vitest

shinyoshiaki · May 20, 2024 · d82e9e4 · d82e9e4
1 parent f3b1a27
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diff --git a/package-lock.json b/package-lock.json
diff --git a/package.json b/package.json
@@ -3,15 +3,21 @@
   "version": "0.0.1",
   "private": "true",
   "description": "WebRTC Implementation for TypeScript (Node.js)",
-  "keywords": ["WebRTC", "node.js"],
+  "keywords": [
+    "WebRTC",
+    "node.js"
+  ],
   "homepage": "https://github.com/shinyoshiaki/werift-webrtc",
   "repository": {
     "type": "git",
     "url": "https://github.com/shinyoshiaki/werift-webrtc.git"
   },
   "license": "MIT",
   "author": "shinyoshiaki <[email protected]>",
-  "workspaces": ["packages/*", "examples/*"],
+  "workspaces": [
+    "packages/*",
+    "examples/*"
+  ],
   "scripts": {
     "build": "npm run build --workspaces --if-present",
     "ci": "npm run type:all && npm run build && npm run test && npm run e2e && npm run doc",
@@ -44,7 +50,8 @@
     "ts-node-dev": "^2.0.0",
     "typedoc": "0.25.5",
     "typedoc-plugin-markdown": "3.17.1",
-    "typescript": "5.3.3"
+    "typescript": "5.3.3",
+    "vitest": "^1.6.0"
   },
   "packageManager": "[email protected]",
   "engines": {

diff --git a/packages/webrtc/package.json b/packages/webrtc/package.json
@@ -33,7 +33,7 @@
     "doc": "rm -rf doc && typedoc --excludePrivate --disableSources --excludeInternal --plugin typedoc-plugin-markdown --out doc ./src/index.ts",
     "format": "biome check --apply src tests",
     "knip": "knip",
-    "test": "npm run type && jest --runInBand --forceExit --coverage",
+    "test": "vitest run ./tests",
     "type": "tsc --noEmit --project ./tsconfig.json",
     "unused": "organize-imports-cli src/**/*.ts",
     "upgrade-interactive": "npx npm-check-updates -i"

diff --git a/packages/webrtc/tests/datachannel/close.test.ts b/packages/webrtc/tests/datachannel/close.test.ts
@@ -1,6 +1,6 @@
 import { createDataChannelPair } from "../utils";
 
-jest.setTimeout(10_000);
+
 
 describe.each([{}, { negotiated: true, id: 0 }])(
   "datachannel/close",

diff --git a/packages/webrtc/tests/datachannel/send.test.ts b/packages/webrtc/tests/datachannel/send.test.ts
@@ -1,6 +1,6 @@
 import { awaitMessage, createDataChannelPair } from "../utils";
 
-jest.setTimeout(15_000);
+
 
 const helloString = "hello";
 const unicodeString = "世界你好";

diff --git a/packages/webrtc/tests/integrate/datachannel.test.ts b/packages/webrtc/tests/integrate/datachannel.test.ts
@@ -1,6 +1,6 @@
 import { RTCPeerConnection } from "../../src";
 
-jest.setTimeout(10_000);
+
 
 describe("datachannel", () => {
   test("send some messages at same time", async () =>

diff --git a/packages/webrtc/tests/integrate/mediaChannel.test.ts b/packages/webrtc/tests/integrate/mediaChannel.test.ts
@@ -9,7 +9,7 @@ import {
   useSdesMid,
 } from "../../src";
 
-jest.setTimeout(10_000);
+
 
 describe("media", () => {
   test("test_sendonly_recvonly", async () =>

diff --git a/packages/webrtc/tests/integrate/peerConnection.test.ts b/packages/webrtc/tests/integrate/peerConnection.test.ts
@@ -8,7 +8,7 @@ import {
 } from "../../src";
 import { SignatureAlgorithm } from "../../src/const";
 
-jest.setTimeout(10_000);
+
 
 describe("peerConnection", () => {
   test("test_connect_datachannel_modern_sdp", async () =>
@@ -135,7 +135,7 @@ describe("peerConnection", () => {
       await assertDataChannelOpen(dc);
     }));
 
-  xtest("portRange", async () => {
+  test.skip("portRange", async () => {
     const peer = new RTCPeerConnection({ icePortRange: [44444, 44455] });
     peer.createDataChannel("test");
     const offer = await peer.createOffer();

diff --git a/packages/webrtc/tests/integrate/trickle.test.ts b/packages/webrtc/tests/integrate/trickle.test.ts
@@ -1,6 +1,6 @@
 import { RTCPeerConnection } from "../../src";
 
-jest.setTimeout(10_000);
+
 
 describe("trickle", () => {
   test(

diff --git a/packages/webrtc/tests/media/router.test.ts b/packages/webrtc/tests/media/router.test.ts
@@ -15,7 +15,7 @@ import { RTCRtpSender } from "../../src/media/rtpSender";
 import { createDtlsTransport } from "../fixture";
 
 describe("media/router", () => {
-  test("routeRtp", (done) => {
+  test("routeRtp", ()=>new Promise<void>((done) => {
     const router = new RtpRouter();
     const dtls = createDtlsTransport();
     const transceiver = new RTCRtpTransceiver(
@@ -56,5 +56,5 @@ describe("media/router", () => {
       done();
     });
     router.routeRtp(packet);
-  });
+  }));
 });
diff --git a/packages/webrtc/tests/media/rtpReceiver.test.ts b/packages/webrtc/tests/media/rtpReceiver.test.ts
@@ -12,6 +12,7 @@ import {
 } from "../../src";
 import { RTCRtpReceiver } from "../../src/media/rtpReceiver";
 import { createDtlsTransport } from "../fixture";
+import {vi} from 'vitest'
 
 describe("packages/webrtc/src/media/rtpReceiver.ts", () => {
   test("abort runRtcp", async () =>
@@ -20,7 +21,7 @@ describe("packages/webrtc/src/media/rtpReceiver.ts", () => {
       const receiver = new RTCRtpReceiver(defaultPeerConfig, "audio", 1234);
       receiver.setDtlsTransport(dtls);
 
-      jest.spyOn(dtls, "sendRtcp");
+      vi.spyOn(dtls, "sendRtcp");
 
       Promise.any([
         setTimeout(200).then(() => false),

diff --git a/packages/webrtc/tests/media/rtpSender.test.ts b/packages/webrtc/tests/media/rtpSender.test.ts
@@ -3,6 +3,7 @@ import { setTimeout } from "timers/promises";
 import { MediaStreamTrack } from "../../src";
 import { RTCRtpSender } from "../../src/media/rtpSender";
 import { createDtlsTransport, createRtpPacket } from "../fixture";
+import {vi} from 'vitest'
 
 describe("media/rtpSender", () => {
   test("stop track", () => {
@@ -11,7 +12,7 @@ describe("media/rtpSender", () => {
     const sender = new RTCRtpSender(track);
     sender.setDtlsTransport(dtls);
 
-    const spy = jest.spyOn(sender, "sendRtp");
+    const spy = vi.spyOn(sender, "sendRtp");
 
     const rtp = createRtpPacket();
 
@@ -31,7 +32,7 @@ describe("media/rtpSender", () => {
     const dtls = createDtlsTransport();
     const sender = new RTCRtpSender(track1);
     sender.setDtlsTransport(dtls);
-    const spy = jest.spyOn(sender, "sendRtp");
+    const spy = vi.spyOn(sender, "sendRtp");
 
     const rtp = createRtpPacket();
 
@@ -55,7 +56,7 @@ describe("media/rtpSender", () => {
       const sender = new RTCRtpSender("audio");
       sender.setDtlsTransport(dtls);
 
-      jest.spyOn(dtls, "sendRtcp");
+
 
       Promise.any([
         setTimeout(200).then(() => false),

diff --git a/packages/webrtc/tests/transport/dtls.test.ts b/packages/webrtc/tests/transport/dtls.test.ts
@@ -2,7 +2,7 @@ import { setTimeout } from "timers/promises";
 
 import { dtlsTransportPair } from "../fixture";
 
-jest.setTimeout(10_000);
+
 
 describe("RTCDtlsTransportTest", () => {
   test("dtls_test_data", async () => {

diff --git a/packages/webrtc/tests/transport/ice.test.ts b/packages/webrtc/tests/transport/ice.test.ts
@@ -2,7 +2,7 @@ import inRange from "lodash/inRange";
 
 import { RTCIceGatherer, RTCIceTransport } from "../../src";
 
-jest.setTimeout(10_000);
+
 
 describe("iceTransport", () => {
   test("test_connect", async () => {
@@ -40,7 +40,7 @@ describe("iceTransport", () => {
     expect(transport2.state).toBe("closed");
   });
 
-  xtest("portRange", async () => {
+  test.skip("portRange", async () => {
     const gatherer = new RTCIceGatherer({
       stunServer: ["stun.l.google.com", 19302],
       portRange: [44444, 44455],
@@ -55,7 +55,7 @@ describe("iceTransport", () => {
     await gatherer.connection.close();
   });
 
-  xtest("minimum target port", async () => {
+  test.skip("minimum target port", async () => {
     const gatherer = new RTCIceGatherer({
       stunServer: ["stun.l.google.com", 19302],
       portRange: [44546, 44547],

diff --git a/packages/webrtc/tests/transport/sctp.test.ts b/packages/webrtc/tests/transport/sctp.test.ts
@@ -2,7 +2,7 @@ import { RTCDataChannel, RTCSctpTransport } from "../../src";
 import { RTCDataChannelParameters } from "../../src/dataChannel";
 import { dtlsTransportPair } from "../fixture";
 
-jest.setTimeout(10_000);
+
 
 describe("RTCSctpTransportTest", () => {
   function trackChannels(transport: RTCSctpTransport) {

diff --git a/packages/webrtc/tests/wpt/addTrack.test.ts b/packages/webrtc/tests/wpt/addTrack.test.ts
@@ -1,7 +1,7 @@
 import { MediaStreamTrack, RTCPeerConnection } from "../../src";
 import { RTCRtpSender } from "../../src/media/rtpSender";
 
-jest.setTimeout(10_000);
+
 
 // https://github.com/web-platform-tests/wpt/blob/master/webrtc/RTCPeerConnection-addTrack.https.html
 

diff --git a/packages/webrtc/tests/wpt/onnegotiaationneeded.test.ts b/packages/webrtc/tests/wpt/onnegotiaationneeded.test.ts
@@ -4,7 +4,7 @@ import { RTCPeerConnection } from "../../src";
 import { generateAnswer, generateOffer } from "../fixture";
 import { generateAudioReceiveOnlyOffer } from "../utils";
 
-jest.setTimeout(10_000);
+
 
 describe("onnegotiationneeded", () => {
   test("Creating first data channel should fire negotiationneeded event", async () => {

diff --git a/packages/webrtc/tests/wpt/ontrack.test.ts b/packages/webrtc/tests/wpt/ontrack.test.ts
@@ -10,7 +10,7 @@ import { RTCRtpReceiver } from "../../src/media/rtpReceiver";
 import { MediaStream } from "../../src/media/track";
 import { addEventListenerPromise, assert_equals, assert_true } from "../utils";
 
-jest.setTimeout(10_000);
+
 
 describe("wpt/ontrack", () => {
   it("setRemoteDescription should trigger ontrack event when the MSID of the stream is parsed.", async () => {

diff --git a/packages/webrtc/tests/wpt/removeTrack.test.ts b/packages/webrtc/tests/wpt/removeTrack.test.ts
@@ -1,7 +1,7 @@
 import { MediaStreamTrack, RTCPeerConnection } from "../../src";
 import { generateAnswer } from "../fixture";
 
-jest.setTimeout(10_000);
+
 
 // webrtc/RTCPeerConnection-removeTrack.https.html
 describe("peerConnection/removeTrack", () => {
@@ -216,7 +216,7 @@ describe("peerConnection/removeTrack", () => {
   });
 
   // todo fix
-  xtest("Calling removeTrack on a stopped transceiver should be a no-op", async () => {
+  test.skip("Calling removeTrack on a stopped transceiver should be a no-op", async () => {
     const pc = new RTCPeerConnection();
 
     const track = new MediaStreamTrack({ kind: "audio" });

diff --git a/packages/webrtc/vitest.config.ts b/packages/webrtc/vitest.config.ts
@@ -0,0 +1,9 @@
+import { defineConfig } from "vitest/config";
+
+export default defineConfig({
+  test: {
+    globals: true,
+    minWorkers: 2,
+    maxWorkers: 2,
+  },
+});