piixishstream 1.0.3 → 1.0.5

package/dist/client/Streamer.js

@@ -82,33 +82,47 @@ export class Streamer {
                 reject("cannot start stream without first joining voice channel");
                 return;
             }
-            this.signalStream();
-            const { guildId: clientGuildId, channelId: clientChannelId, session_id, } = this.voiceConnection;
+            if (this.voiceConnection.streamConnection?.webRtcConn?.ready) {
+                resolve(this.voiceConnection.streamConnection.webRtcConn);
+                return;
+            }
+            const { guildId: clientGuildId, channelId: clientChannelId, session_id } = this.voiceConnection;
             const { id: clientUserId } = this.client.user;
-            if (!session_id)
-                throw new Error("Session doesn't exist yet");
-            const streamConn = new StreamConnection(this, clientGuildId, clientUserId, clientChannelId, (conn) => {
-                resolve(conn);
-            });
-            this.voiceConnection.streamConnection = streamConn;
-            this._gatewayEmitter.on("STREAM_CREATE", (d) => {
+            if (!session_id) throw new Error("Session doesn't exist yet");
+
+            let done = false;
+            const onStreamCreate = (d) => {
                 const { channelId, guildId, userId } = parseStreamKey(d.stream_key);
-                if (clientGuildId !== guildId ||
-                    clientChannelId !== channelId ||
-                    clientUserId !== userId)
-                    return;
+                if (clientGuildId !== guildId || clientChannelId !== channelId || clientUserId !== userId) return;
                 streamConn.serverId = d.rtc_server_id;
                 streamConn.streamKey = d.stream_key;
                 streamConn.setSession(session_id);
-            });
-            this._gatewayEmitter.on("STREAM_SERVER_UPDATE", (d) => {
+            };
+            const onStreamServerUpdate = (d) => {
                 const { channelId, guildId, userId } = parseStreamKey(d.stream_key);
-                if (clientGuildId !== guildId ||
-                    clientChannelId !== channelId ||
-                    clientUserId !== userId)
-                    return;
+                if (clientGuildId !== guildId || clientChannelId !== channelId || clientUserId !== userId) return;
                 streamConn.setTokens(d.endpoint, d.token);
+            };
+            const cleanup = () => {
+                if (done) return;
+                done = true;
+                clearTimeout(timer);
+                this._gatewayEmitter.off("STREAM_CREATE", onStreamCreate);
+                this._gatewayEmitter.off("STREAM_SERVER_UPDATE", onStreamServerUpdate);
+            };
+            const timer = setTimeout(() => {
+                cleanup();
+                reject(new Error("Timeout en attente de la passerelle Discord Go-Live"));
+            }, 10000);
+
+            this.signalStream();
+            const streamConn = new StreamConnection(this, clientGuildId, clientUserId, clientChannelId, (conn) => {
+                cleanup();
+                resolve(conn);
             });
+            this.voiceConnection.streamConnection = streamConn;
+            this._gatewayEmitter.on("STREAM_CREATE", onStreamCreate);
+            this._gatewayEmitter.on("STREAM_SERVER_UPDATE", onStreamServerUpdate);
         });
     }
     async setStreamPreview(image) {

package/dist/client/voice/WebRtcWrapper.js

@@ -60,6 +60,25 @@ export class WebRtcConnWrapper {
     get webRtcConn() {
         return this._webRtcConn;
     }
+    waitUntilReady(timeoutMs = 5000) {
+        if (this.ready) return Promise.resolve();
+        return new Promise((resolve) => {
+            let done = false;
+            const finish = () => {
+                if (!done) {
+                    done = true;
+                    clearTimeout(timer);
+                    resolve();
+                }
+            };
+            const timer = setTimeout(finish, timeoutMs);
+            this._webRtcConn?.onStateChange((state) => {
+                if (state === "connected") {
+                    finish();
+                }
+            });
+        });
+    }
     get ready() {
         return this._webRtcConn?.state() === "connected";
     }
@@ -75,7 +94,7 @@ export class WebRtcConnWrapper {
         const { clockRate } = rtpConfig;
         if (this.mediaConnection.daveReady)
             frame = this.mediaConnection.daveSession.encryptOpus(frame);
-        this._audioTrack?.sendMessageBinary(frame);
+        try { this._audioTrack?.sendMessageBinary(frame); } catch (e) { /* ignore SRTP/WebRTC drop errors */ }
         rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
     }
     sendVideoFrame(frame, frametime) {
@@ -118,7 +137,7 @@ export class WebRtcConnWrapper {
             }
             frame = this.mediaConnection.daveSession.encrypt(1 /* MediaType.VIDEO */, daveCodec, frame);
         }
-        this._videoTrack?.sendMessageBinary(frame);
+        try { this._videoTrack?.sendMessageBinary(frame); } catch (e) { /* ignore SRTP/WebRTC drop errors */ }
         rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
     }
     setPacketizer(videoCodec) {

package/dist/media/newApi.js

@@ -274,6 +274,7 @@ export async function playStream(input, streamer, options = {}, cancelSignal) {
                 ? opts.readrateInitialBurst
                 : defaultOptions.readrateInitialBurst,
             streamPreview: opts.streamPreview ?? defaultOptions.streamPreview,
+            keepConnection: opts.keepConnection ?? false,
         };
     }
     const mergedOptions = mergeOptions(options);
@@ -297,12 +298,12 @@ export async function playStream(input, streamer, options = {}, cancelSignal) {
     let stopStream;
     if (mergedOptions.type === "go-live") {
         conn = await streamer.createStream();
-        stopStream = () => streamer.stopStream();
+        stopStream = () => { if (!mergedOptions.keepConnection) streamer.stopStream(); };
     }
     else {
         conn = streamer.voiceConnection.webRtcConn;
         streamer.signalVideo(true);
-        stopStream = () => streamer.signalVideo(false);
+        stopStream = () => { if (!mergedOptions.keepConnection) streamer.signalVideo(false); };
     }
     conn.setPacketizer(videoCodecMap[video.codec]);
     conn.mediaConnection.setSpeaking(true);
@@ -383,8 +384,10 @@ export async function playStream(input, streamer, options = {}, cancelSignal) {
     return new Promise((resolve, reject) => {
         cleanupFuncs.push(() => {
             stopStream();
-            conn.mediaConnection.setSpeaking(false);
-            conn.mediaConnection.setVideoAttributes(false);
+            if (!mergedOptions.keepConnection) {
+                conn.mediaConnection.setSpeaking(false);
+                conn.mediaConnection.setVideoAttributes(false);
+            }
         });
         let cleanedUp = false;
         const cleanup = () => {

package/package.json

@@ -1,6 +1,6 @@
 {
     "name": "piixishstream",
-    "version": "1.0.3",
+    "version": "1.0.5",
     "description": "Experiment for making video streaming work for discord selfbots",
     "main": "./dist/index.js",
     "exports": "./dist/index.js",
@@ -80,4 +80,5 @@
             "node-datachannel"
         ]
     }
-}
+}
+

package/src/client/Streamer.ts

@@ -104,63 +104,58 @@ export class Streamer {
     });
   }
 
-  public createStream(): Promise<WebRtcConnWrapper> {
-    return new Promise<WebRtcConnWrapper>((resolve, reject) => {
-      if (!this.client.user) {
-        reject("Client not logged in");
-        return;
-      }
-      if (!this.voiceConnection) {
-        reject("cannot start stream without first joining voice channel");
-        return;
-      }
-
-      this.signalStream();
-      const {
-        guildId: clientGuildId,
-        channelId: clientChannelId,
-        session_id,
-      } = this.voiceConnection;
-      const { id: clientUserId } = this.client.user;
-
-      if (!session_id) throw new Error("Session doesn't exist yet");
-      const streamConn = new StreamConnection(
-        this,
-        clientGuildId,
-        clientUserId,
-        clientChannelId,
-        (conn) => {
-          resolve(conn);
-        },
-      );
-      this.voiceConnection.streamConnection = streamConn;
-      this._gatewayEmitter.on("STREAM_CREATE", (d) => {
-        const { channelId, guildId, userId } = parseStreamKey(d.stream_key);
-
-        if (
-          clientGuildId !== guildId ||
-          clientChannelId !== channelId ||
-          clientUserId !== userId
-        )
-          return;
-
-        streamConn.serverId = d.rtc_server_id;
-        streamConn.streamKey = d.stream_key;
-        streamConn.setSession(session_id);
-      });
-      this._gatewayEmitter.on("STREAM_SERVER_UPDATE", (d) => {
-        const { channelId, guildId, userId } = parseStreamKey(d.stream_key);
-
-        if (
-          clientGuildId !== guildId ||
-          clientChannelId !== channelId ||
-          clientUserId !== userId
-        )
-          return;
-
-        streamConn.setTokens(d.endpoint, d.token);
-      });
-    });
+  public createStream(): Promise<WebRtcConnWrapper> {
+    return new Promise<WebRtcConnWrapper>((resolve, reject) => {
+      if (!this.client.user) {
+        reject("Client not logged in");
+        return;
+      }
+      if (!this.voiceConnection) {
+        reject("cannot start stream without first joining voice channel");
+        return;
+      }
+      if (this.voiceConnection.streamConnection?.webRtcConn?.ready) {
+        resolve(this.voiceConnection.streamConnection.webRtcConn);
+        return;
+      }
+      const { guildId: clientGuildId, channelId: clientChannelId, session_id } = this.voiceConnection;
+      const { id: clientUserId } = this.client.user;
+      if (!session_id) throw new Error("Session doesn't exist yet");
+
+      let done = false;
+      const onStreamCreate = (d: any) => {
+        const { channelId, guildId, userId } = parseStreamKey(d.stream_key);
+        if (clientGuildId !== guildId || clientChannelId !== channelId || clientUserId !== userId) return;
+        streamConn.serverId = d.rtc_server_id;
+        streamConn.streamKey = d.stream_key;
+        streamConn.setSession(session_id);
+      };
+      const onStreamServerUpdate = (d: any) => {
+        const { channelId, guildId, userId } = parseStreamKey(d.stream_key);
+        if (clientGuildId !== guildId || clientChannelId !== channelId || clientUserId !== userId) return;
+        streamConn.setTokens(d.endpoint, d.token);
+      };
+      const cleanup = () => {
+        if (done) return;
+        done = true;
+        clearTimeout(timer);
+        this._gatewayEmitter.off("STREAM_CREATE", onStreamCreate);
+        this._gatewayEmitter.off("STREAM_SERVER_UPDATE", onStreamServerUpdate);
+      };
+      const timer = setTimeout(() => {
+        cleanup();
+        reject(new Error("Timeout en attente de la passerelle Discord Go-Live"));
+      }, 10000);
+
+      this.signalStream();
+      const streamConn = new StreamConnection(this, clientGuildId, clientUserId, clientChannelId, (conn) => {
+        cleanup();
+        resolve(conn);
+      });
+      this.voiceConnection.streamConnection = streamConn;
+      this._gatewayEmitter.on("STREAM_CREATE", onStreamCreate);
+      this._gatewayEmitter.on("STREAM_SERVER_UPDATE", onStreamServerUpdate);
+    });
   }
 
   public async setStreamPreview(image: Buffer): Promise<void> {

package/src/client/voice/WebRtcWrapper.ts

@@ -93,7 +93,26 @@ export class WebRtcConnWrapper {
     return this._webRtcConn;
   }
 
-  public get ready() {
+  public waitUntilReady(timeoutMs = 5000) {
+        if (this.ready) return Promise.resolve();
+        return new Promise((resolve) => {
+            let done = false;
+            const finish = () => {
+                if (!done) {
+                    done = true;
+                    clearTimeout(timer);
+                    resolve();
+                }
+            };
+            const timer = setTimeout(finish, timeoutMs);
+            this._webRtcConn?.onStateChange((state) => {
+                if (state === "connected") {
+                    finish();
+                }
+            });
+        });
+    }
+    get ready() {
     return this._webRtcConn?.state() === "connected";
   }
 
@@ -108,7 +127,7 @@ export class WebRtcConnWrapper {
     const { clockRate } = rtpConfig;
     if (this.mediaConnection.daveReady)
       frame = this.mediaConnection.daveSession!.encryptOpus(frame);
-    this._audioTrack?.sendMessageBinary(frame);
+    try { this._audioTrack?.sendMessageBinary(frame); } catch (e) { /* ignore SRTP/WebRTC drop errors */ }
     rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
   }
 
@@ -154,7 +173,7 @@ export class WebRtcConnWrapper {
         frame,
       );
     }
-    this._videoTrack?.sendMessageBinary(frame);
+    try { this._videoTrack?.sendMessageBinary(frame); } catch (e) { /* ignore SRTP/WebRTC drop errors */ }
     rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
   }
 

package/src/media/newApi.ts

@@ -492,7 +492,8 @@ export async function playStream(
           ? opts.readrateInitialBurst
           : defaultOptions.readrateInitialBurst,
 
-      streamPreview: opts.streamPreview ?? defaultOptions.streamPreview,
+      streamPreview: opts.streamPreview ?? defaultOptions.streamPreview,
+            keepConnection: opts.keepConnection ?? false,
     } satisfies PlayStreamOptions;
   }
 
@@ -520,11 +521,11 @@ export async function playStream(
   let stopStream: () => unknown;
   if (mergedOptions.type === "go-live") {
     conn = await streamer.createStream();
-    stopStream = () => streamer.stopStream();
+    stopStream = () => { if (!mergedOptions.keepConnection) streamer.stopStream(); };
   } else {
     conn = streamer.voiceConnection.webRtcConn;
     streamer.signalVideo(true);
-    stopStream = () => streamer.signalVideo(false);
+    stopStream = () => { if (!mergedOptions.keepConnection) streamer.signalVideo(false); };
   }
   conn.setPacketizer(videoCodecMap[video.codec]);
   conn.mediaConnection.setSpeaking(true);