webrtc: performance improvements

plugins/webrtc/package-lock.json

@@ -1,12 +1,12 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.1.19",
+   "version": "0.1.20",
    "lockfileVersion": 2,
    "requires": true,
    "packages": {
       "": {
          "name": "@scrypted/webrtc",
-         "version": "0.1.19",
+         "version": "0.1.20",
          "dependencies": {
             "@scrypted/common": "file:../../common",
             "@scrypted/sdk": "file:../../sdk",

plugins/webrtc/package.json

@@ -1,6 +1,6 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.1.19",
+   "version": "0.1.20",
    "scripts": {
       "scrypted-setup-project": "scrypted-setup-project",
       "prescrypted-setup-project": "scrypted-package-json",

plugins/webrtc/src/ffmpeg-to-wrtc.ts

@@ -177,7 +177,7 @@ export async function createTrackForwarder(options: {
 
     const audioRtpTrack: RtpTrack = {
         codecCopy: audioCodecCopy,
-        onRtp: buffer => audioTransceiver.sender.sendRtp(buffer),
+        onRtp: audioTransceiver.sender.sendRtp.bind(audioTransceiver.sender),
         encoderArguments: [
             ...audioTranscodeArguments,
         ],
@@ -210,6 +210,8 @@ export async function createTrackForwarder(options: {
         packetSize: videoPacketSize,
         onMSection: (videoSection) => spsPps = getSpsPps(videoSection),
         onRtp: (buffer) => {
+            let onRtp: (rtp: Buffer) => void;
+
             if (needPacketization) {
                 if (!h264Repacketizer) {
                     // adjust packet size for the rtp packet header (12).
@@ -217,14 +219,21 @@ export async function createTrackForwarder(options: {
                         ...spsPps,
                     });
                 }
-                const repacketized = h264Repacketizer.repacketize(RtpPacket.deSerialize(buffer));
-                for (const packet of repacketized) {
-                    videoTransceiver.sender.sendRtp(packet);
-                }
+                onRtp = buffer => {
+                    const repacketized = h264Repacketizer.repacketize(RtpPacket.deSerialize(buffer));
+                    for (const packet of repacketized) {
+                        videoTransceiver.sender.sendRtp(packet);
+                    }
+                };
             }
             else {
-                videoTransceiver.sender.sendRtp(buffer);
+                onRtp = buffer => {
+                    videoTransceiver.sender.sendRtp(buffer);
+                };
             }
+
+            videoRtpTrack.onRtp = onRtp;
+            videoRtpTrack.onRtp(buffer);
         },
         encoderArguments: [
             ...videoTranscodeArguments,

plugins/webrtc/src/rtp-forwarders.ts

@@ -46,14 +46,14 @@ export type RtpSockets = {
 };
 
 function createPacketDelivery(track: RtpTrack) {
-    let firstPacket = true;
-    return (rtp: Buffer) => {
-        if (firstPacket) {
-            firstPacket = false;
-            track.firstPacket?.(rtp);
-        }
+    const original = track.onRtp;
+    track.onRtp = rtp => {
+        track.onRtp = original;
+        track.firstPacket?.(rtp);
         track.onRtp(rtp);
     }
+
+    return (rtp: Buffer) => track.onRtp(rtp);
 }
 
 function attachTrackDgram(track: RtpTrack, server: dgram.Socket) {