add example

examples/package.json

@@ -2,7 +2,10 @@
   "name": "example",
   "version": "0.0.31",
   "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",
@@ -13,7 +16,10 @@
     "name": "shinyoshiaki"
   },
   "main": "lib/index.js",
-  "files": ["lib", "src"],
+  "files": [
+    "lib",
+    "src"
+  ],
   "scripts": {
     "format": "biome check --write ../examples"
   },
@@ -49,7 +55,8 @@
     "ws": "^8.17.1",
     "wtfnode": "^0.9.1",
     "xmlbuilder2": "^3.0.2",
-    "yargs": "^17.7.2"
+    "yargs": "^17.7.2",
+    "zx": "^8.1.6"
   },
   "engines": {
     "node": ">=14"

examples/save_to_disk/rtp.ts

@@ -0,0 +1,26 @@
+import { randomUUID } from "crypto";
+import {
+  MediaRecorder,
+  Navigator,
+} from "../../packages/webrtc/src/nonstandard";
+import { randomPort } from "../../packages/webrtc/src";
+import { $ } from "zx";
+
+(async () => {
+  const path = `${__dirname}/tmp${randomUUID()}.webm`;
+  const recorder = new MediaRecorder({
+    numOfTracks: 1,
+    path,
+    disableNtp: true,
+  });
+
+  const port = await randomPort();
+  const navigator = new Navigator();
+  const { track } = navigator.mediaDevices.getUdpMedia({
+    port,
+    codec: { clockRate: 48000, mimeType: "audio/opus", payloadType: 96 },
+  });
+  await recorder.addTrack(track);
+
+  $`gst-launch-1.0 audiotestsrc ! audioconvert ! audioresample ! opusenc ! rtpopuspay ! udpsink host=127.0.0.1 port=${port}`;
+})();

packages/webrtc/src/nonstandard/navigator.ts

@@ -1,6 +1,8 @@
 import { randomBytes } from "crypto";
 import { jspack } from "@shinyoshiaki/jspack";
 import { MediaStream, MediaStreamTrack } from "../media/track";
+import { createSocket } from "dgram";
+import { RTCRtpCodecParameters } from "..";
 
 export class Navigator {
   mediaDevices: MediaDevices;
@@ -58,6 +60,31 @@ export class MediaDevices extends EventTarget {
   };
 
   readonly getDisplayMedia = this.getUserMedia;
+
+  readonly getUdpMedia = ({
+    port,
+    codec,
+  }: {
+    port: number;
+    codec: ConstructorParameters<typeof RTCRtpCodecParameters>[0];
+  }) => {
+    const track = new MediaStreamTrack({
+      kind: "audio",
+      codec: new RTCRtpCodecParameters(codec),
+    });
+
+    const udp = createSocket("udp4");
+    udp.bind(port);
+    udp.on("message", (data) => {
+      track.writeRtp(data);
+    });
+
+    const disposer = () => {
+      udp.close();
+    };
+
+    return { track, disposer };
+  };
 }
 
 interface MediaStreamConstraints {

packages/webrtc/src/nonstandard/recorder/writer/webm.ts

@@ -77,6 +77,10 @@ export class WebmFactory extends MediaWriter {
     });
     const lipsync = new LipsyncCallback();
 
+    if (inputTracks.length === 1) {
+      this.props.disableLipSync = true;
+    }
+
     this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {
       const rtpSource = new RtpSourceCallback();
       const rtcpSource = new RtcpSourceCallback();