fix serializeAudioLevelIndication

shinyoshiaki · Dec 13, 2023 · a50154c · a50154c
1 parent 2717adc
commit a50154c
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Showing 7 changed files with 84 additions and 8 deletions.
diff --git a/package-lock.json b/package-lock.json
diff --git a/packages/rtp/package.json b/packages/rtp/package.json
@@ -1,6 +1,6 @@
 {
   "name": "werift-rtp",
-  "version": "0.7.25",
+  "version": "0.7.26",
   "description": "RTP,RTCP,SRTP,SRTCP implementation for TypeScript.",
   "homepage": "https://github.com/shinyoshiaki/werift-webrtc",
   "repository": {

diff --git a/packages/rtp/src/helper.ts b/packages/rtp/src/helper.ts
@@ -23,7 +23,7 @@ export const timer = {
         args[0]();
       },
       //@ts-ignore
-      ...args.slice(1)
+      ...args.slice(1),
     );
     return () => clearInterval(id);
   },

diff --git a/packages/rtp/src/rtp/headerExtension.ts b/packages/rtp/src/rtp/headerExtension.ts
@@ -24,6 +24,9 @@ export function rtpHeaderExtensionsParser(
   return extensions
     .map((extension) => {
       const uri = extIdUriMap[extension.id];
+      if (!uri) {
+        return { uri: "unknown", value: extension.payload };
+      }
       switch (uri) {
         case RTP_EXTENSION_URI.sdesMid:
         case RTP_EXTENSION_URI.sdesRTPStreamID:
@@ -43,7 +46,7 @@ export function rtpHeaderExtensionsParser(
           };
         }
         default:
-          return { uri, value: 0 };
+          return { uri, value: extension.payload };
       }
     })
     .reduce((acc: { [uri: string]: any }, cur) => {
@@ -78,7 +81,7 @@ export function serializeAbsSendTime(ntpTime: bigint) {
 export function serializeAudioLevelIndication(level: number) {
   const stream = new BitStream(Buffer.alloc(1));
   stream.writeBits(1, 1);
-  stream.writeBits(level, 7);
+  stream.writeBits(7, level);
 
   return stream.uint8Array;
 }

diff --git a/packages/rtp/tests/rtp/packet.test.ts b/packages/rtp/tests/rtp/packet.test.ts
@@ -1,4 +1,4 @@
-import { RtpPacket } from "../../src/rtp/rtp";
+import { ExtensionProfiles, RtpHeader, RtpPacket } from "../../src/rtp/rtp";
 import { load } from "../utils";
 
 describe("packet", () => {
@@ -125,4 +125,33 @@ describe("packet", () => {
     const buf = p.serialize();
     expect(buf).toEqual(data);
   });
+
+  test("serialize_deserialize", () => {
+    const packet = new RtpPacket(
+      new RtpHeader({
+        csrc: [],
+        csrcLength: 0,
+        extension: true,
+        extensionProfile: ExtensionProfiles.OneByte,
+        extensions: [
+          { id: 1, payload: Buffer.from([48]) },
+          { id: 2, payload: Buffer.from([40, 1, 222]) },
+          { id: 10, payload: Buffer.from([128]) },
+        ],
+        marker: false,
+        padding: false,
+        paddingSize: 0,
+        payloadType: 109,
+        sequenceNumber: 15546,
+        ssrc: 2882400001,
+        timestamp: 144,
+        version: 2,
+      }),
+      Buffer.from([1, 2, 3, 4, 5, 6, 7, 8, 9, 10]),
+    );
+    const buf = packet.serialize();
+    const parsed = RtpPacket.deSerialize(buf);
+    const buf2 = parsed.serialize();
+    expect(buf2).toEqual(buf);
+  });
 });
diff --git a/packages/webrtc/package.json b/packages/webrtc/package.json
@@ -1,6 +1,6 @@
 {
   "name": "werift",
-  "version": "0.18.16",
+  "version": "0.18.17",
   "description": "WebRTC Implementation for TypeScript (Node.js)",
   "keywords": [
     "WebRTC",

diff --git a/packages/webrtc/tests/integrate/mediaChannel.test.ts b/packages/webrtc/tests/integrate/mediaChannel.test.ts
@@ -1,9 +1,11 @@
 import {
+  deserializeAudioLevelIndication,
   isMedia,
   MediaStreamTrack,
   RTCPeerConnection,
   RtpHeader,
   RtpPacket,
+  serializeAudioLevelIndication,
   useSdesMid,
 } from "../../src";
 
@@ -122,4 +124,46 @@ describe("media", () => {
     );
     await pc1.setRemoteDescription(pc2.localDescription!);
   });
+
+  test("test_sendonly_recvonly_audioLevel", async () =>
+    new Promise<void>(async (done) => {
+      const sendonly = new RTCPeerConnection();
+      const track = new MediaStreamTrack({ kind: "audio" });
+      sendonly.addTransceiver(track, { direction: "sendonly" });
+      sendonly.connectionStateChange
+        .watch((v) => v === "connected")
+        .then(() => {
+          const rtpPacket = new RtpPacket(
+            new RtpHeader({
+              extensions: [
+                { id: 10, payload: serializeAudioLevelIndication(25) },
+              ],
+            }),
+            Buffer.from("test")
+          ).serialize();
+          expect(isMedia(rtpPacket)).toBe(true);
+          track.writeRtp(rtpPacket);
+        });
+
+      const recvonly = new RTCPeerConnection();
+      recvonly.onRemoteTransceiverAdded.subscribe((transceiver) => {
+        transceiver.onTrack.subscribe((track) => {
+          track.onReceiveRtp.subscribe(async (rtp) => {
+            expect(rtp.header.extensions[0].id).toBe(10);
+            expect(
+              deserializeAudioLevelIndication(rtp.header.extensions[0].payload)
+                .level
+            ).toEqual(25);
+            expect(rtp.payload).toEqual(Buffer.from("test"));
+            await Promise.all([sendonly.close(), recvonly.close()]);
+            done();
+          });
+        });
+      });
+
+      await sendonly.setLocalDescription(await sendonly.createOffer());
+      await recvonly.setRemoteDescription(sendonly.localDescription!);
+      await recvonly.setLocalDescription(await recvonly.createAnswer());
+      await sendonly.setRemoteDescription(recvonly.localDescription!);
+    }));
 });