Move rtcpSenderReportEnabled option to Streamer, make RTCP interval…

… time-based
Discord-RE · Feb 4, 2025 · 87e4468 · 87e4468
1 parent 82e8051
commit 87e4468
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Showing 6 changed files with 39 additions and 35 deletions.
diff --git a/src/client/Streamer.ts b/src/client/Streamer.ts
@@ -16,6 +16,10 @@ export type StreamerOptions = {
      * Force the use of ChaCha20 encryption. Faster on CPUs without AES-NI
      */
     forceChacha20Encryption: boolean;
+    /**
+     * Enable RTCP Sender Report for synchronization
+     */
+    rtcpSenderReportEnabled: boolean
 }
 
 export class Streamer {
@@ -28,6 +32,7 @@ export class Streamer {
         this._client = client;
         this._opts = {
             forceChacha20Encryption: false,
+            rtcpSenderReportEnabled: true,
             ...opts
         };
 

diff --git a/src/client/packet/BaseMediaPacketizer.ts b/src/client/packet/BaseMediaPacketizer.ts
@@ -21,8 +21,9 @@ export class BaseMediaPacketizer {
 
     private _totalBytes: number;
     private _totalPackets: number;
-    private _prevTotalPackets: number;
     private _lastPacketTime: number;
+    private _lastRtcpTime: number;
+    private _currentMediaTimestamp: number;
     private _srInterval: number;
 
     private _mediaUdp: MediaUdp;
@@ -36,12 +37,13 @@ export class BaseMediaPacketizer {
         this._timestamp = 0;
         this._totalBytes = 0;
         this._totalPackets = 0;
-        this._prevTotalPackets = 0;
         this._lastPacketTime = 0;
+        this._lastRtcpTime = 0;
+        this._currentMediaTimestamp = 0;
         this._mtu = 1200;
         this._extensionEnabled = extensionEnabled;
 
-        this._srInterval = 512; // Sane fallback value for interval
+        this._srInterval = 1000;
     }
 
     public get ssrc(): number | undefined
@@ -52,12 +54,11 @@ export class BaseMediaPacketizer {
     public set ssrc(value: number)
     {
         this._ssrc = value;
-        this._totalBytes = this._totalPackets = this._prevTotalPackets = 0;
+        this._totalBytes = this._totalPackets = 0;
     }
 
     /**
-     * The interval (number of packets) between 2 consecutive RTCP Sender
-     * Report packets
+     * The interval between 2 consecutive RTCP Sender Report packets in ms
      */
     public get srInterval(): number
     {
@@ -75,27 +76,32 @@ export class BaseMediaPacketizer {
     }
 
     public async onFrameSent(packetsSent: number, bytesSent: number, frametime: number): Promise<void> {
-        if(!this._mediaUdp.mediaConnection.streamOptions.rtcpSenderReportEnabled) return;
 
-        this._totalPackets = this._totalPackets + packetsSent;
-        this._totalBytes = (this._totalBytes + bytesSent) % max_int32bit;
-
-        // Not using modulo here, since the number of packet sent might not be
-        // exactly a multiple of the interval
-        if (Math.floor(this._totalPackets / this._srInterval) - Math.floor(this._prevTotalPackets / this._srInterval) > 0)
+        if (this._mediaUdp.mediaConnection.streamer.opts.rtcpSenderReportEnabled)
         {
-            const senderReport = await this.makeRtcpSenderReport();
-            this._mediaUdp.sendPacket(senderReport);
-            this._prevTotalPackets = this._totalPackets;
-            this._loggerRtcpSr.debug({
-                stats: {
-                    ssrc: this._ssrc,
-                    timestamp: this._timestamp,
-                    totalPackets: this._totalPackets,
-                    totalBytes: this._totalBytes
-                }
-            }, `Sent RTCP sender report for SSRC ${this._ssrc}`);
+            this._totalPackets = this._totalPackets + packetsSent;
+            this._totalBytes = (this._totalBytes + bytesSent) % max_int32bit;
+
+            /**
+             * Not using modulo here, since the timestamp might not be an exact
+             * multiple of the interval
+             */
+            if (Math.floor(this._currentMediaTimestamp / this._srInterval) - Math.floor(this._lastRtcpTime / this._srInterval) > 0)
+            {
+                const senderReport = await this.makeRtcpSenderReport();
+                this._mediaUdp.sendPacket(senderReport);
+                this._lastRtcpTime = this._currentMediaTimestamp;
+                this._loggerRtcpSr.debug({
+                    stats: {
+                        ssrc: this._ssrc,
+                        timestamp: this._timestamp,
+                        totalPackets: this._totalPackets,
+                        totalBytes: this._totalBytes
+                    }
+                }, `Sent RTCP sender report for SSRC ${this._ssrc}`);
+            }
         }
+        this._currentMediaTimestamp += frametime;
     }
 
     /**

diff --git a/src/client/packet/VideoPacketizerAnnexB.ts b/src/client/packet/VideoPacketizerAnnexB.ts
@@ -63,7 +63,6 @@ class VideoPacketizerAnnexB extends BaseMediaPacketizer {
 
     constructor(connection: MediaUdp, ssrc: number, payloadType: number, nalFunctions: AnnexBHelpers) {
         super(connection, ssrc, payloadType, true);
-        this.srInterval = 5 * connection.mediaConnection.streamOptions.fps * 3; // ~5 seconds, assuming ~3 packets per frame
         this._nalFunctions = nalFunctions;
     }
 

diff --git a/src/client/packet/VideoPacketizerVP8.ts b/src/client/packet/VideoPacketizerVP8.ts
@@ -13,7 +13,6 @@ export class VideoPacketizerVP8 extends BaseMediaPacketizer {
     constructor(connection: MediaUdp, ssrc: number) {
         super(connection, ssrc, CodecPayloadType.VP8.payload_type, true);
         this._pictureId = 0;
-        this.srInterval = 5 * connection.mediaConnection.streamOptions.fps * 3; // ~5 seconds, assuming ~3 packets per frame
     }
 
     private incrementPictureId(): void {

diff --git a/src/client/voice/BaseMediaConnection.ts b/src/client/voice/BaseMediaConnection.ts
@@ -109,6 +109,10 @@ export abstract class BaseMediaConnection extends EventEmitter {
         return this._transportEncryptor;
     }
 
+    public get streamer() {
+        return this._streamer;
+    }
+
     stop(): void {
         this.interval && clearInterval(this.interval);
         this.status.started = false;

diff --git a/src/media/newApi.ts b/src/media/newApi.ts
@@ -314,11 +314,6 @@ export type PlayStreamOptions = {
      * See https://ffmpeg.org/ffmpeg.html#:~:text=%2Dreadrate_initial_burst
      */
     readrateInitialBurst: number | undefined,
-
-    /**
-     * Enable RTCP Sender Report for synchronization
-     */
-    rtcpSenderReportEnabled: boolean
 }
 
 export async function playStream(
@@ -344,7 +339,6 @@ export async function playStream(
         height: video.height,
         frameRate: video.framerate_num / video.framerate_den,
         readrateInitialBurst: undefined,
-        rtcpSenderReportEnabled: true,
     } satisfies PlayStreamOptions;
 
     function mergeOptions(opts: Partial<PlayStreamOptions>)
@@ -373,9 +367,6 @@ export async function playStream(
                 isFiniteNonZero(opts.readrateInitialBurst) && opts.readrateInitialBurst > 0
                     ? opts.readrateInitialBurst
                     : defaultOptions.readrateInitialBurst,
-
-            rtcpSenderReportEnabled:
-                opts.rtcpSenderReportEnabled ?? defaultOptions.rtcpSenderReportEnabled
         } satisfies PlayStreamOptions
     }