test RTP timestamps

Overlord-Server/public/assets/webrtc-low-latency.test.ts

@@ -0,0 +1,49 @@
+import { describe, expect, test } from "bun:test";
+import { P2PClient } from "./webrtc-p2p.js";
+import { WebRTCStatsSampler } from "./webrtc-stats.js";
+
+describe("WebRTC low-latency paths", () => {
+  test("routes motion and reliable input over separate data channels", () => {
+    const client = new P2PClient({ send: () => {}, enableInput: true });
+    const motion: string[] = [];
+    const reliable: string[] = [];
+    client.motionInput = { readyState: "open", bufferedAmount: 0, send: (value: string) => motion.push(value) };
+    client.reliableInput = { readyState: "open", bufferedAmount: 0, send: (value: string) => reliable.push(value) };
+
+    client.inputReady = false;
+    expect(client.sendInput({ type: "key_down", code: "KeyA" })).toBe(false);
+    client.inputReady = true;
+    expect(client.sendInput({ type: "mouse_move", x: 10, y: 20 })).toBe(true);
+    expect(client.sendInput({ type: "key_down", code: "KeyA" })).toBe(true);
+    expect(JSON.parse(motion[0])).toEqual({ type: "mouse_move", x: 10, y: 20 });
+    expect(JSON.parse(reliable[0])).toEqual({ type: "key_down", code: "KeyA" });
+  });
+
+  test("reports interval jitter-buffer delay instead of lifetime average", async () => {
+    let reportIndex = 0;
+    const reports = [
+      new Map([["video", {
+        id: "video", type: "inbound-rtp", kind: "video", timestamp: 1000,
+        bytesReceived: 1000, framesDecoded: 10, framesDropped: 0,
+        packetsReceived: 10, packetsLost: 0, totalDecodeTime: 0,
+        totalProcessingDelay: 0, jitterBufferDelay: 3, jitterBufferEmittedCount: 10,
+      }]]),
+      new Map([["video", {
+        id: "video", type: "inbound-rtp", kind: "video", timestamp: 2000,
+        bytesReceived: 2000, framesDecoded: 20, framesDropped: 0,
+        packetsReceived: 20, packetsLost: 0, totalDecodeTime: 0,
+        totalProcessingDelay: 0, jitterBufferDelay: 3.5, jitterBufferEmittedCount: 20,
+      }]]),
+    ];
+    const samples: any[] = [];
+    const sampler = new WebRTCStatsSampler({
+      connectionState: "connected",
+      getStats: async () => reports[Math.min(reportIndex++, reports.length - 1)],
+    }, (sample: any) => samples.push(sample));
+
+    await sampler.sample();
+    await sampler.sample();
+    expect(samples[0].video.jitterBufferMs).toBe(300);
+    expect(samples[1].video.jitterBufferMs).toBe(50);
+  });
+});