werift 0.22.1 → 0.22.3

package/lib/webrtc/src/media/parameters.d.ts

@@ -1,4 +1,4 @@
-import type { Direction } from "./rtpTransceiver";
+import type { MediaDirection } from "./rtpTransceiver";
 export interface RTCRtpParameters {
     codecs: RTCRtpCodecParameters[];
     headerExtensions: RTCRtpHeaderExtensionParameters[];
@@ -22,7 +22,7 @@ export declare class RTCRtpCodecParameters {
     channels?: number;
     rtcpFeedback: RTCPFB[];
     parameters?: string;
-    direction: Direction | "all";
+    direction: MediaDirection | "all";
     constructor(props: Pick<RTCRtpCodecParameters, "mimeType" | "clockRate"> & Partial<RTCRtpCodecParameters>);
     get name(): string;
     get contentType(): string;

package/lib/webrtc/src/media/parameters.js.map

@@ -1 +1 @@
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+{"version":3,"file":"parameters.js","sourceRoot":"","sources":["../../../../src/media/parameters.ts"],"names":[],"mappings":";;;AAaA,MAAa,qBAAqB;IAahC,YACE,KACgC;QAdlC;;;WAGG;QACH;;;;;WAAqB;QACrB;;;;;WAAkB;QAClB;;;;;WAAmB;QACnB;;;;;WAAkB;QAClB;;;;mBAAyB,EAAE;WAAC;QAC5B;;;;;WAAoB;QACpB;;;;mBAAoC,KAAK;WAAC;QAMxC,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;IAC7B,CAAC;IAED,IAAI,IAAI;QACN,OAAO,IAAI,CAAC,QAAQ,CAAC,KAAK,CAAC,GAAG,CAAC,CAAC,CAAC,CAAC,CAAC;IACrC,CAAC;IAED,IAAI,WAAW;QACb,OAAO,IAAI,CAAC,QAAQ,CAAC,KAAK,CAAC,GAAG,CAAC,CAAC,CAAC,CAAC,CAAC;IACrC,CAAC;IAED,IAAI,GAAG;QACL,IAAI,CAAC,GAAG,GAAG,IAAI,CAAC,IAAI,IAAI,IAAI,CAAC,SAAS,EAAE,CAAC;QACzC,IAAI,IAAI,CAAC,QAAQ,KAAK,CAAC;YAAE,CAAC,IAAI,IAAI,CAAC;QACnC,OAAO,CAAC,CAAC;IACX,CAAC;CACF;AAjCD,sDAiCC;AAED,MAAa,+BAA+B;IAI1C,YACE,KAC8C;QALhD;;;;;WAAY;QACZ;;;;;WAAa;QAMX,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;IAC7B,CAAC;CACF;AAVD,0EAUC;AAED,MAAa,iBAAiB;IAK5B,YAAY,QAAoC,EAAE;QAJlD;;;;;WAAe;QACf;;;;mBAAM,KAAK;WAAC;QACZ;;;;;WAAc;QAGZ,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;IAC7B,CAAC;CACF;AARD,8CAQC;AAED,MAAa,eAAe;IAI1B,YAAY,QAAkC,EAAE;QAHhD;;;;;WAAc;QACd;;;;;WAAmB;QAGjB,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;IAC7B,CAAC;CACF;AAPD,0CAOC;AACD,MAAa,mBAAmB;IAG9B,YAAY,QAAsC,EAAE;QAFpD;;;;;WAAc;QAGZ,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;IAC7B,CAAC;CACF;AAND,kDAMC;AAED,MAAa,sBAAsB;IAKjC,YACE,KACsD;QANxD;;;;;WAAc;QACd;;;;;WAAqB;QACrB;;;;;WAA0B;QAMxB,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;IAC7B,CAAC;CACF;AAXD,wDAWC;AAQD,MAAa,yBAAyB;IAGpC,YAAY,KAAgC;QAF5C;;;;;WAAa;QACb;;;;;WAA4B;QAE1B,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;IAC7B,CAAC;CACF;AAND,8DAMC","sourcesContent":["import type { MediaDirection } from \"./rtpTransceiver\";\n\nexport interface RTCRtpParameters {\n  codecs: RTCRtpCodecParameters[];\n  headerExtensions: RTCRtpHeaderExtensionParameters[];\n  muxId?: string;\n  rtpStreamId?: string;\n  repairedRtpStreamId?: string;\n  rtcp?: RTCRtcpParameters;\n}\n\nexport type RTCPFB = { type: string; parameter?: string };\n\nexport class RTCRtpCodecParameters {\n  /**\n   * When specifying a codec with a fixed payloadType such as PCMU,\n   * it is necessary to set the correct PayloadType in RTCRtpCodecParameters in advance.\n   */\n  payloadType!: number;\n  mimeType!: string;\n  clockRate!: number;\n  channels?: number;\n  rtcpFeedback: RTCPFB[] = [];\n  parameters?: string;\n  direction: MediaDirection | \"all\" = \"all\";\n\n  constructor(\n    props: Pick<RTCRtpCodecParameters, \"mimeType\" | \"clockRate\"> &\n      Partial<RTCRtpCodecParameters>,\n  ) {\n    Object.assign(this, props);\n  }\n\n  get name() {\n    return this.mimeType.split(\"/\")[1];\n  }\n\n  get contentType() {\n    return this.mimeType.split(\"/\")[0];\n  }\n\n  get str() {\n    let s = `${this.name}/${this.clockRate}`;\n    if (this.channels === 2) s += \"/2\";\n    return s;\n  }\n}\n\nexport class RTCRtpHeaderExtensionParameters {\n  id!: number;\n  uri!: string;\n\n  constructor(\n    props: Partial<RTCRtpHeaderExtensionParameters> &\n      Pick<RTCRtpHeaderExtensionParameters, \"uri\">,\n  ) {\n    Object.assign(this, props);\n  }\n}\n\nexport class RTCRtcpParameters {\n  cname?: string;\n  mux = false;\n  ssrc?: number;\n\n  constructor(props: Partial<RTCRtcpParameters> = {}) {\n    Object.assign(this, props);\n  }\n}\n\nexport class RTCRtcpFeedback {\n  type!: string;\n  parameter?: string;\n\n  constructor(props: Partial<RTCRtcpFeedback> = {}) {\n    Object.assign(this, props);\n  }\n}\nexport class RTCRtpRtxParameters {\n  ssrc!: number;\n\n  constructor(props: Partial<RTCRtpRtxParameters> = {}) {\n    Object.assign(this, props);\n  }\n}\n\nexport class RTCRtpCodingParameters {\n  ssrc!: number;\n  payloadType!: number;\n  rtx?: RTCRtpRtxParameters;\n\n  constructor(\n    props: Partial<RTCRtpCodingParameters> &\n      Pick<RTCRtpCodingParameters, \"ssrc\" | \"payloadType\">,\n  ) {\n    Object.assign(this, props);\n  }\n}\n\nexport interface RTCRtpReceiveParameters extends RTCRtpParameters {\n  encodings: RTCRtpCodingParameters[];\n}\n\nexport type RTCRtpSendParameters = RTCRtpParameters;\n\nexport class RTCRtpSimulcastParameters {\n  rid!: string;\n  direction!: \"send\" | \"recv\";\n  constructor(props: RTCRtpSimulcastParameters) {\n    Object.assign(this, props);\n  }\n}\n"]}

package/lib/webrtc/src/media/receiver/nack.js

@@ -1,10 +1,6 @@
 "use strict";
-var __importDefault = (this && this.__importDefault) || function (mod) {
-    return (mod && mod.__esModule) ? mod : { "default": mod };
-};
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.NackHandler = void 0;
-const range_js_1 = __importDefault(require("lodash/range.js"));
 const common_1 = require("../../imports/common");
 const common_2 = require("../../imports/common");
 const rtp_1 = require("../../imports/rtp");
@@ -125,9 +121,9 @@ class NackHandler {
         }
         else if (sequenceNumber > (0, common_2.uint16Add)(this.newEstSeqNum, 1)) {
             // packet lost detected
-            (0, range_js_1.default)((0, common_2.uint16Add)(this.newEstSeqNum, 1), sequenceNumber).forEach((seq) => {
-                this.setLost(seq, 1);
-            });
+            for (let i = (0, common_2.uint16Add)(this.newEstSeqNum, 1); i < sequenceNumber; i = (0, common_2.uint16Add)(i, 1)) {
+                this.setLost(i, 1);
+            }
             // this.receiver.sendRtcpPLI(this.mediaSourceSsrc);
             this.newEstSeqNum = sequenceNumber;
             this.pruneLost();

package/lib/webrtc/src/media/receiver/nack.js.map

@@ -1 +1 @@
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range from \"lodash/range.js\";\nimport { Event } from \"../../imports/common\";\n\nimport { uint16Add } from \"../../imports/common\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n  debug,\n} from \"../../imports/rtp\";\nimport type { RTCRtpReceiver } from \"../rtpReceiver\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/receiver/nack.ts\");\n\nconst LOST_SIZE = 30 * 5;\n\nexport class NackHandler {\n  private newEstSeqNum = 0;\n  private _lost: { [seqNum: number]: number } = {};\n  private nackLoop: any;\n\n  readonly onPacketLost = new Event<[GenericNack]>();\n  mediaSourceSsrc?: number;\n  retryCount = 10;\n  closed = false;\n\n  constructor(private receiver: RTCRtpReceiver) {}\n\n  get lostSeqNumbers() {\n    return Object.keys(this._lost).map(Number).sort();\n  }\n\n  private getLost(seq: number) {\n    return this._lost[seq];\n  }\n\n  private setLost(seq: number, count: number) {\n    this._lost[seq] = count;\n\n    if (this.nackLoop || this.closed) {\n      return;\n    }\n    this.nackLoop = setInterval(async () => {\n      try {\n        await this.sendNack();\n        if (!Object.keys(this._lost).length) {\n          clearInterval(this.nackLoop);\n          this.nackLoop = undefined;\n        }\n      } catch (error) {\n        log(\"failed to send nack\", error);\n      }\n    }, 5);\n  }\n\n  private removeLost(sequenceNumber: number) {\n    delete this._lost[sequenceNumber];\n  }\n\n  addPacket(packet: RtpPacket) {\n    const { sequenceNumber, ssrc } = packet.header;\n    this.mediaSourceSsrc = ssrc;\n\n    if (this.newEstSeqNum === 0) {\n      this.newEstSeqNum = sequenceNumber;\n      return;\n    }\n\n    if (this.getLost(sequenceNumber)) {\n      // log(\"packetLoss resolved\", { sequenceNumber });\n      this.removeLost(sequenceNumber);\n      return;\n    }\n\n    if (sequenceNumber === uint16Add(this.newEstSeqNum, 1)) {\n      this.newEstSeqNum = sequenceNumber;\n    } else if (sequenceNumber > uint16Add(this.newEstSeqNum, 1)) {\n      // packet lost detected\n      range(uint16Add(this.newEstSeqNum, 1), sequenceNumber).forEach((seq) => {\n        this.setLost(seq, 1);\n      });\n      // this.receiver.sendRtcpPLI(this.mediaSourceSsrc);\n\n      this.newEstSeqNum = sequenceNumber;\n      this.pruneLost();\n    }\n  }\n\n  private pruneLost() {\n    if (this.lostSeqNumbers.length > LOST_SIZE) {\n      this._lost = Object.entries(this._lost)\n        .slice(-LOST_SIZE)\n        .reduce(\n          (acc, [key, v]) => {\n            acc[key] = v;\n            return acc;\n          },\n          {} as { [seqNum: number]: number },\n        );\n    }\n  }\n\n  close() {\n    this.closed = true;\n    clearInterval(this.nackLoop);\n    this._lost = {};\n  }\n\n  private updateRetryCount() {\n    this.lostSeqNumbers.forEach((seq) => {\n      const count = this._lost[seq]++;\n      if (count > this.retryCount) {\n        this.removeLost(seq);\n        return seq;\n      }\n    });\n  }\n\n  private sendNack = () =>\n    new Promise((r, f) => {\n      if (this.lostSeqNumbers.length > 0 && this.mediaSourceSsrc) {\n        const nack = new GenericNack({\n          senderSsrc: this.receiver.rtcpSsrc,\n          mediaSourceSsrc: this.mediaSourceSsrc,\n          lost: this.lostSeqNumbers,\n        });\n\n        const rtcp = new RtcpTransportLayerFeedback({\n          feedback: nack,\n        });\n        this.receiver.dtlsTransport.sendRtcp([rtcp]).then(r).catch(f);\n\n        this.updateRetryCount();\n        this.onPacketLost.execute(nack);\n      }\n    });\n}\n"]}
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{ Event } from \"../../imports/common\";\n\nimport { uint16Add } from \"../../imports/common\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n  debug,\n} from \"../../imports/rtp\";\nimport type { RTCRtpReceiver } from \"../rtpReceiver\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/receiver/nack.ts\");\n\nconst LOST_SIZE = 30 * 5;\n\nexport class NackHandler {\n  private newEstSeqNum = 0;\n  private _lost: { [seqNum: number]: number } = {};\n  private nackLoop: any;\n\n  readonly onPacketLost = new Event<[GenericNack]>();\n  mediaSourceSsrc?: number;\n  retryCount = 10;\n  closed = false;\n\n  constructor(private receiver: RTCRtpReceiver) {}\n\n  get lostSeqNumbers() {\n    return Object.keys(this._lost).map(Number).sort();\n  }\n\n  private getLost(seq: number) {\n    return this._lost[seq];\n  }\n\n  private setLost(seq: number, count: number) {\n    this._lost[seq] = count;\n\n    if (this.nackLoop || this.closed) {\n      return;\n    }\n    this.nackLoop = setInterval(async () => {\n      try {\n        await this.sendNack();\n        if (!Object.keys(this._lost).length) {\n          clearInterval(this.nackLoop);\n          this.nackLoop = undefined;\n        }\n      } catch (error) {\n        log(\"failed to send nack\", error);\n      }\n    }, 5);\n  }\n\n  private removeLost(sequenceNumber: number) {\n    delete this._lost[sequenceNumber];\n  }\n\n  addPacket(packet: RtpPacket) {\n    const { sequenceNumber, ssrc } = packet.header;\n    this.mediaSourceSsrc = ssrc;\n\n    if (this.newEstSeqNum === 0) {\n      this.newEstSeqNum = sequenceNumber;\n      return;\n    }\n\n    if (this.getLost(sequenceNumber)) {\n      // log(\"packetLoss resolved\", { sequenceNumber });\n      this.removeLost(sequenceNumber);\n      return;\n    }\n\n    if (sequenceNumber === uint16Add(this.newEstSeqNum, 1)) {\n      this.newEstSeqNum = sequenceNumber;\n    } else if (sequenceNumber > uint16Add(this.newEstSeqNum, 1)) {\n      // packet lost detected\n      for (\n        let i = uint16Add(this.newEstSeqNum, 1);\n        i < sequenceNumber;\n        i = uint16Add(i, 1)\n      ) {\n        this.setLost(i, 1);\n      }\n      // this.receiver.sendRtcpPLI(this.mediaSourceSsrc);\n\n      this.newEstSeqNum = sequenceNumber;\n      this.pruneLost();\n    }\n  }\n\n  private pruneLost() {\n    if (this.lostSeqNumbers.length > LOST_SIZE) {\n      this._lost = Object.entries(this._lost)\n        .slice(-LOST_SIZE)\n        .reduce(\n          (acc, [key, v]) => {\n            acc[key] = v;\n            return acc;\n          },\n          {} as { [seqNum: number]: number },\n        );\n    }\n  }\n\n  close() {\n    this.closed = true;\n    clearInterval(this.nackLoop);\n    this._lost = {};\n  }\n\n  private updateRetryCount() {\n    this.lostSeqNumbers.forEach((seq) => {\n      const count = this._lost[seq]++;\n      if (count > this.retryCount) {\n        this.removeLost(seq);\n        return seq;\n      }\n    });\n  }\n\n  private sendNack = () =>\n    new Promise((r, f) => {\n      if (this.lostSeqNumbers.length > 0 && this.mediaSourceSsrc) {\n        const nack = new GenericNack({\n          senderSsrc: this.receiver.rtcpSsrc,\n          mediaSourceSsrc: this.mediaSourceSsrc,\n          lost: this.lostSeqNumbers,\n        });\n\n        const rtcp = new RtcpTransportLayerFeedback({\n          feedback: nack,\n        });\n        this.receiver.dtlsTransport.sendRtcp([rtcp]).then(r).catch(f);\n\n        this.updateRetryCount();\n        this.onPacketLost.execute(nack);\n      }\n    });\n}\n"]}

package/lib/webrtc/src/media/receiver/receiverTwcc.d.ts

@@ -1,7 +1,7 @@
 import type { RTCDtlsTransport } from "../../transport/dtls";
 type ExtensionInfo = {
     tsn: number;
-    timestamp: number;
+    timestamp: bigint;
 };
 export declare class ReceiverTWCC {
     private dtlsTransport;
@@ -13,7 +13,7 @@ export declare class ReceiverTWCC {
     twccRunning: boolean;
     /** uint8 */
     fbPktCount: number;
-    lastTimestamp?: number;
+    lastTimestamp?: bigint;
     constructor(dtlsTransport: RTCDtlsTransport, rtcpSsrc: number, mediaSourceSsrc: number);
     handleTWCC(transportSequenceNumber: number): void;
     private runTWCC;

package/lib/webrtc/src/media/receiver/receiverTwcc.js

@@ -136,7 +136,7 @@ class ReceiverTWCC {
                 mediaSourceSsrc: this.mediaSourceSsrc,
                 baseSequenceNumber,
                 packetStatusCount,
-                referenceTime: (0, common_1.uint24)(Math.floor(referenceTime / 1000 / 64)),
+                referenceTime: (0, common_1.uint24)(Math.floor(Number(referenceTime / 1000n / 64n))),
                 fbPktCount: this.fbPktCount,
                 recvDeltas,
                 packetChunks,

package/lib/webrtc/src/media/receiver/receiverTwcc.js.map

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timestamp: number };\n\nexport class ReceiverTWCC {\n  extensionInfo: {\n    [tsn: number]: ExtensionInfo;\n  } = {};\n  twccRunning = false;\n  /** uint8 */\n  fbPktCount = 0;\n  lastTimestamp?: number;\n\n  constructor(\n    private dtlsTransport: RTCDtlsTransport,\n    private rtcpSsrc: number,\n    private mediaSourceSsrc: number,\n  ) {\n    this.runTWCC();\n  }\n\n  handleTWCC(transportSequenceNumber: number) {\n    this.extensionInfo[transportSequenceNumber] = {\n      tsn: transportSequenceNumber,\n      timestamp: microTime(),\n    };\n\n    if (Object.keys(this.extensionInfo).length > 10) {\n      this.sendTWCC();\n    }\n  }\n\n  private async runTWCC() {\n    while (this.twccRunning) {\n      this.sendTWCC();\n      await setTimeout(100);\n    }\n  }\n\n  private sendTWCC() {\n    if (Object.keys(this.extensionInfo).length === 0) return;\n    const extensionsArr = Object.values(this.extensionInfo).sort(\n      (a, b) => a.tsn - b.tsn,\n    );\n\n    const minTSN = extensionsArr[0].tsn;\n    const maxTSN = extensionsArr.slice(-1)[0].tsn;\n\n    const packetChunks: (RunLengthChunk | StatusVectorChunk)[] = [];\n    const baseSequenceNumber = extensionsArr[0].tsn;\n    const packetStatusCount = uint16Add(maxTSN - minTSN, 1);\n    /**micro sec */\n    let referenceTime!: number;\n    let lastPacketStatus: { status: PacketStatus; minTSN: number } | undefined;\n    const recvDeltas: RecvDelta[] = [];\n\n    for (let i = minTSN; i <= maxTSN; i++) {\n      /**micro sec */\n      const timestamp = this.extensionInfo[i]?.timestamp;\n\n      if (timestamp) {\n        if (!this.lastTimestamp) {\n          this.lastTimestamp = timestamp;\n        }\n        if (!referenceTime) {\n          referenceTime = this.lastTimestamp;\n        }\n\n        const delta = timestamp - this.lastTimestamp;\n        this.lastTimestamp = timestamp;\n\n        const recvDelta = new RecvDelta({\n          delta: Number(delta),\n        });\n        recvDelta.parseDelta();\n        recvDeltas.push(recvDelta);\n\n        // when status changed\n        if (\n          lastPacketStatus != undefined &&\n          lastPacketStatus.status !== recvDelta.type\n        ) {\n          packetChunks.push(\n            new RunLengthChunk({\n              packetStatus: lastPacketStatus.status,\n              runLength: i - lastPacketStatus.minTSN,\n            }),\n          );\n          lastPacketStatus = { minTSN: i, status: recvDelta.type! };\n        }\n        // last status\n        if (i === maxTSN) {\n          if (lastPacketStatus != undefined) {\n            packetChunks.push(\n              new RunLengthChunk({\n                packetStatus: lastPacketStatus.status,\n                runLength: i - lastPacketStatus.minTSN + 1,\n              }),\n            );\n          } else {\n            packetChunks.push(\n              new RunLengthChunk({\n                packetStatus: recvDelta.type,\n                runLength: 1,\n              }),\n            );\n          }\n        }\n\n        if (lastPacketStatus == undefined) {\n          lastPacketStatus = { minTSN: i, status: recvDelta.type! };\n        }\n      }\n    }\n\n    if (!referenceTime) {\n      return;\n    }\n\n    const packet = new RtcpTransportLayerFeedback({\n      feedback: new TransportWideCC({\n        senderSsrc: this.rtcpSsrc,\n        mediaSourceSsrc: this.mediaSourceSsrc,\n        baseSequenceNumber,\n        packetStatusCount,\n        referenceTime: uint24(Math.floor(referenceTime / 1000 / 64)),\n        fbPktCount: this.fbPktCount,\n        recvDeltas,\n        packetChunks,\n      }),\n    });\n\n    this.dtlsTransport.sendRtcp([packet]).catch((err) => {\n      log(err);\n    });\n    this.extensionInfo = {};\n    this.fbPktCount = uint8Add(this.fbPktCount, 1);\n  }\n}\n"]}
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timestamp: bigint };\n\nexport class ReceiverTWCC {\n  extensionInfo: {\n    [tsn: number]: ExtensionInfo;\n  } = {};\n  twccRunning = false;\n  /** uint8 */\n  fbPktCount = 0;\n  lastTimestamp?: bigint;\n\n  constructor(\n    private dtlsTransport: RTCDtlsTransport,\n    private rtcpSsrc: number,\n    private mediaSourceSsrc: number,\n  ) {\n    this.runTWCC();\n  }\n\n  handleTWCC(transportSequenceNumber: number) {\n    this.extensionInfo[transportSequenceNumber] = {\n      tsn: transportSequenceNumber,\n      timestamp: microTime(),\n    };\n\n    if (Object.keys(this.extensionInfo).length > 10) {\n      this.sendTWCC();\n    }\n  }\n\n  private async runTWCC() {\n    while (this.twccRunning) {\n      this.sendTWCC();\n      await setTimeout(100);\n    }\n  }\n\n  private sendTWCC() {\n    if (Object.keys(this.extensionInfo).length === 0) return;\n    const extensionsArr = Object.values(this.extensionInfo).sort(\n      (a, b) => a.tsn - b.tsn,\n    );\n\n    const minTSN = extensionsArr[0].tsn;\n    const maxTSN = extensionsArr.slice(-1)[0].tsn;\n\n    const packetChunks: (RunLengthChunk | StatusVectorChunk)[] = [];\n    const baseSequenceNumber = extensionsArr[0].tsn;\n    const packetStatusCount = uint16Add(maxTSN - minTSN, 1);\n    /**micro sec */\n    let referenceTime!: bigint;\n    let lastPacketStatus: { status: PacketStatus; minTSN: number } | undefined;\n    const recvDeltas: RecvDelta[] = [];\n\n    for (let i = minTSN; i <= maxTSN; i++) {\n      /**micro sec */\n      const timestamp = this.extensionInfo[i]?.timestamp;\n\n      if (timestamp) {\n        if (!this.lastTimestamp) {\n          this.lastTimestamp = timestamp;\n        }\n        if (!referenceTime) {\n          referenceTime = this.lastTimestamp;\n        }\n\n        const delta = timestamp - this.lastTimestamp;\n        this.lastTimestamp = timestamp;\n\n        const recvDelta = new RecvDelta({\n          delta: Number(delta),\n        });\n        recvDelta.parseDelta();\n        recvDeltas.push(recvDelta);\n\n        // when status changed\n        if (\n          lastPacketStatus != undefined &&\n          lastPacketStatus.status !== recvDelta.type\n        ) {\n          packetChunks.push(\n            new RunLengthChunk({\n              packetStatus: lastPacketStatus.status,\n              runLength: i - lastPacketStatus.minTSN,\n            }),\n          );\n          lastPacketStatus = { minTSN: i, status: recvDelta.type! };\n        }\n        // last status\n        if (i === maxTSN) {\n          if (lastPacketStatus != undefined) {\n            packetChunks.push(\n              new RunLengthChunk({\n                packetStatus: lastPacketStatus.status,\n                runLength: i - lastPacketStatus.minTSN + 1,\n              }),\n            );\n          } else {\n            packetChunks.push(\n              new RunLengthChunk({\n                packetStatus: recvDelta.type,\n                runLength: 1,\n              }),\n            );\n          }\n        }\n\n        if (lastPacketStatus == undefined) {\n          lastPacketStatus = { minTSN: i, status: recvDelta.type! };\n        }\n      }\n    }\n\n    if (!referenceTime) {\n      return;\n    }\n\n    const packet = new RtcpTransportLayerFeedback({\n      feedback: new TransportWideCC({\n        senderSsrc: this.rtcpSsrc,\n        mediaSourceSsrc: this.mediaSourceSsrc,\n        baseSequenceNumber,\n        packetStatusCount,\n        referenceTime: uint24(Math.floor(Number(referenceTime / 1000n / 64n))),\n        fbPktCount: this.fbPktCount,\n        recvDeltas,\n        packetChunks,\n      }),\n    });\n\n    this.dtlsTransport.sendRtcp([packet]).catch((err) => {\n      log(err);\n    });\n    this.extensionInfo = {};\n    this.fbPktCount = uint8Add(this.fbPktCount, 1);\n  }\n}\n"]}

package/lib/webrtc/src/media/router.js

@@ -115,7 +115,8 @@ class RtpRouter {
                                     }
                                     break;
                                 default:
-                                    recipients.push(this.ssrcTable[psfb.feedback.senderSsrc]);
+                                    recipients.push(this.ssrcTable[psfb.feedback.senderSsrc] ||
+                                        this.ssrcTable[psfb.feedback.mediaSsrc]);
                             }
                         }
                         break;

package/lib/webrtc/src/media/router.js.map

@@ -1 +1 @@
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{\n  type Extensions,\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n  debug,\n  rtpHeaderExtensionsParser,\n} from \"../imports/rtp\";\nimport type {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport type { RTCRtpSender } from \"./rtpSender\";\nimport type { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport class RtpRouter {\n  ssrcTable: { [ssrc: number]: RTCRtpReceiver | RTCRtpSender } = {};\n  ridTable: { [rid: string]: RTCRtpReceiver | RTCRtpSender } = {};\n  extIdUriMap: { [id: number]: string } = {};\n\n  constructor() {}\n\n  registerRtpSender(sender: RTCRtpSender) {\n    this.ssrcTable[sender.ssrc] = sender;\n  }\n\n  private registerRtpReceiver(receiver: RTCRtpReceiver, ssrc: number) {\n    log(\"registerRtpReceiver\", ssrc);\n    this.ssrcTable[ssrc] = receiver;\n  }\n\n  registerRtpReceiverBySsrc(\n    transceiver: RTCRtpTransceiver,\n    params: RTCRtpReceiveParameters,\n  ) {\n    log(\"registerRtpReceiverBySsrc\", params);\n\n    params.encodings\n      .filter((e) => e.ssrc != undefined) // todo fix\n      .forEach((encode, i) => {\n        this.registerRtpReceiver(transceiver.receiver, encode.ssrc);\n        transceiver.addTrack(\n          new MediaStreamTrack({\n            ssrc: encode.ssrc,\n            kind: transceiver.kind,\n            id: transceiver.sender.trackId,\n            remote: true,\n            codec: params.codecs[i],\n          }),\n        );\n        if (encode.rtx) {\n          this.registerRtpReceiver(transceiver.receiver, encode.rtx.ssrc);\n        }\n      });\n\n    params.headerExtensions.forEach((extension) => {\n      this.extIdUriMap[extension.id] = extension.uri;\n    });\n  }\n\n  registerRtpReceiverByRid(\n    transceiver: RTCRtpTransceiver,\n    param: RTCRtpSimulcastParameters,\n    params: RTCRtpReceiveParameters,\n  ) {\n    // サイマルキャスト利用時のRTXをサポートしていないのでcodecs/encodingsは常に一つ\n    const [codec] = params.codecs;\n\n    log(\"registerRtpReceiverByRid\", param);\n    transceiver.addTrack(\n      new MediaStreamTrack({\n        rid: param.rid,\n        kind: transceiver.kind,\n        id: transceiver.sender.trackId,\n        remote: true,\n        codec,\n      }),\n    );\n    this.ridTable[param.rid] = transceiver.receiver;\n  }\n\n  routeRtp = (packet: RtpPacket) => {\n    const extensions: Extensions = rtpHeaderExtensionsParser(\n      packet.header.extensions,\n      this.extIdUriMap,\n    );\n\n    let rtpReceiver: RTCRtpReceiver | undefined = this.ssrcTable[\n      packet.header.ssrc\n    ] as RTCRtpReceiver;\n\n    const rid = extensions[RTP_EXTENSION_URI.sdesRTPStreamID];\n    if (typeof rid === \"string\") {\n      rtpReceiver = this.ridTable[rid] as RTCRtpReceiver;\n      rtpReceiver.latestRid = rid;\n      rtpReceiver.handleRtpByRid(packet, rid, extensions);\n    } else if (rtpReceiver) {\n      rtpReceiver.handleRtpBySsrc(packet, extensions);\n    } else {\n      // simulcast after send receiver report\n      rtpReceiver = Object.values(this.ridTable)\n        .filter((r): r is RTCRtpReceiver => r instanceof RTCRtpReceiver)\n        .find((r) => r.trackBySSRC[packet.header.ssrc]);\n      if (rtpReceiver) {\n        log(\"simulcast register receiver by ssrc\", packet.header.ssrc);\n        this.registerRtpReceiver(rtpReceiver, packet.header.ssrc);\n        rtpReceiver.handleRtpBySsrc(packet, extensions);\n      } else {\n        // bug\n      }\n    }\n\n    if (!rtpReceiver) {\n      log(\"ssrcReceiver not found\");\n      return;\n    }\n\n    const sdesMid = extensions[RTP_EXTENSION_URI.sdesMid];\n    if (typeof sdesMid === \"string\") {\n      rtpReceiver.sdesMid = sdesMid;\n    }\n\n    const repairedRid = extensions[\n      RTP_EXTENSION_URI.repairedRtpStreamId\n    ] as string;\n    if (typeof repairedRid === \"string\") {\n      rtpReceiver.latestRepairedRid = repairedRid;\n    }\n  };\n\n  routeRtcp = (packet: RtcpPacket) => {\n    const recipients: (RTCRtpReceiver | RTCRtpSender)[] = [];\n\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          packet = packet as RtcpSrPacket;\n          recipients.push(this.ssrcTable[packet.ssrc]);\n        }\n        break;\n      case RtcpRrPacket.type:\n        {\n          packet = packet as RtcpRrPacket;\n          packet.reports.forEach((report) => {\n            recipients.push(this.ssrcTable[report.ssrc]);\n          });\n        }\n        break;\n      case RtcpSourceDescriptionPacket.type:\n        {\n          const sdes = packet as RtcpSourceDescriptionPacket;\n          // log(\"sdes\", JSON.stringify(sdes.chunks));\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const rtpfb = packet as RtcpTransportLayerFeedback;\n          if (rtpfb.feedback) {\n            recipients.push(this.ssrcTable[rtpfb.feedback.mediaSourceSsrc]);\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const psfb = packet as RtcpPayloadSpecificFeedback;\n          switch (psfb.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              {\n                const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;\n                recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);\n              }\n              break;\n            default:\n              recipients.push(this.ssrcTable[psfb.feedback.senderSsrc]);\n          }\n        }\n        break;\n    }\n    recipients\n      .filter((v) => v) // todo simulcast\n      .forEach((recipient) => recipient.handleRtcpPacket(packet));\n  };\n}\n"]}
+{"version":3,"file":"router.js","sourceRoot":"","sources":["../../../../src/media/router.ts"],"names":[],"mappings":";;;AAAA,wCAawB;AAKxB,+CAA+C;AAG/C,mCAA2C;AAE3C,MAAM,GAAG,GAAG,IAAA,WAAK,EAAC,4CAA4C,CAAC,CAAC;AAEhE,MAAa,SAAS;IAKpB;QAJA;;;;mBAA+D,EAAE;WAAC;QAClE;;;;mBAA6D,EAAE;WAAC;QAChE;;;;mBAAwC,EAAE;WAAC;QA+D3C;;;;mBAAW,CAAC,MAAiB,EAAE,EAAE;gBAC/B,MAAM,UAAU,GAAe,IAAA,+BAAyB,EACtD,MAAM,CAAC,MAAM,CAAC,UAAU,EACxB,IAAI,CAAC,WAAW,CACjB,CAAC;gBAEF,IAAI,WAAW,GAA+B,IAAI,CAAC,SAAS,CAC1D,MAAM,CAAC,MAAM,CAAC,IAAI,CACD,CAAC;gBAEpB,MAAM,GAAG,GAAG,UAAU,CAAC,uBAAiB,CAAC,eAAe,CAAC,CAAC;gBAC1D,IAAI,OAAO,GAAG,KAAK,QAAQ,EAAE,CAAC;oBAC5B,WAAW,GAAG,IAAI,CAAC,QAAQ,CAAC,GAAG,CAAmB,CAAC;oBACnD,WAAW,CAAC,SAAS,GAAG,GAAG,CAAC;oBAC5B,WAAW,CAAC,cAAc,CAAC,MAAM,EAAE,GAAG,EAAE,UAAU,CAAC,CAAC;gBACtD,CAAC;qBAAM,IAAI,WAAW,EAAE,CAAC;oBACvB,WAAW,CAAC,eAAe,CAAC,MAAM,EAAE,UAAU,CAAC,CAAC;gBAClD,CAAC;qBAAM,CAAC;oBACN,uCAAuC;oBACvC,WAAW,GAAG,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,QAAQ,CAAC;yBACvC,MAAM,CAAC,CAAC,CAAC,EAAuB,EAAE,CAAC,CAAC,YAAY,4BAAc,CAAC;yBAC/D,IAAI,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,WAAW,CAAC,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC,CAAC;oBAClD,IAAI,WAAW,EAAE,CAAC;wBAChB,GAAG,CAAC,qCAAqC,EAAE,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC;wBAC/D,IAAI,CAAC,mBAAmB,CAAC,WAAW,EAAE,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC;wBAC1D,WAAW,CAAC,eAAe,CAAC,MAAM,EAAE,UAAU,CAAC,CAAC;oBAClD,CAAC;yBAAM,CAAC;wBACN,MAAM;oBACR,CAAC;gBACH,CAAC;gBAED,IAAI,CAAC,WAAW,EAAE,CAAC;oBACjB,GAAG,CAAC,wBAAwB,CAAC,CAAC;oBAC9B,OAAO;gBACT,CAAC;gBAED,MAAM,OAAO,GAAG,UAAU,CAAC,uBAAiB,CAAC,OAAO,CAAC,CAAC;gBACtD,IAAI,OAAO,OAAO,KAAK,QAAQ,EAAE,CAAC;oBAChC,WAAW,CAAC,OAAO,GAAG,OAAO,CAAC;gBAChC,CAAC;gBAED,MAAM,WAAW,GAAG,UAAU,CAC5B,uBAAiB,CAAC,mBAAmB,CAC5B,CAAC;gBACZ,IAAI,OAAO,WAAW,KAAK,QAAQ,EAAE,CAAC;oBACpC,WAAW,CAAC,iBAAiB,GAAG,WAAW,CAAC;gBAC9C,CAAC;YACH,CAAC;WAAC;QAEF;;;;mBAAY,CAAC,MAAkB,EAAE,EAAE;gBACjC,MAAM,UAAU,GAAsC,EAAE,CAAC;gBAEzD,QAAQ,MAAM,CAAC,IAAI,EAAE,CAAC;oBACpB,KAAK,kBAAY,CAAC,IAAI;wBACpB,CAAC;4BACC,MAAM,GAAG,MAAsB,CAAC;4BAChC,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,SAAS,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC,CAAC;wBAC/C,CAAC;wBACD,MAAM;oBACR,KAAK,kBAAY,CAAC,IAAI;wBACpB,CAAC;4BACC,MAAM,GAAG,MAAsB,CAAC;4BAChC,MAAM,CAAC,OAAO,CAAC,OAAO,CAAC,CAAC,MAAM,EAAE,EAAE;gCAChC,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,SAAS,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC,CAAC;4BAC/C,CAAC,CAAC,CAAC;wBACL,CAAC;wBACD,MAAM;oBACR,KAAK,iCAA2B,CAAC,IAAI;wBACnC,CAAC;4BACC,MAAM,IAAI,GAAG,MAAqC,CAAC;4BACnD,4CAA4C;wBAC9C,CAAC;wBACD,MAAM;oBACR,KAAK,gCAA0B,CAAC,IAAI;wBAClC,CAAC;4BACC,MAAM,KAAK,GAAG,MAAoC,CAAC;4BACnD,IAAI,KAAK,CAAC,QAAQ,EAAE,CAAC;gCACnB,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,SAAS,CAAC,KAAK,CAAC,QAAQ,CAAC,eAAe,CAAC,CAAC,CAAC;4BAClE,CAAC;wBACH,CAAC;wBACD,MAAM;oBACR,KAAK,iCAA2B,CAAC,IAAI;wBACnC,CAAC;4BACC,MAAM,IAAI,GAAG,MAAqC,CAAC;4BACnD,QAAQ,IAAI,CAAC,QAAQ,CAAC,KAAK,EAAE,CAAC;gCAC5B,KAAK,iCAA2B,CAAC,KAAK;oCACpC,CAAC;wCACC,MAAM,IAAI,GAAG,IAAI,CAAC,QAAuC,CAAC;wCAC1D,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,SAAS,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;oCACzD,CAAC;oCACD,MAAM;gCACR;oCACE,UAAU,CAAC,IAAI,CACb,IAAI,CAAC,SAAS,CAAC,IAAI,CAAC,QAAQ,CAAC,UAAU,CAAC;wCACtC,IAAI,CAAC,SAAS,CAAC,IAAI,CAAC,QAAQ,CAAC,SAAS,CAAC,CAC1C,CAAC;4BACN,CAAC;wBACH,CAAC;wBACD,MAAM;gBACV,CAAC;gBACD,UAAU;qBACP,MAAM,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC,iBAAiB;qBAClC,OAAO,CAAC,CAAC,SAAS,EAAE,EAAE,CAAC,SAAS,CAAC,gBAAgB,CAAC,MAAM,CAAC,CAAC,CAAC;YAChE,CAAC;WAAC;IApKa,CAAC;IAEhB,iBAAiB,CAAC,MAAoB;QACpC,IAAI,CAAC,SAAS,CAAC,MAAM,CAAC,IAAI,CAAC,GAAG,MAAM,CAAC;IACvC,CAAC;IAEO,mBAAmB,CAAC,QAAwB,EAAE,IAAY;QAChE,GAAG,CAAC,qBAAqB,EAAE,IAAI,CAAC,CAAC;QACjC,IAAI,CAAC,SAAS,CAAC,IAAI,CAAC,GAAG,QAAQ,CAAC;IAClC,CAAC;IAED,yBAAyB,CACvB,WAA8B,EAC9B,MAA+B;QAE/B,GAAG,CAAC,2BAA2B,EAAE,MAAM,CAAC,CAAC;QAEzC,MAAM,CAAC,SAAS;aACb,MAAM,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,IAAI,IAAI,SAAS,CAAC,CAAC,WAAW;aAC9C,OAAO,CAAC,CAAC,MAAM,EAAE,CAAC,EAAE,EAAE;YACrB,IAAI,CAAC,mBAAmB,CAAC,WAAW,CAAC,QAAQ,EAAE,MAAM,CAAC,IAAI,CAAC,CAAC;YAC5D,WAAW,CAAC,QAAQ,CAClB,IAAI,wBAAgB,CAAC;gBACnB,IAAI,EAAE,MAAM,CAAC,IAAI;gBACjB,IAAI,EAAE,WAAW,CAAC,IAAI;gBACtB,EAAE,EAAE,WAAW,CAAC,MAAM,CAAC,OAAO;gBAC9B,MAAM,EAAE,IAAI;gBACZ,KAAK,EAAE,MAAM,CAAC,MAAM,CAAC,CAAC,CAAC;aACxB,CAAC,CACH,CAAC;YACF,IAAI,MAAM,CAAC,GAAG,EAAE,CAAC;gBACf,IAAI,CAAC,mBAAmB,CAAC,WAAW,CAAC,QAAQ,EAAE,MAAM,CAAC,GAAG,CAAC,IAAI,CAAC,CAAC;YAClE,CAAC;QACH,CAAC,CAAC,CAAC;QAEL,MAAM,CAAC,gBAAgB,CAAC,OAAO,CAAC,CAAC,SAAS,EAAE,EAAE;YAC5C,IAAI,CAAC,WAAW,CAAC,SAAS,CAAC,EAAE,CAAC,GAAG,SAAS,CAAC,GAAG,CAAC;QACjD,CAAC,CAAC,CAAC;IACL,CAAC;IAED,wBAAwB,CACtB,WAA8B,EAC9B,KAAgC,EAChC,MAA+B;QAE/B,mDAAmD;QACnD,MAAM,CAAC,KAAK,CAAC,GAAG,MAAM,CAAC,MAAM,CAAC;QAE9B,GAAG,CAAC,0BAA0B,EAAE,KAAK,CAAC,CAAC;QACvC,WAAW,CAAC,QAAQ,CAClB,IAAI,wBAAgB,CAAC;YACnB,GAAG,EAAE,KAAK,CAAC,GAAG;YACd,IAAI,EAAE,WAAW,CAAC,IAAI;YACtB,EAAE,EAAE,WAAW,CAAC,MAAM,CAAC,OAAO;YAC9B,MAAM,EAAE,IAAI;YACZ,KAAK;SACN,CAAC,CACH,CAAC;QACF,IAAI,CAAC,QAAQ,CAAC,KAAK,CAAC,GAAG,CAAC,GAAG,WAAW,CAAC,QAAQ,CAAC;IAClD,CAAC;CA0GF;AA1KD,8BA0KC","sourcesContent":["import {\n  type Extensions,\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n  debug,\n  rtpHeaderExtensionsParser,\n} from \"../imports/rtp\";\nimport type {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport type { RTCRtpSender } from \"./rtpSender\";\nimport type { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport class RtpRouter {\n  ssrcTable: { [ssrc: number]: RTCRtpReceiver | RTCRtpSender } = {};\n  ridTable: { [rid: string]: RTCRtpReceiver | RTCRtpSender } = {};\n  extIdUriMap: { [id: number]: string } = {};\n\n  constructor() {}\n\n  registerRtpSender(sender: RTCRtpSender) {\n    this.ssrcTable[sender.ssrc] = sender;\n  }\n\n  private registerRtpReceiver(receiver: RTCRtpReceiver, ssrc: number) {\n    log(\"registerRtpReceiver\", ssrc);\n    this.ssrcTable[ssrc] = receiver;\n  }\n\n  registerRtpReceiverBySsrc(\n    transceiver: RTCRtpTransceiver,\n    params: RTCRtpReceiveParameters,\n  ) {\n    log(\"registerRtpReceiverBySsrc\", params);\n\n    params.encodings\n      .filter((e) => e.ssrc != undefined) // todo fix\n      .forEach((encode, i) => {\n        this.registerRtpReceiver(transceiver.receiver, encode.ssrc);\n        transceiver.addTrack(\n          new MediaStreamTrack({\n            ssrc: encode.ssrc,\n            kind: transceiver.kind,\n            id: transceiver.sender.trackId,\n            remote: true,\n            codec: params.codecs[i],\n          }),\n        );\n        if (encode.rtx) {\n          this.registerRtpReceiver(transceiver.receiver, encode.rtx.ssrc);\n        }\n      });\n\n    params.headerExtensions.forEach((extension) => {\n      this.extIdUriMap[extension.id] = extension.uri;\n    });\n  }\n\n  registerRtpReceiverByRid(\n    transceiver: RTCRtpTransceiver,\n    param: RTCRtpSimulcastParameters,\n    params: RTCRtpReceiveParameters,\n  ) {\n    // サイマルキャスト利用時のRTXをサポートしていないのでcodecs/encodingsは常に一つ\n    const [codec] = params.codecs;\n\n    log(\"registerRtpReceiverByRid\", param);\n    transceiver.addTrack(\n      new MediaStreamTrack({\n        rid: param.rid,\n        kind: transceiver.kind,\n        id: transceiver.sender.trackId,\n        remote: true,\n        codec,\n      }),\n    );\n    this.ridTable[param.rid] = transceiver.receiver;\n  }\n\n  routeRtp = (packet: RtpPacket) => {\n    const extensions: Extensions = rtpHeaderExtensionsParser(\n      packet.header.extensions,\n      this.extIdUriMap,\n    );\n\n    let rtpReceiver: RTCRtpReceiver | undefined = this.ssrcTable[\n      packet.header.ssrc\n    ] as RTCRtpReceiver;\n\n    const rid = extensions[RTP_EXTENSION_URI.sdesRTPStreamID];\n    if (typeof rid === \"string\") {\n      rtpReceiver = this.ridTable[rid] as RTCRtpReceiver;\n      rtpReceiver.latestRid = rid;\n      rtpReceiver.handleRtpByRid(packet, rid, extensions);\n    } else if (rtpReceiver) {\n      rtpReceiver.handleRtpBySsrc(packet, extensions);\n    } else {\n      // simulcast after send receiver report\n      rtpReceiver = Object.values(this.ridTable)\n        .filter((r): r is RTCRtpReceiver => r instanceof RTCRtpReceiver)\n        .find((r) => r.trackBySSRC[packet.header.ssrc]);\n      if (rtpReceiver) {\n        log(\"simulcast register receiver by ssrc\", packet.header.ssrc);\n        this.registerRtpReceiver(rtpReceiver, packet.header.ssrc);\n        rtpReceiver.handleRtpBySsrc(packet, extensions);\n      } else {\n        // bug\n      }\n    }\n\n    if (!rtpReceiver) {\n      log(\"ssrcReceiver not found\");\n      return;\n    }\n\n    const sdesMid = extensions[RTP_EXTENSION_URI.sdesMid];\n    if (typeof sdesMid === \"string\") {\n      rtpReceiver.sdesMid = sdesMid;\n    }\n\n    const repairedRid = extensions[\n      RTP_EXTENSION_URI.repairedRtpStreamId\n    ] as string;\n    if (typeof repairedRid === \"string\") {\n      rtpReceiver.latestRepairedRid = repairedRid;\n    }\n  };\n\n  routeRtcp = (packet: RtcpPacket) => {\n    const recipients: (RTCRtpReceiver | RTCRtpSender)[] = [];\n\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          packet = packet as RtcpSrPacket;\n          recipients.push(this.ssrcTable[packet.ssrc]);\n        }\n        break;\n      case RtcpRrPacket.type:\n        {\n          packet = packet as RtcpRrPacket;\n          packet.reports.forEach((report) => {\n            recipients.push(this.ssrcTable[report.ssrc]);\n          });\n        }\n        break;\n      case RtcpSourceDescriptionPacket.type:\n        {\n          const sdes = packet as RtcpSourceDescriptionPacket;\n          // log(\"sdes\", JSON.stringify(sdes.chunks));\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const rtpfb = packet as RtcpTransportLayerFeedback;\n          if (rtpfb.feedback) {\n            recipients.push(this.ssrcTable[rtpfb.feedback.mediaSourceSsrc]);\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const psfb = packet as RtcpPayloadSpecificFeedback;\n          switch (psfb.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              {\n                const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;\n                recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);\n              }\n              break;\n            default:\n              recipients.push(\n                this.ssrcTable[psfb.feedback.senderSsrc] ||\n                  this.ssrcTable[psfb.feedback.mediaSsrc],\n              );\n          }\n        }\n        break;\n    }\n    recipients\n      .filter((v) => v) // todo simulcast\n      .forEach((recipient) => recipient.handleRtcpPacket(packet));\n  };\n}\n"]}

package/lib/webrtc/src/media/rtpReceiver.d.ts

@@ -5,6 +5,7 @@ import type { RTCDtlsTransport } from "../transport/dtls";
 import type { Kind } from "../types/domain";
 import type { RTCRtpReceiveParameters } from "./parameters";
 import { ReceiverTWCC } from "./receiver/receiverTwcc";
+import { type RTCStats } from "./stats";
 import { MediaStreamTrack } from "./track";
 export declare class RTCRtpReceiver {
     readonly config: PeerConfig;
@@ -34,7 +35,7 @@ export declare class RTCRtpReceiver {
     readonly receiveLastSRTimestamp: {
         [ssrc: number]: number;
     };
-    readonly onPacketLost: Event<[import("werift-rtp/src/rtcp/rtpfb/nack").GenericNack]>;
+    readonly onPacketLost: Event<[import("../imports/rtp").GenericNack]>;
     readonly onRtcp: Event<[RtcpPacket]>;
     dtlsTransport: RTCDtlsTransport;
     sdesMid?: string;
@@ -61,8 +62,7 @@ export declare class RTCRtpReceiver {
     addTrack(track: MediaStreamTrack): boolean;
     stop(): void;
     runRtcp(): Promise<void>;
-    /**todo impl */
-    getStats(): void;
+    getStats(): Promise<RTCStats[]>;
     sendRtcpPLI(mediaSsrc: number): Promise<void>;
     handleRtcpPacket(packet: RtcpPacket): void;
     handleRtpBySsrc: (packet: RtpPacket, extensions: Extensions) => void;

package/lib/webrtc/src/media/rtpReceiver.js

@@ -1,8 +1,8 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.RTCRtpReceiver = void 0;
+const crypto_1 = require("crypto");
 const promises_1 = require("timers/promises");
-const uuid_1 = require("uuid");
 const common_1 = require("../imports/common");
 const rtp_1 = require("../imports/rtp");
 const utils_1 = require("../utils");
@@ -11,6 +11,7 @@ const receiverTwcc_1 = require("./receiver/receiverTwcc");
 const statistics_1 = require("./receiver/statistics");
 const sdp_1 = require("../sdp");
 const rtcpFeedback_1 = require("./extension/rtcpFeedback");
+const stats_1 = require("./stats");
 const track_1 = require("./track");
 const log = (0, rtp_1.debug)("werift:packages/webrtc/src/media/rtpReceiver.ts");
 class RTCRtpReceiver {
@@ -70,7 +71,7 @@ class RTCRtpReceiver {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: (0, uuid_1.v4)()
+            value: (0, crypto_1.randomUUID)().toString()
         });
         Object.defineProperty(this, "tracks", {
             enumerable: true,
@@ -312,8 +313,56 @@ class RTCRtpReceiver {
         }
         catch (error) { }
     }
-    /**todo impl */
-    getStats() { }
+    async getStats() {
+        const timestamp = (0, stats_1.getStatsTimestamp)();
+        const stats = [];
+        if (!this.dtlsTransport) {
+            return stats;
+        }
+        const transportId = (0, stats_1.generateStatsId)("transport", this.dtlsTransport.id);
+        // Collect stats for each track
+        for (const track of this.tracks) {
+            if (!track.ssrc)
+                continue;
+            const streamStats = this.remoteStreams[track.ssrc];
+            if (!streamStats)
+                continue;
+            // Inbound RTP stats
+            const inboundRtpStats = {
+                type: "inbound-rtp",
+                id: (0, stats_1.generateStatsId)("inbound-rtp", track.ssrc),
+                timestamp,
+                ssrc: track.ssrc,
+                kind: this.kind,
+                transportId,
+                codecId: this.codecs[0]
+                    ? (0, stats_1.generateStatsId)("codec", this.codecs[0].payloadType, transportId)
+                    : undefined,
+                mid: this.sdesMid,
+                trackIdentifier: track.id,
+                packetsReceived: streamStats.packets_received,
+                packetsLost: streamStats.packets_lost,
+                jitter: streamStats.jitter,
+            };
+            stats.push(inboundRtpStats);
+            // Remote outbound RTP stats (if we have SR info)
+            if (this.lastSRtimestamp[track.ssrc]) {
+                const remoteOutboundStats = {
+                    type: "remote-outbound-rtp",
+                    id: (0, stats_1.generateStatsId)("remote-outbound-rtp", track.ssrc),
+                    timestamp,
+                    ssrc: track.ssrc,
+                    kind: this.kind,
+                    transportId,
+                    codecId: inboundRtpStats.codecId,
+                    localId: inboundRtpStats.id,
+                    remoteTimestamp: this.receiveLastSRTimestamp[track.ssrc] * 1000, // Convert to ms
+                };
+                stats.push(remoteOutboundStats);
+            }
+        }
+        return stats;
+    }
     async sendRtcpPLI(mediaSsrc) {
         if (!this.pliEnabled) {
             log("pli not supported", { mediaSsrc });

package/lib/webrtc/src/media/rtpReceiver.js.map

@@ -1 +1 @@
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{ setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\nimport { Event, int } from \"../imports/common\";\n\nimport {\n  type Extensions,\n  PictureLossIndication,\n  RTP_EXTENSION_URI,\n  Red,\n  RedHandler,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  type RtpPacket,\n  type TransportWideCCPayload,\n  debug,\n  unwrapRtx,\n} from \"../imports/rtp\";\nimport type { PeerConfig } from \"../peerConnection\";\nimport type { RTCDtlsTransport } from \"../transport/dtls\";\nimport type { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpReceiveParameters,\n} from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\n\nimport { codecParametersFromString } from \"../sdp\";\nimport { usePLI, useTWCC } from \"./extension/rtcpFeedback\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private get codecArray() {\n    return Object.values(this.codecs).sort(\n      (a, b) => a.payloadType - b.payloadType,\n    );\n  }\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new NackHandler(this);\n  private readonly audioRedHandler = new RedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = uuid();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  /**last sender Report Timestamp\n   * compactNtp\n   */\n  readonly lastSRtimestamp: { [ssrc: number]: number } = {};\n  /**seconds */\n  readonly receiveLastSRTimestamp: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  dtlsTransport!: RTCDtlsTransport;\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number,\n  ) {}\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0] ?? new MediaStreamTrack({ kind: this.kind });\n  }\n\n  get nackEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === \"nack\");\n  }\n\n  get twccEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === useTWCC().type,\n    );\n  }\n\n  get pliEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === usePLI().type,\n    );\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc: number) {\n    if (this.twccEnabled && !this.receiverTWCC) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc,\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) {\n        return t.rid === track.rid;\n      }\n      if (t.ssrc) {\n        return t.ssrc === track.ssrc;\n      }\n    });\n    if (exist) {\n      return false;\n    }\n    this.tracks.push(track);\n    if (track.ssrc) {\n      this.trackBySSRC[track.ssrc] = track;\n    }\n    if (track.rid) {\n      this.trackByRID[track.rid] = track;\n    }\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lastSRtimestamp = 0,\n              delaySinceLastSR = 0;\n            if (this.lastSRtimestamp[ssrc]) {\n              lastSRtimestamp = this.lastSRtimestamp[ssrc];\n              const delaySeconds =\n                timestampSeconds() - this.receiveLastSRTimestamp[ssrc];\n              if (delaySeconds > 0 && delaySeconds < 65536) {\n                delaySinceLastSR = int(delaySeconds * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: lastSRtimestamp,\n              dlsr: delaySinceLastSR,\n            });\n          },\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          if (this.config.debug.receiverReportDelay) {\n            await setTimeout(this.config.debug.receiverReportDelay);\n          }\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\n    }\n\n    if (this.stopped) {\n      return;\n    }\n\n    log(\"sendRtcpPLI\", { mediaSsrc });\n\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lastSRtimestamp[sr.ssrc] = compactNtp(\n            sr.senderInfo.ntpTimestamp,\n          );\n          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();\n\n          const track = this.trackBySSRC[packet.ssrc];\n          if (track) {\n            track.onReceiveRtcp.execute(packet);\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack,\n  ) {\n    if (this.stopped) {\n      return;\n    }\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      // log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as TransportWideCCPayload;\n\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.twccEnabled) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      const rtxCodec = this.codecs[codecParams[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    let red: Red | undefined;\n    if (codec.name.toLowerCase() === \"red\") {\n      red = Red.deSerialize(packet.payload);\n      if (\n        !Object.keys(this.codecs).includes(\n          red.header.fields[0].blockPT.toString(),\n        )\n      ) {\n        return;\n      }\n    }\n\n    if (track?.kind === \"video\" && this.nackEnabled) {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        if (track.kind === \"audio\") {\n          const payloads = this.audioRedHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone(), extensions);\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone(), extensions);\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n"]}
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{ randomUUID } from \"crypto\";\nimport { setTimeout } from \"timers/promises\";\nimport { Event, int } from \"../imports/common\";\n\nimport {\n  type Extensions,\n  PictureLossIndication,\n  RTP_EXTENSION_URI,\n  Red,\n  RedHandler,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  type RtpPacket,\n  type TransportWideCCPayload,\n  debug,\n  unwrapRtx,\n} from \"../imports/rtp\";\nimport type { PeerConfig } from \"../peerConnection\";\nimport type { RTCDtlsTransport } from \"../transport/dtls\";\nimport type { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpReceiveParameters,\n} from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\n\nimport { codecParametersFromString } from \"../sdp\";\nimport { usePLI, useTWCC } from \"./extension/rtcpFeedback\";\nimport {\n  type RTCInboundRtpStreamStats,\n  type RTCRemoteOutboundRtpStreamStats,\n  type RTCStats,\n  generateStatsId,\n  getStatsTimestamp,\n} from \"./stats\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private get codecArray() {\n    return Object.values(this.codecs).sort(\n      (a, b) => a.payloadType - b.payloadType,\n    );\n  }\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new NackHandler(this);\n  private readonly audioRedHandler = new RedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = randomUUID().toString();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  /**last sender Report Timestamp\n   * compactNtp\n   */\n  readonly lastSRtimestamp: { [ssrc: number]: number } = {};\n  /**seconds */\n  readonly receiveLastSRTimestamp: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  dtlsTransport!: RTCDtlsTransport;\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number,\n  ) {}\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0] ?? new MediaStreamTrack({ kind: this.kind });\n  }\n\n  get nackEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === \"nack\");\n  }\n\n  get twccEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === useTWCC().type,\n    );\n  }\n\n  get pliEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === usePLI().type,\n    );\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc: number) {\n    if (this.twccEnabled && !this.receiverTWCC) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc,\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) {\n        return t.rid === track.rid;\n      }\n      if (t.ssrc) {\n        return t.ssrc === track.ssrc;\n      }\n    });\n    if (exist) {\n      return false;\n    }\n    this.tracks.push(track);\n    if (track.ssrc) {\n      this.trackBySSRC[track.ssrc] = track;\n    }\n    if (track.rid) {\n      this.trackByRID[track.rid] = track;\n    }\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lastSRtimestamp = 0,\n              delaySinceLastSR = 0;\n            if (this.lastSRtimestamp[ssrc]) {\n              lastSRtimestamp = this.lastSRtimestamp[ssrc];\n              const delaySeconds =\n                timestampSeconds() - this.receiveLastSRTimestamp[ssrc];\n              if (delaySeconds > 0 && delaySeconds < 65536) {\n                delaySinceLastSR = int(delaySeconds * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: lastSRtimestamp,\n              dlsr: delaySinceLastSR,\n            });\n          },\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          if (this.config.debug.receiverReportDelay) {\n            await setTimeout(this.config.debug.receiverReportDelay);\n          }\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  async getStats(): Promise<RTCStats[]> {\n    const timestamp = getStatsTimestamp();\n    const stats: RTCStats[] = [];\n\n    if (!this.dtlsTransport) {\n      return stats;\n    }\n\n    const transportId = generateStatsId(\"transport\", this.dtlsTransport.id);\n\n    // Collect stats for each track\n    for (const track of this.tracks) {\n      if (!track.ssrc) continue;\n\n      const streamStats = this.remoteStreams[track.ssrc];\n      if (!streamStats) continue;\n\n      // Inbound RTP stats\n      const inboundRtpStats: RTCInboundRtpStreamStats = {\n        type: \"inbound-rtp\",\n        id: generateStatsId(\"inbound-rtp\", track.ssrc),\n        timestamp,\n        ssrc: track.ssrc,\n        kind: this.kind,\n        transportId,\n        codecId: this.codecs[0]\n          ? generateStatsId(\"codec\", this.codecs[0].payloadType, transportId)\n          : undefined,\n        mid: this.sdesMid,\n        trackIdentifier: track.id,\n        packetsReceived: streamStats.packets_received,\n        packetsLost: streamStats.packets_lost,\n        jitter: streamStats.jitter,\n      };\n      stats.push(inboundRtpStats);\n\n      // Remote outbound RTP stats (if we have SR info)\n      if (this.lastSRtimestamp[track.ssrc]) {\n        const remoteOutboundStats: RTCRemoteOutboundRtpStreamStats = {\n          type: \"remote-outbound-rtp\",\n          id: generateStatsId(\"remote-outbound-rtp\", track.ssrc),\n          timestamp,\n          ssrc: track.ssrc,\n          kind: this.kind,\n          transportId,\n          codecId: inboundRtpStats.codecId,\n          localId: inboundRtpStats.id,\n          remoteTimestamp: this.receiveLastSRTimestamp[track.ssrc] * 1000, // Convert to ms\n        };\n        stats.push(remoteOutboundStats);\n      }\n    }\n\n    return stats;\n  }\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\n    }\n\n    if (this.stopped) {\n      return;\n    }\n\n    log(\"sendRtcpPLI\", { mediaSsrc });\n\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lastSRtimestamp[sr.ssrc] = compactNtp(\n            sr.senderInfo.ntpTimestamp,\n          );\n          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();\n\n          const track = this.trackBySSRC[packet.ssrc];\n          if (track) {\n            track.onReceiveRtcp.execute(packet);\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack,\n  ) {\n    if (this.stopped) {\n      return;\n    }\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      // log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as TransportWideCCPayload;\n\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.twccEnabled) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      const rtxCodec = this.codecs[codecParams[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    let red: Red | undefined;\n    if (codec.name.toLowerCase() === \"red\") {\n      red = Red.deSerialize(packet.payload);\n      if (\n        !Object.keys(this.codecs).includes(\n          red.header.fields[0].blockPT.toString(),\n        )\n      ) {\n        return;\n      }\n    }\n\n    if (track?.kind === \"video\" && this.nackEnabled) {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        if (track.kind === \"audio\") {\n          const payloads = this.audioRedHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone(), extensions);\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone(), extensions);\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n"]}

package/lib/webrtc/src/media/rtpSender.d.ts

@@ -25,6 +25,7 @@ import type { RTCDtlsTransport } from "../transport/dtls";
 import type { Kind } from "../types/domain";
 import type { RTCRtpCodecParameters, RTCRtpSendParameters } from "./parameters";
 import { SenderBandwidthEstimator } from "./sender/senderBWE";
+import { type RTCStats } from "./stats";
 import type { MediaStreamTrack } from "./track";
 export declare class RTCRtpSender {
     trackOrKind: Kind | MediaStreamTrack;
@@ -82,4 +83,9 @@ export declare class RTCRtpSender {
     replaceRTP({ sequenceNumber, timestamp, }: Pick<RtpHeader, "sequenceNumber" | "timestamp">, discontinuity?: boolean): void;
     sendRtp(rtp: Buffer | RtpPacket): Promise<void>;
     handleRtcpPacket(rtcpPacket: RtcpPacket): void;
+    getParameters(): {
+        encodings: never[];
+    };
+    setParameters(params: any): void;
+    getStats(): Promise<RTCStats[]>;
 }

package/lib/webrtc/src/media/rtpSender.js

@@ -20,40 +20,18 @@
         
 Figure 2: Example for round-trip time computation
  */
-var __createBinding = (this && this.__createBinding) || (Object.create ? (function(o, m, k, k2) {
-    if (k2 === undefined) k2 = k;
-    var desc = Object.getOwnPropertyDescriptor(m, k);
-    if (!desc || ("get" in desc ? !m.__esModule : desc.writable || desc.configurable)) {
-      desc = { enumerable: true, get: function() { return m[k]; } };
-    }
-    Object.defineProperty(o, k2, desc);
-}) : (function(o, m, k, k2) {
-    if (k2 === undefined) k2 = k;
-    o[k2] = m[k];
-}));
-var __setModuleDefault = (this && this.__setModuleDefault) || (Object.create ? (function(o, v) {
-    Object.defineProperty(o, "default", { enumerable: true, value: v });
-}) : function(o, v) {
-    o["default"] = v;
-});
-var __importStar = (this && this.__importStar) || function (mod) {
-    if (mod && mod.__esModule) return mod;
-    var result = {};
-    if (mod != null) for (var k in mod) if (k !== "default" && Object.prototype.hasOwnProperty.call(mod, k)) __createBinding(result, mod, k);
-    __setModuleDefault(result, mod);
-    return result;
-};
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.RTCRtpSender = void 0;
 const crypto_1 = require("crypto");
 const jspack_1 = require("@shinyoshiaki/jspack");
+const crypto_2 = require("crypto");
 const promises_1 = require("timers/promises");
-const uuid = __importStar(require("uuid"));
 const common_1 = require("../imports/common");
 const __1 = require("..");
 const rtp_1 = require("../imports/rtp");
 const utils_1 = require("../utils");
 const senderBWE_1 = require("./sender/senderBWE");
+const stats_1 = require("./stats");
 const log = (0, rtp_1.debug)("werift:packages/webrtc/src/media/rtpSender.ts");
 const RTP_HISTORY_SIZE = 128;
 const RTT_ALPHA = 0.85;
@@ -93,13 +71,13 @@ class RTCRtpSender {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: uuid.v4()
+            value: (0, crypto_2.randomUUID)().toString()
         });
         Object.defineProperty(this, "trackId", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: uuid.v4()
+            value: (0, crypto_2.randomUUID)().toString()
         });
         Object.defineProperty(this, "onReady", {
             enumerable: true,
@@ -634,6 +612,66 @@ class RTCRtpSender {
         }
         this.onRtcp.execute(rtcpPacket);
     }
+    // todo impl
+    getParameters() {
+        return {
+            encodings: [],
+        };
+    }
+    // todo impl
+    setParameters(params) { }
+    async getStats() {
+        const timestamp = (0, stats_1.getStatsTimestamp)();
+        const stats = [];
+        if (!this.dtlsTransport) {
+            return stats;
+        }
+        const transportId = (0, stats_1.generateStatsId)("transport", this.dtlsTransport.id);
+        // Outbound RTP stats
+        const outboundRtpStats = {
+            type: "outbound-rtp",
+            id: (0, stats_1.generateStatsId)("outbound-rtp", this.ssrc),
+            timestamp,
+            ssrc: this.ssrc,
+            kind: this.kind,
+            transportId,
+            codecId: this.codec
+                ? (0, stats_1.generateStatsId)("codec", this.codec.payloadType, transportId)
+                : undefined,
+            mid: this.mid,
+            packetsSent: this.packetCount,
+            bytesSent: this.octetCount,
+            rtxSsrc: this.rtxPayloadType ? this.rtxSsrc : undefined,
+        };
+        stats.push(outboundRtpStats);
+        // Media source stats
+        if (this.track) {
+            const mediaSourceStats = {
+                type: "media-source",
+                id: (0, stats_1.generateStatsId)("media-source", this.trackId),
+                timestamp,
+                trackIdentifier: this.trackId,
+                kind: this.kind,
+            };
+            stats.push(mediaSourceStats);
+        }
+        // Remote inbound RTP stats (if we have RTT)
+        if (this.rtt !== undefined) {
+            const remoteInboundStats = {
+                type: "remote-inbound-rtp",
+                id: (0, stats_1.generateStatsId)("remote-inbound-rtp", this.ssrc),
+                timestamp,
+                ssrc: this.ssrc,
+                kind: this.kind,
+                transportId,
+                codecId: outboundRtpStats.codecId,
+                localId: outboundRtpStats.id,
+                roundTripTime: this.rtt,
+            };
+            stats.push(remoteInboundStats);
+        }
+        return stats;
+    }
 }
 exports.RTCRtpSender = RTCRtpSender;
 //# sourceMappingURL=rtpSender.js.map