werift 0.19.3 → 0.19.5

package/lib/webrtc/src/media/extension/rtpExtension.js.map

@@ -1 +1 @@
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+{"version":3,"file":"rtpExtension.js","sourceRoot":"","sources":["../../../../../src/media/extension/rtpExtension.ts"],"names":[],"mappings":";;AAGA,gCAIC;AAED,gDAIC;AAED,wDAIC;AAED,gDAIC;AAED,wCAIC;AAED,0DAIC;AAED,0DAIC;AA3CD,6CAAwD;AACxD,8CAAgE;AAEhE,SAAgB,UAAU;IACxB,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,OAAO;KAC/B,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,kBAAkB;IAChC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,eAAe;KACvC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,sBAAsB;IACpC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,mBAAmB;KAC3C,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,kBAAkB;IAChC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,eAAe;KACvC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,cAAc;IAC5B,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,WAAW;KACnC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,uBAAuB;IACrC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,oBAAoB;KAC5C,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,uBAAuB;IACrC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,oBAAoB;KAC5C,CAAC,CAAC;AACL,CAAC","sourcesContent":["import { RTP_EXTENSION_URI } from \"../../../../rtp/src\";\nimport { RTCRtpHeaderExtensionParameters } from \"../parameters\";\n\nexport function useSdesMid() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.sdesMid,\n  });\n}\n\nexport function useSdesRTPStreamId() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.sdesRTPStreamID,\n  });\n}\n\nexport function useRepairedRtpStreamId() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.repairedRtpStreamId,\n  });\n}\n\nexport function useTransportWideCC() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.transportWideCC,\n  });\n}\n\nexport function useAbsSendTime() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.absSendTime,\n  });\n}\n\nexport function useDependencyDescriptor() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.dependencyDescriptor,\n  });\n}\n\nexport function useAudioLevelIndication() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.audioLevelIndication,\n  });\n}\n"]}

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

@@ -1,4 +1,4 @@
-import { Direction } from "./rtpTransceiver";
+import type { Direction } from "./rtpTransceiver";
 export interface RTCRtpParameters {
     codecs: RTCRtpCodecParameters[];
     headerExtensions: RTCRtpHeaderExtensionParameters[];

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;;;;mBAA+B,KAAK;WAAC;QAMnC,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 { Direction } 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: Direction | \"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.d.ts

@@ -1,6 +1,6 @@
 import Event from "rx.mini";
-import { GenericNack, RtpPacket } from "../../../../rtp/src";
-import { RTCRtpReceiver } from "../rtpReceiver";
+import { GenericNack, type RtpPacket } from "../../../../rtp/src";
+import type { RTCRtpReceiver } from "../rtpReceiver";
 export declare class NackHandler {
     private receiver;
     private newEstSeqNum;

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

@@ -1 +1 @@
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debug from \"debug\";\nimport range from \"lodash/range\";\nimport Event from \"rx.mini\";\n\nimport { uint16Add } from \"../../../../common/src\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../../rtp/src\";\nimport { 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 > 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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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debug from \"debug\";\nimport range from \"lodash/range\";\nimport Event from \"rx.mini\";\n\nimport { uint16Add } from \"../../../../common/src\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n} from \"../../../../rtp/src\";\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"]}

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

@@ -1,4 +1,4 @@
-import { RTCDtlsTransport } from "../../transport/dtls";
+import type { RTCDtlsTransport } from "../../transport/dtls";
 type ExtensionInfo = {
     tsn: number;
     timestamp: number;

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

@@ -1 +1 @@
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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: 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"]}

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

@@ -1,4 +1,4 @@
-import { RtpPacket } from "../../../../rtp/src";
+import type { RtpPacket } from "../../../../rtp/src";
 export declare class StreamStatistics {
     base_seq?: number;
     max_seq?: number;

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

@@ -1 +1 @@
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package/lib/webrtc/src/media/router.d.ts

@@ -1,8 +1,8 @@
-import { RtcpPacket, RtpPacket } from "../../../rtp/src";
-import { RTCRtpReceiveParameters, RTCRtpSimulcastParameters } from "./parameters";
+import { type RtcpPacket, type RtpPacket } from "../../../rtp/src";
+import type { RTCRtpReceiveParameters, RTCRtpSimulcastParameters } from "./parameters";
 import { RTCRtpReceiver } from "./rtpReceiver";
-import { RTCRtpSender } from "./rtpSender";
-import { RTCRtpTransceiver } from "./rtpTransceiver";
+import type { RTCRtpSender } from "./rtpSender";
+import type { RTCRtpTransceiver } from "./rtpTransceiver";
 export type Extensions = {
     [uri: string]: number | string;
 };

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

@@ -1 +1 @@
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debug from \"debug\";\n\nimport {\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n  rtpHeaderExtensionsParser,\n} from \"../../../rtp/src\";\nimport {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport { RTCRtpSender } from \"./rtpSender\";\nimport { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport type Extensions = { [uri: string]: number | string };\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 = 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        packet = packet as RtcpSrPacket;\n        recipients.push(this.ssrcTable[packet.ssrc]);\n      }\n      break;\n      case RtcpRrPacket.type: {\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        const sdes = packet as RtcpSourceDescriptionPacket;\n        // log(\"sdes\", JSON.stringify(sdes.chunks));\n      }\n      break;\n      case RtcpTransportLayerFeedback.type: {\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        const psfb = packet as RtcpPayloadSpecificFeedback;\n        switch (psfb.feedback.count) {\n          case ReceiverEstimatedMaxBitrate.count: {\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"]}
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debug from \"debug\";\n\nimport {\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n  rtpHeaderExtensionsParser,\n} from \"../../../rtp/src\";\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 type Extensions = { [uri: string]: number | string };\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 = 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"]}

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

@@ -1,12 +1,12 @@
 import Event from "rx.mini";
-import { PeerConfig } from "..";
-import { RtcpPacket, RtpPacket } from "../../../rtp/src";
-import { RTCDtlsTransport } from "../transport/dtls";
-import { Kind } from "../types/domain";
-import { RTCRtpReceiveParameters } from "./parameters";
+import { type PeerConfig } from "..";
+import { type RtcpPacket, type RtpPacket } from "../../../rtp/src";
+import type { RTCDtlsTransport } from "../transport/dtls";
+import type { Kind } from "../types/domain";
+import type { RTCRtpReceiveParameters } from "./parameters";
 import { ReceiverTWCC } from "./receiver/receiverTwcc";
-import { Extensions } from "./router";
-import { MediaStreamTrack } from "./track";
+import type { Extensions } from "./router";
+import type { MediaStreamTrack } from "./track";
 export declare class RTCRtpReceiver {
     readonly config: PeerConfig;
     kind: Kind;

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

@@ -1 +1 @@
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{ debug } from \"debug\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { PeerConfig, codecParametersFromString, usePLI, useTWCC } from \"..\";\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  RTP_EXTENSION_URI,\n  Red,\n  RedHandler,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpPacket,\n  TransportWideCCPayload,\n  unwrapRtx,\n} from \"../../../rtp/src\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\nimport { Extensions } from \"./router\";\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];\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        const sr = packet as RtcpSrPacket;\n        this.lastSRtimestamp[sr.ssrc] = compactNtp(sr.senderInfo.ntpTimestamp);\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());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n"]}
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{ debug } from \"debug\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport {\n  type PeerConfig,\n  codecParametersFromString,\n  usePLI,\n  useTWCC,\n} from \"..\";\nimport { int } from \"../../../common/src\";\nimport {\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  unwrapRtx,\n} from \"../../../rtp/src\";\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\";\nimport type { Extensions } from \"./router\";\nimport type { 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];\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());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n"]}

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

@@ -19,14 +19,13 @@
         
 Figure 2: Example for round-trip time computation
  */
-/// <reference types="node" />
 import Event from "rx.mini";
-import { GenericNack, RedEncoder, RtcpPacket, RtpHeader, RtpPacket } from "../../../rtp/src";
-import { RTCDtlsTransport } from "../transport/dtls";
-import { Kind } from "../types/domain";
-import { RTCRtpCodecParameters, RTCRtpSendParameters } from "./parameters";
+import { GenericNack, RedEncoder, type RtcpPacket, type RtpHeader, RtpPacket } from "../../../rtp/src";
+import type { RTCDtlsTransport } from "../transport/dtls";
+import type { Kind } from "../types/domain";
+import type { RTCRtpCodecParameters, RTCRtpSendParameters } from "./parameters";
 import { SenderBandwidthEstimator } from "./sender/senderBWE";
-import { MediaStreamTrack } from "./track";
+import type { MediaStreamTrack } from "./track";
 export declare class RTCRtpSender {
     trackOrKind: Kind | MediaStreamTrack;
     readonly type = "sender";

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

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  [10 Nov 1995 11:33:25.125 UTC]       [10 Nov 1995 11:33:36.5 UTC]\n   n                 SR(n)              A=b710:8000 (46864.500 s)\n   ---------------------------------------------------------------->\n                      v                 ^\n   ntp_sec =0xb44db705 v               ^ dlsr=0x0005:4000 (    5.250s)\n   ntp_frac=0x20000000  v             ^  lsr =0xb705:2000 (46853.125s)\n     (3024992005.125 s)  v           ^\n   r                      v         ^ RR(n)\n   ---------------------------------------------------------------->\n                          |<-DLSR->|\n                           (5.250 s)\n        \n   A     0xb710:8000 (46864.500 s)\n   DLSR -0x0005:4000 (    5.250 s)\n   LSR  -0xb705:2000 (46853.125 s)\n   -------------------------------\n   delay 0x0006:2000 (    6.125 s)\n        \nFigure 2: Example for round-trip time computation\n */\n\nimport { randomBytes } from \"crypto\";\nimport { jspack } from \"@shinyoshiaki/jspack\";\nimport debug from \"debug\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport * as uuid from \"uuid\";\n\nimport { codecParametersFromString } from \"..\";\nimport { random16, uint16Add, uint32Add } from \"../../../common/src\";\nimport {\n  Extension,\n  GenericNack,\n  PictureLossIndication,\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  RedEncoder,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSenderInfo,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  RtpHeader,\n  RtpPacket,\n  SourceDescriptionChunk,\n  SourceDescriptionItem,\n  TransportWideCC,\n  serializeAbsSendTime,\n  serializeRepairedRtpStreamId,\n  serializeSdesMid,\n  serializeSdesRTPStreamID,\n  serializeTransportWideCC,\n  wrapRtx,\n} from \"../../../rtp/src\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime, timestampSeconds } from \"../utils\";\nimport {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n  RTCRtpSendParameters,\n} from \"./parameters\";\nimport { SenderBandwidthEstimator, SentInfo } from \"./sender/senderBWE\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpSender.ts\");\n\nconst RTP_HISTORY_SIZE = 128;\nconst RTT_ALPHA = 0.85;\n\nexport class RTCRtpSender {\n  readonly type = \"sender\";\n  readonly kind: Kind;\n  readonly ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  readonly rtxSsrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  streamId = uuid.v4();\n  readonly trackId = uuid.v4();\n  readonly onReady = new Event();\n  readonly onRtcp = new Event<[RtcpPacket]>();\n  readonly onPictureLossIndication = new Event<[]>();\n  readonly onGenericNack = new Event<[GenericNack]>();\n  readonly senderBWE = new SenderBandwidthEstimator();\n\n  private cname?: string;\n  private mid?: string;\n  private rtpStreamId?: string;\n  private repairedRtpStreamId?: string;\n  private rtxPayloadType?: number;\n  private rtxSequenceNumber = random16();\n  redRedundantPayloadType?: number;\n  private _redDistance = 2;\n  redEncoder = new RedEncoder(this._redDistance);\n  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lastSRtimestamp?: number;\n  private lastSentSRTimestamp?: number;\n  private ntpTimestamp = 0n;\n  private rtpTimestamp = 0;\n  private octetCount = 0;\n  private packetCount = 0;\n  private rtt?: number;\n  receiverEstimatedMaxBitrate = 0n;\n\n  // rtp\n  private sequenceNumber?: number;\n  private timestamp?: number;\n  private timestampOffset = 0;\n  private seqOffset = 0;\n  private rtpCache: RtpPacket[] = [];\n  codec?: RTCRtpCodecParameters;\n  public dtlsTransport!: RTCDtlsTransport;\n  private dtlsDisposer: (() => void)[] = [];\n\n  track?: MediaStreamTrack;\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(public trackOrKind: Kind | MediaStreamTrack) {\n    this.kind =\n      typeof this.trackOrKind === \"string\"\n        ? this.trackOrKind\n        : this.trackOrKind.kind;\n    if (typeof trackOrKind !== \"string\") {\n      if (trackOrKind.streamId) {\n        this.streamId = trackOrKind.streamId;\n      }\n      this.registerTrack(trackOrKind);\n    }\n  }\n\n  setDtlsTransport(dtlsTransport: RTCDtlsTransport) {\n    if (this.dtlsTransport) {\n      this.dtlsDisposer.forEach((dispose) => dispose());\n    }\n\n    this.dtlsTransport = dtlsTransport;\n    this.dtlsDisposer = [\n      this.dtlsTransport.onStateChange.subscribe((state) => {\n        if (state === \"connected\") {\n          this.onReady.execute();\n        }\n      }).unSubscribe,\n    ];\n  }\n\n  get redDistance() {\n    return this._redDistance;\n  }\n  set redDistance(n: number) {\n    this._redDistance = n;\n    this.redEncoder.distance = n;\n  }\n\n  prepareSend(params: RTCRtpSendParameters) {\n    this.cname = params.rtcp?.cname;\n    this.mid = params.muxId;\n    this.headerExtensions = params.headerExtensions;\n    this.rtpStreamId = params.rtpStreamId;\n    this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n    this.codec = params.codecs[0];\n    if (this.track) {\n      this.track.codec = this.codec;\n    }\n\n    params.codecs.forEach((codec) => {\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      if (\n        codec.name.toLowerCase() === \"rtx\" &&\n        codecParams[\"apt\"] === this.codec?.payloadType\n      ) {\n        this.rtxPayloadType = codec.payloadType;\n      }\n      if (codec.name.toLowerCase() === \"red\") {\n        this.redRedundantPayloadType = Number(\n          (codec.parameters ?? \"\").split(\"/\")[0],\n        );\n      }\n    });\n  }\n\n  registerTrack(track: MediaStreamTrack) {\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n\n    track.id = this.trackId;\n\n    const { unSubscribe } = track.onReceiveRtp.subscribe(async (rtp) => {\n      await this.sendRtp(rtp);\n    });\n    this.track = track;\n    this.disposeTrack = unSubscribe;\n\n    if (this.codec) {\n      track.codec = this.codec;\n    }\n\n    track.onSourceChanged.subscribe((header) => {\n      this.replaceRTP(header);\n    });\n  }\n\n  async replaceTrack(track: MediaStreamTrack | null) {\n    if (track === null) {\n      // todo impl\n      return;\n    }\n\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.sequenceNumber != undefined) {\n      const header =\n        track.header || (await track.onReceiveRtp.asPromise())[0].header;\n\n      this.replaceRTP(header);\n    }\n\n    this.registerTrack(track);\n    log(\"replaceTrack\", \"ssrc\", track.ssrc, \"rid\", track.rid);\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n    this.track = undefined;\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 packets: RtcpPacket[] = [\n          new RtcpSrPacket({\n            ssrc: this.ssrc,\n            senderInfo: new RtcpSenderInfo({\n              ntpTimestamp: this.ntpTimestamp,\n              rtpTimestamp: this.rtpTimestamp,\n              packetCount: this.packetCount,\n              octetCount: this.octetCount,\n            }),\n          }),\n        ];\n        this.lastSRtimestamp = compactNtp(this.ntpTimestamp);\n        this.lastSentSRTimestamp = timestampSeconds();\n\n        if (this.cname) {\n          packets.push(\n            new RtcpSourceDescriptionPacket({\n              chunks: [\n                new SourceDescriptionChunk({\n                  source: this.ssrc,\n                  items: [\n                    new SourceDescriptionItem({ type: 1, text: this.cname }),\n                  ],\n                }),\n              ],\n            }),\n          );\n        }\n\n        try {\n          await this.dtlsTransport.sendRtcp(packets);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  replaceRTP(\n    {\n      sequenceNumber,\n      timestamp,\n    }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">,\n    discontinuity = false,\n  ) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n      if (discontinuity) {\n        this.seqOffset = uint16Add(this.seqOffset, 2);\n      }\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = uint32Add(this.timestamp, -timestamp);\n      if (discontinuity) {\n        this.timestampOffset = uint16Add(this.timestampOffset, 1);\n      }\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  async sendRtp(rtp: Buffer | RtpPacket) {\n    if (this.dtlsTransport.state !== \"connected\" || !this.codec) {\n      return;\n    }\n\n    rtp = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp;\n\n    const { header, payload } = rtp;\n    header.ssrc = this.ssrc;\n    header.payloadType = this.codec.payloadType;\n    header.timestamp = uint32Add(header.timestamp, this.timestampOffset);\n    header.sequenceNumber = uint16Add(header.sequenceNumber, this.seqOffset);\n    this.timestamp = header.timestamp;\n    this.sequenceNumber = header.sequenceNumber;\n\n    const ntpTimestamp = ntpTime();\n\n    const originalHeaderExtensions = [...header.extensions];\n    header.extensions = this.headerExtensions\n      .map((extension) => {\n        const payload = (() => {\n          switch (extension.uri) {\n            case RTP_EXTENSION_URI.sdesMid:\n              if (this.mid) {\n                return serializeSdesMid(this.mid);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.sdesRTPStreamID:\n              if (this.rtpStreamId) {\n                return serializeSdesRTPStreamID(this.rtpStreamId);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.repairedRtpStreamId:\n              if (this.repairedRtpStreamId) {\n                return serializeRepairedRtpStreamId(this.repairedRtpStreamId);\n              }\n              return;\n            case RTP_EXTENSION_URI.transportWideCC:\n              this.dtlsTransport.transportSequenceNumber = uint16Add(\n                this.dtlsTransport.transportSequenceNumber,\n                1,\n              );\n              return serializeTransportWideCC(\n                this.dtlsTransport.transportSequenceNumber,\n              );\n            case RTP_EXTENSION_URI.absSendTime:\n              return serializeAbsSendTime(ntpTimestamp);\n          }\n        })();\n\n        if (payload) return { id: extension.id, payload };\n      })\n      .filter((v) => v) as Extension[];\n    for (const ext of originalHeaderExtensions) {\n      const exist = header.extensions.find((v) => v.id === ext.id);\n      if (exist) {\n        exist.payload = ext.payload;\n      } else {\n        header.extensions.push(ext);\n      }\n    }\n    header.extensions = header.extensions.sort((a, b) => a.id - b.id);\n\n    this.ntpTimestamp = ntpTimestamp;\n    this.rtpTimestamp = header.timestamp;\n    this.octetCount += payload.length;\n    this.packetCount = uint32Add(this.packetCount, 1);\n\n    this.rtpCache[header.sequenceNumber % RTP_HISTORY_SIZE] = rtp;\n\n    let rtpPayload = payload;\n\n    if (this.redRedundantPayloadType) {\n      this.redEncoder.push({\n        block: rtpPayload,\n        timestamp: header.timestamp,\n        blockPT: this.redRedundantPayloadType,\n      });\n      const red = this.redEncoder.build();\n      rtpPayload = red.serialize();\n    }\n\n    const size = await this.dtlsTransport.sendRtp(rtpPayload, header);\n\n    this.runRtcp();\n    const millitime = milliTime();\n    const sentInfo: SentInfo = {\n      wideSeq: this.dtlsTransport.transportSequenceNumber,\n      size,\n      sendingAtMs: millitime,\n      sentAtMs: millitime,\n    };\n    this.senderBWE.rtpPacketSent(sentInfo);\n  }\n\n  handleRtcpPacket(rtcpPacket: RtcpPacket) {\n    switch (rtcpPacket.type) {\n      case RtcpSrPacket.type:\n      case RtcpRrPacket.type: {\n        const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;\n        packet.reports\n          .filter((report) => report.ssrc === this.ssrc)\n          .forEach((report) => {\n            if (this.lastSRtimestamp === report.lsr && report.dlsr) {\n              if (this.lastSentSRTimestamp) {\n                const rtt =\n                  timestampSeconds() -\n                  this.lastSentSRTimestamp -\n                  report.dlsr / 65536;\n                if (this.rtt === undefined) {\n                  this.rtt = rtt;\n                } else {\n                  this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;\n                }\n              }\n            }\n          });\n      }\n      break;\n      case RtcpTransportLayerFeedback.type: {\n        const packet = rtcpPacket as RtcpTransportLayerFeedback;\n        switch (packet.feedback.count) {\n          case TransportWideCC.count: {\n            const feedback = packet.feedback as TransportWideCC;\n            this.senderBWE.receiveTWCC(feedback);\n          }\n          break;\n          case GenericNack.count: {\n            const feedback = packet.feedback as GenericNack;\n            feedback.lost.forEach(async (seqNum) => {\n              let packet: RtpPacket | undefined =\n                this.rtpCache[seqNum % RTP_HISTORY_SIZE];\n              if (packet && packet.header.sequenceNumber !== seqNum) {\n                packet = undefined;\n              }\n              if (packet) {\n                if (this.rtxPayloadType != undefined) {\n                  packet = wrapRtx(\n                    packet,\n                    this.rtxPayloadType,\n                    this.rtxSequenceNumber,\n                    this.rtxSsrc,\n                  );\n                  this.rtxSequenceNumber = uint16Add(this.rtxSequenceNumber, 1);\n                }\n                await this.dtlsTransport.sendRtp(packet.payload, packet.header);\n              }\n            });\n            this.onGenericNack.execute(feedback);\n          }\n          break;\n        }\n      }\n      break;\n      case RtcpPayloadSpecificFeedback.type: {\n        const packet = rtcpPacket as RtcpPayloadSpecificFeedback;\n        switch (packet.feedback.count) {\n          case ReceiverEstimatedMaxBitrate.count: {\n            const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;\n            this.receiverEstimatedMaxBitrate = feedback.bitrate;\n          }\n          break;\n          case PictureLossIndication.count: {\n            this.onPictureLossIndication.execute();\n          }\n          break;\n        }\n      }\n      break;\n    }\n    this.onRtcp.execute(rtcpPacket);\n  }\n}\n"]}
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  [10 Nov 1995 11:33:25.125 UTC]       [10 Nov 1995 11:33:36.5 UTC]\n   n                 SR(n)              A=b710:8000 (46864.500 s)\n   ---------------------------------------------------------------->\n                      v                 ^\n   ntp_sec =0xb44db705 v               ^ dlsr=0x0005:4000 (    5.250s)\n   ntp_frac=0x20000000  v             ^  lsr =0xb705:2000 (46853.125s)\n     (3024992005.125 s)  v           ^\n   r                      v         ^ RR(n)\n   ---------------------------------------------------------------->\n                          |<-DLSR->|\n                           (5.250 s)\n        \n   A     0xb710:8000 (46864.500 s)\n   DLSR -0x0005:4000 (    5.250 s)\n   LSR  -0xb705:2000 (46853.125 s)\n   -------------------------------\n   delay 0x0006:2000 (    6.125 s)\n        \nFigure 2: Example for round-trip time computation\n */\n\nimport { randomBytes } from \"crypto\";\nimport { jspack } from \"@shinyoshiaki/jspack\";\nimport debug from \"debug\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport * as uuid from \"uuid\";\n\nimport { codecParametersFromString } from \"..\";\nimport { random16, uint16Add, uint32Add } from \"../../../common/src\";\nimport {\n  type Extension,\n  GenericNack,\n  PictureLossIndication,\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  RedEncoder,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSenderInfo,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  type RtpHeader,\n  RtpPacket,\n  SourceDescriptionChunk,\n  SourceDescriptionItem,\n  TransportWideCC,\n  serializeAbsSendTime,\n  serializeRepairedRtpStreamId,\n  serializeSdesMid,\n  serializeSdesRTPStreamID,\n  serializeTransportWideCC,\n  wrapRtx,\n} from \"../../../rtp/src\";\nimport type { RTCDtlsTransport } from \"../transport/dtls\";\nimport type { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime, timestampSeconds } from \"../utils\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n  RTCRtpSendParameters,\n} from \"./parameters\";\nimport { SenderBandwidthEstimator, type SentInfo } from \"./sender/senderBWE\";\nimport type { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpSender.ts\");\n\nconst RTP_HISTORY_SIZE = 128;\nconst RTT_ALPHA = 0.85;\n\nexport class RTCRtpSender {\n  readonly type = \"sender\";\n  readonly kind: Kind;\n  readonly ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  readonly rtxSsrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  streamId = uuid.v4();\n  readonly trackId = uuid.v4();\n  readonly onReady = new Event();\n  readonly onRtcp = new Event<[RtcpPacket]>();\n  readonly onPictureLossIndication = new Event<[]>();\n  readonly onGenericNack = new Event<[GenericNack]>();\n  readonly senderBWE = new SenderBandwidthEstimator();\n\n  private cname?: string;\n  private mid?: string;\n  private rtpStreamId?: string;\n  private repairedRtpStreamId?: string;\n  private rtxPayloadType?: number;\n  private rtxSequenceNumber = random16();\n  redRedundantPayloadType?: number;\n  private _redDistance = 2;\n  redEncoder = new RedEncoder(this._redDistance);\n  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lastSRtimestamp?: number;\n  private lastSentSRTimestamp?: number;\n  private ntpTimestamp = 0n;\n  private rtpTimestamp = 0;\n  private octetCount = 0;\n  private packetCount = 0;\n  private rtt?: number;\n  receiverEstimatedMaxBitrate = 0n;\n\n  // rtp\n  private sequenceNumber?: number;\n  private timestamp?: number;\n  private timestampOffset = 0;\n  private seqOffset = 0;\n  private rtpCache: RtpPacket[] = [];\n  codec?: RTCRtpCodecParameters;\n  public dtlsTransport!: RTCDtlsTransport;\n  private dtlsDisposer: (() => void)[] = [];\n\n  track?: MediaStreamTrack;\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(public trackOrKind: Kind | MediaStreamTrack) {\n    this.kind =\n      typeof this.trackOrKind === \"string\"\n        ? this.trackOrKind\n        : this.trackOrKind.kind;\n    if (typeof trackOrKind !== \"string\") {\n      if (trackOrKind.streamId) {\n        this.streamId = trackOrKind.streamId;\n      }\n      this.registerTrack(trackOrKind);\n    }\n  }\n\n  setDtlsTransport(dtlsTransport: RTCDtlsTransport) {\n    if (this.dtlsTransport) {\n      this.dtlsDisposer.forEach((dispose) => dispose());\n    }\n\n    this.dtlsTransport = dtlsTransport;\n    this.dtlsDisposer = [\n      this.dtlsTransport.onStateChange.subscribe((state) => {\n        if (state === \"connected\") {\n          this.onReady.execute();\n        }\n      }).unSubscribe,\n    ];\n  }\n\n  get redDistance() {\n    return this._redDistance;\n  }\n  set redDistance(n: number) {\n    this._redDistance = n;\n    this.redEncoder.distance = n;\n  }\n\n  prepareSend(params: RTCRtpSendParameters) {\n    this.cname = params.rtcp?.cname;\n    this.mid = params.muxId;\n    this.headerExtensions = params.headerExtensions;\n    this.rtpStreamId = params.rtpStreamId;\n    this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n    this.codec = params.codecs[0];\n    if (this.track) {\n      this.track.codec = this.codec;\n    }\n\n    params.codecs.forEach((codec) => {\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      if (\n        codec.name.toLowerCase() === \"rtx\" &&\n        codecParams[\"apt\"] === this.codec?.payloadType\n      ) {\n        this.rtxPayloadType = codec.payloadType;\n      }\n      if (codec.name.toLowerCase() === \"red\") {\n        this.redRedundantPayloadType = Number(\n          (codec.parameters ?? \"\").split(\"/\")[0],\n        );\n      }\n    });\n  }\n\n  registerTrack(track: MediaStreamTrack) {\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n\n    track.id = this.trackId;\n\n    const { unSubscribe } = track.onReceiveRtp.subscribe(async (rtp) => {\n      await this.sendRtp(rtp);\n    });\n    this.track = track;\n    this.disposeTrack = unSubscribe;\n\n    if (this.codec) {\n      track.codec = this.codec;\n    }\n\n    track.onSourceChanged.subscribe((header) => {\n      this.replaceRTP(header);\n    });\n  }\n\n  async replaceTrack(track: MediaStreamTrack | null) {\n    if (track === null) {\n      // todo impl\n      return;\n    }\n\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.sequenceNumber != undefined) {\n      const header =\n        track.header || (await track.onReceiveRtp.asPromise())[0].header;\n\n      this.replaceRTP(header);\n    }\n\n    this.registerTrack(track);\n    log(\"replaceTrack\", \"ssrc\", track.ssrc, \"rid\", track.rid);\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n    this.track = undefined;\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 packets: RtcpPacket[] = [\n          new RtcpSrPacket({\n            ssrc: this.ssrc,\n            senderInfo: new RtcpSenderInfo({\n              ntpTimestamp: this.ntpTimestamp,\n              rtpTimestamp: this.rtpTimestamp,\n              packetCount: this.packetCount,\n              octetCount: this.octetCount,\n            }),\n          }),\n        ];\n        this.lastSRtimestamp = compactNtp(this.ntpTimestamp);\n        this.lastSentSRTimestamp = timestampSeconds();\n\n        if (this.cname) {\n          packets.push(\n            new RtcpSourceDescriptionPacket({\n              chunks: [\n                new SourceDescriptionChunk({\n                  source: this.ssrc,\n                  items: [\n                    new SourceDescriptionItem({ type: 1, text: this.cname }),\n                  ],\n                }),\n              ],\n            }),\n          );\n        }\n\n        try {\n          await this.dtlsTransport.sendRtcp(packets);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  replaceRTP(\n    {\n      sequenceNumber,\n      timestamp,\n    }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">,\n    discontinuity = false,\n  ) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n      if (discontinuity) {\n        this.seqOffset = uint16Add(this.seqOffset, 2);\n      }\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = uint32Add(this.timestamp, -timestamp);\n      if (discontinuity) {\n        this.timestampOffset = uint16Add(this.timestampOffset, 1);\n      }\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  async sendRtp(rtp: Buffer | RtpPacket) {\n    if (this.dtlsTransport.state !== \"connected\" || !this.codec) {\n      return;\n    }\n\n    rtp = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp;\n\n    const { header, payload } = rtp;\n    header.ssrc = this.ssrc;\n    header.payloadType = this.codec.payloadType;\n    header.timestamp = uint32Add(header.timestamp, this.timestampOffset);\n    header.sequenceNumber = uint16Add(header.sequenceNumber, this.seqOffset);\n    this.timestamp = header.timestamp;\n    this.sequenceNumber = header.sequenceNumber;\n\n    const ntpTimestamp = ntpTime();\n\n    const originalHeaderExtensions = [...header.extensions];\n    header.extensions = this.headerExtensions\n      .map((extension) => {\n        const payload = (() => {\n          switch (extension.uri) {\n            case RTP_EXTENSION_URI.sdesMid:\n              if (this.mid) {\n                return serializeSdesMid(this.mid);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.sdesRTPStreamID:\n              if (this.rtpStreamId) {\n                return serializeSdesRTPStreamID(this.rtpStreamId);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.repairedRtpStreamId:\n              if (this.repairedRtpStreamId) {\n                return serializeRepairedRtpStreamId(this.repairedRtpStreamId);\n              }\n              return;\n            case RTP_EXTENSION_URI.transportWideCC:\n              this.dtlsTransport.transportSequenceNumber = uint16Add(\n                this.dtlsTransport.transportSequenceNumber,\n                1,\n              );\n              return serializeTransportWideCC(\n                this.dtlsTransport.transportSequenceNumber,\n              );\n            case RTP_EXTENSION_URI.absSendTime:\n              return serializeAbsSendTime(ntpTimestamp);\n          }\n        })();\n\n        if (payload) return { id: extension.id, payload };\n      })\n      .filter((v) => v) as Extension[];\n    for (const ext of originalHeaderExtensions) {\n      const exist = header.extensions.find((v) => v.id === ext.id);\n      if (exist) {\n        exist.payload = ext.payload;\n      } else {\n        header.extensions.push(ext);\n      }\n    }\n    header.extensions = header.extensions.sort((a, b) => a.id - b.id);\n\n    this.ntpTimestamp = ntpTimestamp;\n    this.rtpTimestamp = header.timestamp;\n    this.octetCount += payload.length;\n    this.packetCount = uint32Add(this.packetCount, 1);\n\n    this.rtpCache[header.sequenceNumber % RTP_HISTORY_SIZE] = rtp;\n\n    let rtpPayload = payload;\n\n    if (this.redRedundantPayloadType) {\n      this.redEncoder.push({\n        block: rtpPayload,\n        timestamp: header.timestamp,\n        blockPT: this.redRedundantPayloadType,\n      });\n      const red = this.redEncoder.build();\n      rtpPayload = red.serialize();\n    }\n\n    const size = await this.dtlsTransport.sendRtp(rtpPayload, header);\n\n    this.runRtcp();\n    const millitime = milliTime();\n    const sentInfo: SentInfo = {\n      wideSeq: this.dtlsTransport.transportSequenceNumber,\n      size,\n      sendingAtMs: millitime,\n      sentAtMs: millitime,\n    };\n    this.senderBWE.rtpPacketSent(sentInfo);\n  }\n\n  handleRtcpPacket(rtcpPacket: RtcpPacket) {\n    switch (rtcpPacket.type) {\n      case RtcpSrPacket.type:\n      case RtcpRrPacket.type:\n        {\n          const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;\n          packet.reports\n            .filter((report) => report.ssrc === this.ssrc)\n            .forEach((report) => {\n              if (this.lastSRtimestamp === report.lsr && report.dlsr) {\n                if (this.lastSentSRTimestamp) {\n                  const rtt =\n                    timestampSeconds() -\n                    this.lastSentSRTimestamp -\n                    report.dlsr / 65536;\n                  if (this.rtt === undefined) {\n                    this.rtt = rtt;\n                  } else {\n                    this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;\n                  }\n                }\n              }\n            });\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpTransportLayerFeedback;\n          switch (packet.feedback.count) {\n            case TransportWideCC.count:\n              {\n                const feedback = packet.feedback as TransportWideCC;\n                this.senderBWE.receiveTWCC(feedback);\n              }\n              break;\n            case GenericNack.count:\n              {\n                const feedback = packet.feedback as GenericNack;\n                feedback.lost.forEach(async (seqNum) => {\n                  let packet: RtpPacket | undefined =\n                    this.rtpCache[seqNum % RTP_HISTORY_SIZE];\n                  if (packet && packet.header.sequenceNumber !== seqNum) {\n                    packet = undefined;\n                  }\n                  if (packet) {\n                    if (this.rtxPayloadType != undefined) {\n                      packet = wrapRtx(\n                        packet,\n                        this.rtxPayloadType,\n                        this.rtxSequenceNumber,\n                        this.rtxSsrc,\n                      );\n                      this.rtxSequenceNumber = uint16Add(\n                        this.rtxSequenceNumber,\n                        1,\n                      );\n                    }\n                    await this.dtlsTransport.sendRtp(\n                      packet.payload,\n                      packet.header,\n                    );\n                  }\n                });\n                this.onGenericNack.execute(feedback);\n              }\n              break;\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpPayloadSpecificFeedback;\n          switch (packet.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              {\n                const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;\n                this.receiverEstimatedMaxBitrate = feedback.bitrate;\n              }\n              break;\n            case PictureLossIndication.count:\n              {\n                this.onPictureLossIndication.execute();\n              }\n              break;\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(rtcpPacket);\n  }\n}\n"]}

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

@@ -1,10 +1,10 @@
 import Event from "rx.mini";
-import { RTCDtlsTransport } from "..";
-import { Kind } from "../types/domain";
-import { RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters } from "./parameters";
-import { RTCRtpReceiver } from "./rtpReceiver";
-import { RTCRtpSender } from "./rtpSender";
-import { MediaStreamTrack } from "./track";
+import type { RTCDtlsTransport } from "..";
+import type { Kind } from "../types/domain";
+import type { RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters } from "./parameters";
+import type { RTCRtpReceiver } from "./rtpReceiver";
+import type { RTCRtpSender } from "./rtpSender";
+import type { MediaStreamTrack } from "./track";
 export declare class RTCRtpTransceiver {
     readonly kind: Kind;
     receiver: RTCRtpReceiver;

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

@@ -1 +1 @@
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\"\")) {\n      this.usedForSender = true;\n    }\n  }\n\n  /**RFC 8829 4.2.5. last negotiated direction */\n  get currentDirection(): Direction | undefined {\n    return this._currentDirection;\n  }\n\n  setCurrentDirection(direction: Direction | undefined) {\n    this._currentDirection = direction;\n  }\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.receiver.setDtlsTransport(dtls);\n    this.sender.setDtlsTransport(dtls);\n  }\n\n  get msid() {\n    return `${this.sender.streamId} ${this.sender.trackId}`;\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const res = this.receiver.addTrack(track);\n    if (res) {\n      this.onTrack.execute(track, this);\n    }\n  }\n\n  // todo impl\n  // https://www.w3.org/TR/webrtc/#methods-8\n  stop() {\n    if (this.stopping) {\n      return;\n    }\n\n    // todo Stop sending and receiving with transceiver.\n\n    this.stopping = true;\n  }\n\n  getPayloadType(mimeType: string) {\n    return this.codecs.find((codec) =>\n      codec.mimeType.toLowerCase().includes(mimeType.toLowerCase()),\n    )?.payloadType;\n  }\n}\n\nexport const Inactive = \"inactive\";\nexport const Sendonly = \"sendonly\";\nexport const Recvonly = \"recvonly\";\nexport const Sendrecv = \"sendrecv\";\n\nexport const Directions = [Inactive, Sendonly, Recvonly, Sendrecv] as const;\n\nexport type Direction = (typeof Directions)[number];\n\ntype SimulcastDirection = \"send\" | \"recv\";\n\nexport interface TransceiverOptions {\n  direction: Direction;\n  simulcast: { direction: SimulcastDirection; rid: string }[];\n}\n"]}
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\"\")) {\n      this.usedForSender = true;\n    }\n  }\n\n  /**RFC 8829 4.2.5. last negotiated direction */\n  get currentDirection(): Direction | undefined {\n    return this._currentDirection;\n  }\n\n  setCurrentDirection(direction: Direction | undefined) {\n    this._currentDirection = direction;\n  }\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.receiver.setDtlsTransport(dtls);\n    this.sender.setDtlsTransport(dtls);\n  }\n\n  get msid() {\n    return `${this.sender.streamId} ${this.sender.trackId}`;\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const res = this.receiver.addTrack(track);\n    if (res) {\n      this.onTrack.execute(track, this);\n    }\n  }\n\n  // todo impl\n  // https://www.w3.org/TR/webrtc/#methods-8\n  stop() {\n    if (this.stopping) {\n      return;\n    }\n\n    // todo Stop sending and receiving with transceiver.\n\n    this.stopping = true;\n  }\n\n  getPayloadType(mimeType: string) {\n    return this.codecs.find((codec) =>\n      codec.mimeType.toLowerCase().includes(mimeType.toLowerCase()),\n    )?.payloadType;\n  }\n}\n\nexport const Inactive = \"inactive\";\nexport const Sendonly = \"sendonly\";\nexport const Recvonly = \"recvonly\";\nexport const Sendrecv = \"sendrecv\";\n\nexport const Directions = [Inactive, Sendonly, Recvonly, Sendrecv] as const;\n\nexport type Direction = (typeof Directions)[number];\n\ntype SimulcastDirection = \"send\" | \"recv\";\n\nexport interface TransceiverOptions {\n  direction: Direction;\n  simulcast: { direction: SimulcastDirection; rid: string }[];\n}\n"]}

package/lib/webrtc/src/media/sender/senderBWE.d.ts

@@ -1,5 +1,5 @@
 import Event from "rx.mini";
-import { TransportWideCC } from "../../../../rtp/src";
+import type { TransportWideCC } from "../../../../rtp/src";
 export declare class SenderBandwidthEstimator {
     congestion: boolean;
     readonly onAvailableBitrate: Event<[number]>;