Revert "webrtc: startRtpForwarderProcess remove werift dependency"

This reverts commit 143a0b2.
koush · Apr 3, 2023 · 2863756 · 2863756
1 parent cc40885
commit 2863756
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Showing 3 changed files with 9 additions and 7 deletions.
diff --git a/plugins/webrtc/package-lock.json b/plugins/webrtc/package-lock.json
diff --git a/plugins/webrtc/package.json b/plugins/webrtc/package.json
@@ -1,6 +1,6 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.1.39",
+   "version": "0.1.40",
    "scripts": {
       "scrypted-setup-project": "scrypted-setup-project",
       "prescrypted-setup-project": "scrypted-package-json",

diff --git a/plugins/webrtc/src/rtp-forwarders.ts b/plugins/webrtc/src/rtp-forwarders.ts
@@ -1,3 +1,4 @@
+import { RtpPacket } from "../../../external/werift/packages/rtp/src/rtp/rtp";
 import { Deferred } from "@scrypted/common/src/deferred";
 import { closeQuiet, createBindZero, listenZeroSingleClient, reserveUdpPort } from "@scrypted/common/src/listen-cluster";
 import { ffmpegLogInitialOutput, safeKillFFmpeg, safePrintFFmpegArguments } from "@scrypted/common/src/media-helpers";
@@ -232,17 +233,17 @@ export async function startRtpForwarderProcess(console: Console, ffmpegInput: FF
                         // if the rtsp client is over tcp, then the restream server must also be tcp, as
                         // the rtp packets (which can be a max of 64k) may be too large for udp.
                         const clientIsTcp = await setupRtspClient(console, rtspClient, channel, audioSection, false, rtp => {
-                            const payload = rtp.subarray(12);
-
                             // live555 sends rtp aac packets without AU header followed by ADTS packets (which contain codec info)
                             // which ffmpeg can not handle.
                             // the solution is to demux the adts and send that to ffmpeg raw.
                             // https://github.com/mpv-player/mpv/issues/5669#issuecomment-932519409
                             if (firstPacket) {
                                 firstPacket = false;
                                 if (audioSection.codec === 'aac') {
+                                    const packet = RtpPacket.deSerialize(rtp);
+                                    const buf = packet.payload;
                                     // adts header is 12 bits of 1s
-                                    if (payload[0] == 0xff && (payload[1] & 0xf0) == 0xf0) {
+                                    if (buf[0] == 0xff && (buf[1] & 0xf0) == 0xf0) {
                                         adts = true;
                                         allowAudioTranscoderExit = true;
                                         const ffmpegArgs = [
@@ -270,7 +271,8 @@ export async function startRtpForwarderProcess(console: Console, ffmpegInput: FF
                                 rtspServer?.sendTrack(audioControl, rtp, false);
                             }
                             else {
-                                audioPipe?.write(payload);
+                                const packet = RtpPacket.deSerialize(rtp);
+                                audioPipe?.write(packet.payload);
                             }
                         });