
+  const deadline = Date.now() + timeoutMs;
+
+  let prev = await getVideoStats(page);
+
+  while (Date.now() < deadline) {
+    await page.waitForTimeout(pollIntervalMs);
+    const curr = await getVideoStats(page);
+    const delta = videoSnapshotToDelta(curr, prev);
+
+    if (delta.outboundBytesDelta > 0 || delta.outboundPacketsDelta > 0) {
+      return delta;
+    }
+
+    prev = curr;
+  }
+
+  return {
+    outboundBytesDelta: 0,
+    inboundBytesDelta: 0,
+    outboundPacketsDelta: 0,
+    inboundPacketsDelta: 0
+  };
+}
+
+/**
+ * Wait until inbound video bytes are increasing (receiver side).
+ * Use on the page that is viewing someone else's screen share.
+ */
+export async function waitForInboundVideoFlow(
+  page: Page,
+  timeoutMs = 30_000,
+  pollIntervalMs = 1_000
+): Promise<VideoFlowDelta> {
+  const deadline = Date.now() + timeoutMs;
+
+  let prev = await getVideoStats(page);
+
+  while (Date.now() < deadline) {
+    await page.waitForTimeout(pollIntervalMs);
+    const curr = await getVideoStats(page);
+    const delta = videoSnapshotToDelta(curr, prev);
+
+    if (delta.inboundBytesDelta > 0 || delta.inboundPacketsDelta > 0) {
+      return delta;
+    }
+
+    prev = curr;
+  }
+
+  return {
+    outboundBytesDelta: 0,
+    inboundBytesDelta: 0,
+    outboundPacketsDelta: 0,
+    inboundPacketsDelta: 0
+  };
+}
+
+/**
+ * Dump full RTC connection diagnostics for debugging audio flow failures.
+ */
+export async function dumpRtcDiagnostics(page: Page): Promise<string> {
+  return page.evaluate(async () => {
+    const conns = (window as any).__rtcConnections as RTCPeerConnection[] | undefined;
+
+    if (!conns?.length)
+      return 'No connections tracked';
+
+    const lines: string[] = [`Total connections: ${conns.length}`];
+
+    for (let idx = 0; idx < conns.length; idx++) {
+      const pc = conns[idx];
+
+      lines.push(`PC[${idx}]: connection=${pc.connectionState}, signaling=${pc.signalingState}`);
+
+      const senders = pc.getSenders().map(
+        (sender) => `${sender.track?.kind ?? 'none'}:enabled=${sender.track?.enabled}:${sender.track?.readyState ?? 'null'}`
+      );
+      const receivers = pc.getReceivers().map(
+        (recv) => `${recv.track?.kind ?? 'none'}:enabled=${recv.track?.enabled}:${recv.track?.readyState ?? 'null'}`
+      );
+
+      lines.push(`  senders=[${senders.join(', ')}]`);
+      lines.push(`  receivers=[${receivers.join(', ')}]`);
+
+      try {
+        const stats = await pc.getStats();
+
+        stats.forEach((report: any) => {
+          if (report.type !== 'outbound-rtp' && report.type !== 'inbound-rtp')
+            return;
+
+          const kind = report.kind ?? report.mediaType;
+          const bytes = report.type === 'outbound-rtp' ? report.bytesSent : report.bytesReceived;
+          const packets = report.type === 'outbound-rtp' ? report.packetsSent : report.packetsReceived;
+
+          lines.push(`  ${report.type}: kind=${kind}, bytes=${bytes}, packets=${packets}`);
+        });
+      } catch (err: any) {
+        lines.push(`  getStats() failed: ${err?.message ?? err}`);
+      }
+    }
+
+    return lines.join('\n');
+  });
+}