werift 0.24.4 → 0.25.0

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

@@ -1 +1 @@
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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\";\n\nimport { randomUUID } from \"crypto\";\nimport { setTimeout } from \"timers/promises\";\nimport { Event, random16, uint16Add, uint32Add } from \"../imports/common\";\n\nimport { codecParametersFromString } from \"..\";\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  debug,\n  serializeAbsSendTime,\n  serializeRepairedRtpStreamId,\n  serializeSdesMid,\n  serializeSdesRTPStreamID,\n  serializeTransportWideCC,\n  wrapRtx,\n} from \"../imports/rtp\";\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 {\n  type RTCCodecStats,\n  type RTCMediaSourceStats,\n  type RTCOutboundRtpStreamStats,\n  type RTCRemoteInboundRtpStreamStats,\n  type RTCStats,\n  type RTCStatsReport,\n  buildStatsReport,\n  generateCodecStatsId,\n  generateStatsId,\n  getStatsTimestamp,\n} from \"./stats\";\nimport type { MediaStream, 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 = randomBytes(4).readUInt32BE(0);\n  readonly rtxSsrc = randomBytes(4).readUInt32BE(0);\n  readonly trackId = randomUUID().toString();\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  private sendEncodings: Array<Record<string, unknown>> = [{}];\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 headerBytesSent = 0;\n  private rtt?: number;\n  private totalRoundTripTime = 0;\n  private roundTripTimeMeasurements = 0;\n  private retransmittedPacketsSent = 0;\n  private retransmittedBytesSent = 0;\n  private nackCount = 0;\n  private pliCount = 0;\n  private firCount = 0;\n  private remotePacketsLost?: number;\n  private remoteFractionLost?: 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 | null = null;\n  streamIds: string[] = [];\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      this.registerTrack(trackOrKind);\n    }\n  }\n\n  get transport() {\n    return this.dtlsTransport ?? null;\n  }\n\n  get streamId() {\n    return this.streamIds[0];\n  }\n\n  set streamId(value: string | undefined) {\n    this.streamIds = value ? [value] : [];\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  setStreams(streams: MediaStream[] = []) {\n    this.streamIds = [...new Set(streams.map((stream) => stream.id))];\n  }\n\n  setSendEncodings(encodings: Array<Record<string, unknown>> = []) {\n    this.sendEncodings =\n      encodings.length > 0\n        ? encodings.map((encoding) => ({ ...encoding }))\n        : [{}];\n  }\n\n  async replaceTrack(track: MediaStreamTrack | null) {\n    if (track === null) {\n      if (this.disposeTrack) {\n        this.disposeTrack();\n      }\n      this.track = null;\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 = null;\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.headerBytesSent += header.serializeSize;\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              this.remotePacketsLost = report.packetsLost;\n              this.remoteFractionLost = report.fractionLost / 256;\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                  this.totalRoundTripTime += rtt;\n                  this.roundTripTimeMeasurements++;\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                this.nackCount++;\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                    this.retransmittedPacketsSent++;\n                    this.retransmittedBytesSent += packet.payload.length;\n                    this.headerBytesSent += packet.header.serializeSize;\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.pliCount++;\n                this.onPictureLossIndication.execute();\n              }\n              break;\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(rtcpPacket);\n  }\n\n  // todo impl\n  getParameters() {\n    return {\n      encodings: this.sendEncodings.map((encoding) => ({ ...encoding })),\n    };\n  }\n\n  // todo impl\n  setParameters(params: { encodings?: Array<Record<string, unknown>> }) {\n    if (params.encodings) {\n      this.setSendEncodings(params.encodings);\n    }\n  }\n\n  private get outboundRtpStatsId() {\n    return generateStatsId(\"outbound-rtp\", this.trackId);\n  }\n\n  private get mediaSourceStatsId() {\n    return generateStatsId(\"media-source\", this.trackId);\n  }\n\n  private get remoteInboundRtpStatsId() {\n    return generateStatsId(\"remote-inbound-rtp\", this.trackId);\n  }\n\n  getStatsRootIds() {\n    return [this.outboundRtpStatsId];\n  }\n\n  collectStats(timestamp: number): RTCStats[] {\n    const stats: RTCStats[] = [];\n    const transportId = this.dtlsTransport\n      ? generateStatsId(\"transport\", this.dtlsTransport.id)\n      : undefined;\n    const codecId =\n      this.codec && transportId\n        ? generateCodecStatsId(\n            transportId,\n            this.codec.payloadType,\n            this.trackId,\n          )\n        : undefined;\n\n    // Outbound RTP stats\n    const outboundRtpStats: RTCOutboundRtpStreamStats = {\n      type: \"outbound-rtp\",\n      id: this.outboundRtpStatsId,\n      timestamp,\n      ssrc: this.ssrc,\n      kind: this.kind,\n      transportId,\n      codecId,\n      mid: this.mid,\n      packetsSent: this.packetCount,\n      bytesSent: this.octetCount,\n      headerBytesSent: this.headerBytesSent,\n      retransmittedPacketsSent: this.retransmittedPacketsSent || undefined,\n      retransmittedBytesSent: this.retransmittedBytesSent || undefined,\n      rtxSsrc: this.rtxPayloadType ? this.rtxSsrc : undefined,\n      mediaSourceId: this.track ? this.mediaSourceStatsId : undefined,\n      remoteId:\n        this.rtt !== undefined ||\n        this.remotePacketsLost !== undefined ||\n        this.remoteFractionLost !== undefined\n          ? this.remoteInboundRtpStatsId\n          : undefined,\n      nackCount: this.nackCount || undefined,\n      pliCount: this.pliCount || undefined,\n      firCount: this.firCount || undefined,\n    };\n    stats.push(outboundRtpStats);\n\n    // Media source stats\n    if (this.track) {\n      const mediaSourceStats: RTCMediaSourceStats = {\n        type: \"media-source\",\n        id: this.mediaSourceStatsId,\n        timestamp,\n        trackIdentifier: this.track.id ?? this.trackId,\n        kind: this.kind,\n      };\n      stats.push(mediaSourceStats);\n    }\n\n    if (this.codec && transportId) {\n      const codecStats: RTCCodecStats = {\n        type: \"codec\",\n        id: codecId!,\n        timestamp,\n        payloadType: this.codec.payloadType,\n        transportId,\n        mimeType: this.codec.mimeType,\n        clockRate: this.codec.clockRate,\n        channels: this.codec.channels,\n        sdpFmtpLine: this.codec.parameters,\n      };\n      stats.push(codecStats);\n    }\n\n    // Remote inbound RTP stats (if we have RTT)\n    if (\n      this.rtt !== undefined ||\n      this.remotePacketsLost !== undefined ||\n      this.remoteFractionLost !== undefined\n    ) {\n      const remoteInboundStats: RTCRemoteInboundRtpStreamStats = {\n        type: \"remote-inbound-rtp\",\n        id: this.remoteInboundRtpStatsId,\n        timestamp,\n        ssrc: this.ssrc,\n        kind: this.kind,\n        transportId,\n        codecId: outboundRtpStats.codecId,\n        localId: outboundRtpStats.id,\n        roundTripTime: this.rtt,\n        totalRoundTripTime: this.totalRoundTripTime,\n        roundTripTimeMeasurements: this.roundTripTimeMeasurements,\n        packetsLost: this.remotePacketsLost,\n        fractionLost: this.remoteFractionLost,\n      };\n      stats.push(remoteInboundStats);\n    }\n\n    return stats;\n  }\n\n  async getStats(): Promise<RTCStatsReport> {\n    const timestamp = getStatsTimestamp();\n    const stats = this.collectStats(timestamp);\n\n    if (this.dtlsTransport) {\n      stats.push(...(await this.dtlsTransport.getStats(timestamp)));\n    }\n\n    return buildStatsReport(stats, this.getStatsRootIds());\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\";\n\nimport { randomUUID } from \"crypto\";\nimport { setTimeout } from \"timers/promises\";\nimport { Event, random16, uint16Add, uint32Add } from \"../imports/common\";\n\nimport { codecParametersFromString } from \"..\";\nimport { createWebRtcDomException } from \"../errors\";\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  debug,\n  serializeAbsSendTime,\n  serializeRepairedRtpStreamId,\n  serializeSdesMid,\n  serializeSdesRTPStreamID,\n  serializeTransportWideCC,\n  wrapRtx,\n} from \"../imports/rtp\";\nimport type { RTCDtlsTransport } from \"../transport/dtls\";\nimport type { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime, timestampSeconds } from \"../utils\";\nimport { isCodecCompatible } from \"./codecCompatibility\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n  RTCRtpSendParameters,\n} from \"./parameters\";\nimport { SenderBandwidthEstimator, type SentInfo } from \"./sender/senderBWE\";\nimport {\n  type RTCCodecStats,\n  type RTCMediaSourceStats,\n  type RTCOutboundRtpStreamStats,\n  type RTCRemoteInboundRtpStreamStats,\n  type RTCStats,\n  type RTCStatsReport,\n  buildStatsReport,\n  generateCodecStatsId,\n  generateStatsId,\n  getStatsTimestamp,\n} from \"./stats\";\nimport {\n  type MediaStream,\n  type MediaStreamTrack,\n  captureTrackSourceCodecs,\n  getTrackSourceCodecs,\n} from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpSender.ts\");\n\nconst RTP_HISTORY_SIZE = 128;\nconst DEFAULT_PENDING_RTP_MAX_LENGTH = 256;\nconst RTT_ALPHA = 0.85;\n\nexport type PendingRtpOptions = {\n  /** Queue RTP until DTLS is connected and a codec is set. Default true when this object is passed. */\n  enabled?: boolean;\n  /** Max queued packets when enabled. Oldest packets are dropped. Default 256. */\n  maxLength?: number;\n};\n\nexport type RTCRtpSenderOptions = {\n  /** Pending RTP cache. Disabled by default. Pass `true` or `{ maxLength }` to enable. */\n  pendingRtp?: boolean | PendingRtpOptions;\n};\n\ntype PendingRtpItem = {\n  packet: RtpPacket;\n  resolve: () => void;\n  reject: (error: unknown) => void;\n};\n\nfunction resolvePendingRtpOptions(\n  pendingRtp: RTCRtpSenderOptions[\"pendingRtp\"],\n): { enabled: boolean; maxLength: number } {\n  if (pendingRtp === true) {\n    return { enabled: true, maxLength: DEFAULT_PENDING_RTP_MAX_LENGTH };\n  }\n  if (pendingRtp == undefined || pendingRtp === false) {\n    return { enabled: false, maxLength: DEFAULT_PENDING_RTP_MAX_LENGTH };\n  }\n  const maxLength =\n    typeof pendingRtp.maxLength === \"number\" &&\n    Number.isFinite(pendingRtp.maxLength) &&\n    pendingRtp.maxLength >= 1\n      ? Math.floor(pendingRtp.maxLength)\n      : DEFAULT_PENDING_RTP_MAX_LENGTH;\n  return { enabled: pendingRtp.enabled ?? true, maxLength };\n}\n\nfunction freezeRtpContinuityOffsets(\n  lastOutputSeq: number,\n  lastOutputTimestamp: number,\n  firstInputSeq: number,\n  firstInputTimestamp: number,\n  timestampStep: number,\n) {\n  return {\n    seqOffset: uint16Add(uint16Add(lastOutputSeq, 1), -firstInputSeq),\n    timestampOffset: uint32Add(\n      uint32Add(lastOutputTimestamp, timestampStep),\n      -firstInputTimestamp,\n    ),\n  };\n}\n\nexport class RTCRtpSender {\n  readonly type = \"sender\";\n  readonly kind: Kind;\n  readonly ssrc = randomBytes(4).readUInt32BE(0);\n  readonly rtxSsrc = randomBytes(4).readUInt32BE(0);\n  readonly trackId = randomUUID().toString();\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  private sendEncodings: Array<Record<string, unknown>> = [{}];\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 headerBytesSent = 0;\n  private rtt?: number;\n  private totalRoundTripTime = 0;\n  private roundTripTimeMeasurements = 0;\n  private retransmittedPacketsSent = 0;\n  private retransmittedBytesSent = 0;\n  private nackCount = 0;\n  private pliCount = 0;\n  private firCount = 0;\n  private remotePacketsLost?: number;\n  private remoteFractionLost?: 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 rtpContinuityPending = false;\n  private pendingTimestampStep = 1;\n  private rtpCache: RtpPacket[] = [];\n  private pendingRtp: PendingRtpItem[] = [];\n  private drainingPendingRtp = false;\n  private readonly pendingRtpEnabled: boolean;\n  private readonly pendingRtpMaxLength: number;\n  codec?: RTCRtpCodecParameters;\n  private negotiatedCodecs: RTCRtpCodecParameters[] = [];\n  private sendPrimaryCodec?: RTCRtpCodecParameters;\n  public dtlsTransport!: RTCDtlsTransport;\n  private dtlsDisposer: (() => void)[] = [];\n\n  track: MediaStreamTrack | null = null;\n  streamIds: string[] = [];\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(\n    public trackOrKind: Kind | MediaStreamTrack,\n    options: RTCRtpSenderOptions = {},\n  ) {\n    const pendingRtp = resolvePendingRtpOptions(options.pendingRtp);\n    this.pendingRtpEnabled = pendingRtp.enabled;\n    this.pendingRtpMaxLength = pendingRtp.maxLength;\n    this.kind =\n      typeof this.trackOrKind === \"string\"\n        ? this.trackOrKind\n        : this.trackOrKind.kind;\n    if (typeof trackOrKind !== \"string\") {\n      this.registerTrack(trackOrKind);\n    }\n  }\n\n  get transport() {\n    return this.dtlsTransport ?? null;\n  }\n\n  get streamId() {\n    return this.streamIds[0];\n  }\n\n  set streamId(value: string | undefined) {\n    this.streamIds = value ? [value] : [];\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          void this.drainPendingRtp();\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 ?? this.rtpStreamId;\n    this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n    this.negotiatedCodecs = [...params.codecs];\n    this.codec =\n      params.codecs.find((codec) => codec.name.toLowerCase() !== \"rtx\") ??\n      params.codecs[0];\n    const redPrimaryPayloadType =\n      this.codec?.name.toLowerCase() === \"red\"\n        ? Number((this.codec.parameters ?? \"\").split(\"/\")[0])\n        : undefined;\n    this.sendPrimaryCodec =\n      this.codec?.name.toLowerCase() === \"red\"\n        ? params.codecs.find(\n            (codec) =>\n              ![\"red\", \"rtx\"].includes(codec.name.toLowerCase()) &&\n              codec.payloadType === redPrimaryPayloadType,\n          )\n        : this.codec;\n    if (this.track) {\n      this.track.codec = this.codec;\n    }\n\n    this.rtxPayloadType = undefined;\n    this.redRedundantPayloadType = undefined;\n    params.codecs.forEach((codec) => {\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      if (\n        codec.name.toLowerCase() === \"rtx\" &&\n        this.sendPrimaryCodec?.payloadType === codecParams[\"apt\"]\n      ) {\n        this.rtxPayloadType = codec.payloadType;\n      }\n    });\n    if (\n      this.codec?.name.toLowerCase() === \"red\" &&\n      this.sendPrimaryCodec != undefined\n    ) {\n      this.redRedundantPayloadType = this.sendPrimaryCodec.payloadType;\n    }\n    void this.drainPendingRtp();\n  }\n\n  private canSendRtp() {\n    return (\n      !this.stopped && this.dtlsTransport?.state === \"connected\" && !!this.codec\n    );\n  }\n\n  private settlePendingRtp(item: PendingRtpItem, error?: unknown) {\n    if (error == undefined) {\n      item.resolve();\n      return;\n    }\n    item.reject(error);\n  }\n\n  private discardPendingRtp() {\n    const dropped = this.pendingRtp.splice(0);\n    for (const item of dropped) {\n      this.settlePendingRtp(item);\n    }\n  }\n\n  private enqueuePendingRtp(\n    rtp: Buffer | RtpPacket,\n    resolve: () => void,\n    reject: (error: unknown) => void,\n  ) {\n    const packet = Buffer.isBuffer(rtp)\n      ? RtpPacket.deSerialize(rtp)\n      : rtp.clone();\n    this.pendingRtp.push({ packet, resolve, reject });\n    while (this.pendingRtp.length > this.pendingRtpMaxLength) {\n      const dropped = this.pendingRtp.shift();\n      if (dropped) {\n        this.settlePendingRtp(dropped);\n      }\n    }\n  }\n\n  /**\n   * Send queued RTP once DTLS is connected and a codec is set. Used only when\n   * pending RTP is enabled: `writeRtp` / `sendRtp` may run before ICE/DTLS\n   * completes; dropping those packets would lose the start of a media source.\n   * `drainingPendingRtp` blocks re-entry while `dispatchRtp` is awaited;\n   * packets enqueued during that wait are drained by the recursive call after\n   * the flag is cleared.\n   */\n  private async drainPendingRtp() {\n    if (this.drainingPendingRtp) {\n      return;\n    }\n    this.drainingPendingRtp = true;\n    try {\n      while (this.pendingRtp.length > 0 && this.canSendRtp()) {\n        const item = this.pendingRtp.shift()!;\n        if (!this.canSendRtp()) {\n          this.settlePendingRtp(item);\n          continue;\n        }\n        try {\n          await this.dispatchRtp(item.packet);\n          this.settlePendingRtp(item);\n        } catch (error) {\n          this.settlePendingRtp(item, error);\n        }\n      }\n    } finally {\n      this.drainingPendingRtp = false;\n    }\n    if (this.pendingRtp.length > 0 && this.canSendRtp()) {\n      await this.drainPendingRtp();\n    }\n  }\n\n  registerTrack(track: MediaStreamTrack) {\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    captureTrackSourceCodecs(track);\n\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n\n    track.id = this.trackId;\n\n    const { unSubscribe: unSubscribeRtp } = track.onReceiveRtp.subscribe(\n      async (rtp) => {\n        await this.sendRtp(rtp);\n      },\n    );\n    const { unSubscribe: unSubscribeSourceChanged } =\n      track.onSourceChanged.subscribe((header) => {\n        this.replaceRTP(header);\n      });\n    this.track = track;\n    this.disposeTrack = () => {\n      unSubscribeRtp();\n      unSubscribeSourceChanged();\n    };\n\n    if (this.codec) {\n      track.codec = this.codec;\n    }\n  }\n\n  setStreams(streams: MediaStream[] = []) {\n    this.streamIds = [...new Set(streams.map((stream) => stream.id))];\n  }\n\n  setSendEncodings(encodings: Array<Record<string, unknown>> = []) {\n    this.sendEncodings =\n      encodings.length > 0\n        ? encodings.map((encoding) => ({ ...encoding }))\n        : [{}];\n    const rid = this.sendEncodings.find(\n      (encoding) => typeof encoding.rid === \"string\",\n    )?.rid;\n    if (typeof rid === \"string\") {\n      this.rtpStreamId = rid;\n    }\n  }\n\n  async replaceTrack(track: MediaStreamTrack | null) {\n    if (track === null) {\n      this.detachTrack();\n      return;\n    }\n\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    captureTrackSourceCodecs(track);\n    const sourceCodecs = getTrackSourceCodecs(track);\n    if (\n      this.sendPrimaryCodec != undefined &&\n      sourceCodecs != undefined &&\n      !sourceCodecs.some((source) =>\n        isCodecCompatible(source, this.sendPrimaryCodec!),\n      )\n    ) {\n      throw createWebRtcDomException(\n        \"InvalidModificationError\",\n        `Track codec ${sourceCodecs.map((codec) => codec.mimeType).join(\", \")} is incompatible with negotiated codecs ${this.negotiatedCodecs.map((codec) => codec.mimeType).join(\", \")}`,\n      );\n    }\n\n    if (this.sequenceNumber != undefined) {\n      this.scheduleRtpContinuity();\n    }\n\n    this.registerTrack(track);\n    log(\"replaceTrack\", \"ssrc\", track.ssrc, \"rid\", track.rid);\n  }\n\n  /**\n   * Detach the current track without stopping the sender (`removeTrack`).\n   * The same sender can resume sending after `replaceTrack(track)`.\n   */\n  detachTrack() {\n    this.rtpContinuityPending = false;\n    this.discardPendingRtp();\n    if (this.disposeTrack) {\n      this.disposeTrack();\n      this.disposeTrack = undefined;\n    }\n    this.track = null;\n  }\n\n  stop({ keepTrack = false }: { keepTrack?: boolean } = {}) {\n    this.stopped = true;\n    this.rtpContinuityPending = false;\n    this.discardPendingRtp();\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n    if (this.disposeTrack) {\n      this.disposeTrack();\n      this.disposeTrack = undefined;\n    }\n    if (!keepTrack) {\n      this.track = null;\n    }\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  /**\n   * Send an RTP packet. Pending RTP is disabled by default: the packet is\n   * written immediately if DTLS is connected and a codec is set, otherwise\n   * dropped.\n   *\n   * When pending RTP is enabled, the packet is queued until it can be sent.\n   * The returned promise then settles for this packet only:\n   * - resolve: DTLS send completed, or the packet was dropped by `stop()`,\n   *   `replaceTrack(null)`, or pending-queue overflow\n   * - reject: DTLS send threw while writing this packet\n   *\n   * Later `sendRtp()` calls stay pending until their own packet is sent or\n   * dropped, even if a drain is already in progress.\n   */\n  async sendRtp(rtp: Buffer | RtpPacket) {\n    if (this.stopped) {\n      return;\n    }\n    if (!this.pendingRtpEnabled) {\n      if (!this.canSendRtp()) {\n        return;\n      }\n      await this.dispatchRtp(rtp);\n      return;\n    }\n    await new Promise<void>((resolve, reject) => {\n      this.enqueuePendingRtp(rtp, resolve, reject);\n      void this.drainPendingRtp();\n    });\n  }\n\n  /**\n   * Schedule RTP continuity rewrite for the next packet that is actually sent.\n   * The header argument is kept for API compatibility and is not used to compute\n   * offsets. `discontinuity` does not change sequence or timestamp mapping.\n   * `timestampStep` (default 1) is the only way to choose the timestamp increment\n   * at the source-switch boundary.\n   */\n  replaceRTP(\n    header: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">,\n    discontinuity = false,\n    timestampStep = 1,\n  ) {\n    this.scheduleRtpContinuity(timestampStep);\n    log(\n      \"replaceRTP\",\n      this.sequenceNumber,\n      header.sequenceNumber,\n      discontinuity,\n      timestampStep,\n    );\n  }\n\n  private scheduleRtpContinuity(timestampStep = 1) {\n    this.rtpContinuityPending = true;\n    this.pendingTimestampStep = timestampStep;\n    this.rtpCache = [];\n  }\n\n  private async dispatchRtp(rtp: Buffer | RtpPacket) {\n    if (!this.canSendRtp()) {\n      return;\n    }\n\n    const codec = this.codec;\n    if (!codec) {\n      return;\n    }\n\n    rtp = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp.clone();\n\n    const { header, payload } = rtp;\n    const inputSequenceNumber = header.sequenceNumber;\n    const inputTimestamp = header.timestamp;\n\n    if (this.rtpContinuityPending) {\n      if (this.sequenceNumber != undefined && this.timestamp != undefined) {\n        const offsets = freezeRtpContinuityOffsets(\n          this.sequenceNumber,\n          this.timestamp,\n          inputSequenceNumber,\n          inputTimestamp,\n          this.pendingTimestampStep,\n        );\n        this.seqOffset = offsets.seqOffset;\n        this.timestampOffset = offsets.timestampOffset;\n      }\n      this.rtpContinuityPending = false;\n      this.pendingTimestampStep = 1;\n    }\n\n    header.ssrc = this.ssrc;\n    header.payloadType = 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.headerBytesSent += header.serializeSize;\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              this.remotePacketsLost = report.packetsLost;\n              this.remoteFractionLost = report.fractionLost / 256;\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                  this.totalRoundTripTime += rtt;\n                  this.roundTripTimeMeasurements++;\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                this.nackCount++;\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                    this.retransmittedPacketsSent++;\n                    this.retransmittedBytesSent += packet.payload.length;\n                    this.headerBytesSent += packet.header.serializeSize;\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.pliCount++;\n                this.onPictureLossIndication.execute();\n              }\n              break;\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(rtcpPacket);\n  }\n\n  // todo impl\n  getParameters() {\n    return {\n      encodings: this.sendEncodings.map((encoding) => ({ ...encoding })),\n    };\n  }\n\n  // todo impl\n  setParameters(params: { encodings?: Array<Record<string, unknown>> }) {\n    if (params.encodings) {\n      this.setSendEncodings(params.encodings);\n    }\n  }\n\n  private get outboundRtpStatsId() {\n    return generateStatsId(\"outbound-rtp\", this.trackId);\n  }\n\n  private get mediaSourceStatsId() {\n    return generateStatsId(\"media-source\", this.trackId);\n  }\n\n  private get remoteInboundRtpStatsId() {\n    return generateStatsId(\"remote-inbound-rtp\", this.trackId);\n  }\n\n  getStatsRootIds() {\n    return [this.outboundRtpStatsId];\n  }\n\n  collectStats(timestamp: number): RTCStats[] {\n    const stats: RTCStats[] = [];\n    const transportId = this.dtlsTransport\n      ? generateStatsId(\"transport\", this.dtlsTransport.id)\n      : undefined;\n    const codecId =\n      this.codec && transportId\n        ? generateCodecStatsId(\n            transportId,\n            this.codec.payloadType,\n            this.trackId,\n          )\n        : undefined;\n\n    // Outbound RTP stats\n    const outboundRtpStats: RTCOutboundRtpStreamStats = {\n      type: \"outbound-rtp\",\n      id: this.outboundRtpStatsId,\n      timestamp,\n      ssrc: this.ssrc,\n      kind: this.kind,\n      transportId,\n      codecId,\n      mid: this.mid,\n      packetsSent: this.packetCount,\n      bytesSent: this.octetCount,\n      headerBytesSent: this.headerBytesSent,\n      retransmittedPacketsSent: this.retransmittedPacketsSent || undefined,\n      retransmittedBytesSent: this.retransmittedBytesSent || undefined,\n      rtxSsrc: this.rtxPayloadType ? this.rtxSsrc : undefined,\n      mediaSourceId: this.track ? this.mediaSourceStatsId : undefined,\n      remoteId:\n        this.rtt !== undefined ||\n        this.remotePacketsLost !== undefined ||\n        this.remoteFractionLost !== undefined\n          ? this.remoteInboundRtpStatsId\n          : undefined,\n      nackCount: this.nackCount || undefined,\n      pliCount: this.pliCount || undefined,\n      firCount: this.firCount || undefined,\n    };\n    stats.push(outboundRtpStats);\n\n    // Media source stats\n    if (this.track) {\n      const mediaSourceStats: RTCMediaSourceStats = {\n        type: \"media-source\",\n        id: this.mediaSourceStatsId,\n        timestamp,\n        trackIdentifier: this.track.id ?? this.trackId,\n        kind: this.kind,\n      };\n      stats.push(mediaSourceStats);\n    }\n\n    if (this.codec && transportId) {\n      const codecStats: RTCCodecStats = {\n        type: \"codec\",\n        id: codecId!,\n        timestamp,\n        payloadType: this.codec.payloadType,\n        transportId,\n        mimeType: this.codec.mimeType,\n        clockRate: this.codec.clockRate,\n        channels: this.codec.channels,\n        sdpFmtpLine: this.codec.parameters,\n      };\n      stats.push(codecStats);\n    }\n\n    // Remote inbound RTP stats (if we have RTT)\n    if (\n      this.rtt !== undefined ||\n      this.remotePacketsLost !== undefined ||\n      this.remoteFractionLost !== undefined\n    ) {\n      const remoteInboundStats: RTCRemoteInboundRtpStreamStats = {\n        type: \"remote-inbound-rtp\",\n        id: this.remoteInboundRtpStatsId,\n        timestamp,\n        ssrc: this.ssrc,\n        kind: this.kind,\n        transportId,\n        codecId: outboundRtpStats.codecId,\n        localId: outboundRtpStats.id,\n        roundTripTime: this.rtt,\n        totalRoundTripTime: this.totalRoundTripTime,\n        roundTripTimeMeasurements: this.roundTripTimeMeasurements,\n        packetsLost: this.remotePacketsLost,\n        fractionLost: this.remoteFractionLost,\n      };\n      stats.push(remoteInboundStats);\n    }\n\n    return stats;\n  }\n\n  async getStats(): Promise<RTCStatsReport> {\n    const timestamp = getStatsTimestamp();\n    const stats = this.collectStats(timestamp);\n\n    if (this.dtlsTransport) {\n      stats.push(...(await this.dtlsTransport.getStats(timestamp)));\n    }\n\n    return buildStatsReport(stats, this.getStatsRootIds());\n  }\n}\n"]}

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

@@ -24,9 +24,33 @@ export declare class RTCRtpTransceiver {
     set codecs(codecs: RTCRtpCodecParameters[]);
     get codecs(): RTCRtpCodecParameters[];
     headerExtensions: RTCRtpHeaderExtensionParameters[];
+    private _codecPreferences?;
+    readonly onCodecPreferencesChanged: Event<[]>;
+    pendingLocalOfferCodecs?: RTCRtpCodecParameters[];
+    codecPreferencesNeedResolution: boolean;
+    get codecPreferences(): readonly RTCRtpCodecParameters[] | undefined;
+    setCodecPreferences(codecs: RTCRtpCodecParameters[]): void;
     options: Partial<TransceiverOptions>;
+    /**stop() 済み、または停止が確定した transceiver */
     stopping: boolean;
+    /**port 0 の交渉が確定し、m-line が停止した transceiver */
     stopped: boolean;
+    /**
+     * 共通 codec がない / remote port 0 のため answer で拒否することが確定した。
+     * `inactive` や app の `stop()` とは区別し、確定後は `stopped` も true になる。
+     */
+    rejected: boolean;
+    /**
+     * remote offer の m-line を拒否予定 (answer 未確定)。
+     * 確定するまで既存の RTP pipeline / track は維持し、rollback で false に戻す。
+     */
+    pendingRejection: boolean;
+    /**@private remote から受信中として track event を通知済みか */
+    firedReceiving: boolean;
+    /**@private app の stop() 要求を manager に伝える */
+    readonly onStopRequested: Event<[]>;
+    /**@private 停止の確定または stop() でメディア資源を解放する際に通知する */
+    readonly onRelease: Event<[]>;
     constructor(kind: Kind, dtlsTransport: RTCDtlsTransport | undefined, receiver: RTCRtpReceiver, sender: RTCRtpSender, 
     /**RFC 8829 4.2.4.  direction the transceiver was initialized with */
     _direction: MediaDirection);
@@ -42,7 +66,23 @@ export declare class RTCRtpTransceiver {
     get msid(): string;
     get msids(): string[];
     addTrack(track: MediaStreamTrack): void;
+    /**m-line (MID / index) と関連付け済みか */
+    get associated(): boolean;
+    /**
+     * https://www.w3.org/TR/webrtc/#dom-rtcrtptransceiver-stop
+     * 送受信をただちに止めて資源を解放し、次の自分の offer で port 0 を交渉する。
+     * 冪等で、2 回目以降は何もしない。
+     */
     stop(): void;
+    /**
+     * @private
+     * port 0 の交渉確定 (自分の stop の answer / remote による拒否) を反映する。
+     */
+    commitStopped({ rejected }: {
+        rejected: boolean;
+    }): void;
+    private markStopped;
+    private releaseMedia;
     forceStop(): void;
     getPayloadType(mimeType: string): number | undefined;
     getCodecStats(): RTCStats[];
@@ -58,6 +98,8 @@ export type CurrentDirection = MediaDirection | "stopped";
 type SimulcastDirection = "send" | "recv";
 export interface RTCRtpEncodingParameters {
     active?: boolean;
+    rid?: string;
+    maxBitrate?: number;
 }
 export interface TransceiverOptions {
     direction: MediaDirection;

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

@@ -4,6 +4,8 @@ exports.Directions = exports.Sendrecv = exports.Recvonly = exports.Sendonly = ex
 const crypto_1 = require("crypto");
 const common_1 = require("../imports/common");
 const const_1 = require("../const");
+const errors_1 = require("../errors");
+const codec_1 = require("./codec");
 const stats_1 = require("./stats");
 class RTCRtpTransceiver {
     set codecs(codecs) {
@@ -12,6 +14,24 @@ class RTCRtpTransceiver {
     get codecs() {
         return this._codecs;
     }
+    get codecPreferences() {
+        return this._codecPreferences;
+    }
+    setCodecPreferences(codecs) {
+        if (!Array.isArray(codecs)) {
+            throw (0, errors_1.createWebRtcTypeError)("codecs must be an array");
+        }
+        const next = codecs.length === 0 ? undefined : codecs.map(codec_1.cloneCodecParameters);
+        if (sameCodecPreferences(this._codecPreferences, next))
+            return;
+        this._codecPreferences = next;
+        // 現在の交渉結果とは別に、次回 offer/answer 用の再解決要求を保持する。
+        // pending offer への answer が現在の codec を書き戻しても、この要求は
+        // 次回の createOffer() まで失われない。
+        this.codecPreferencesNeedResolution = true;
+        this._codecs = [];
+        this.onCodecPreferencesChanged.execute();
+    }
     constructor(kind, dtlsTransport, receiver, sender, 
     /**RFC 8829 4.2.4.  direction the transceiver was initialized with */
     _direction) {
@@ -94,24 +114,91 @@ class RTCRtpTransceiver {
             writable: true,
             value: []
         });
+        Object.defineProperty(this, "_codecPreferences", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "onCodecPreferencesChanged", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new common_1.Event()
+        });
+        Object.defineProperty(this, "pendingLocalOfferCodecs", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "codecPreferencesNeedResolution", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: false
+        });
         Object.defineProperty(this, "options", {
             enumerable: true,
             configurable: true,
             writable: true,
             value: {}
         });
+        /**stop() 済み、または停止が確定した transceiver */
         Object.defineProperty(this, "stopping", {
             enumerable: true,
             configurable: true,
             writable: true,
             value: false
         });
+        /**port 0 の交渉が確定し、m-line が停止した transceiver */
         Object.defineProperty(this, "stopped", {
             enumerable: true,
             configurable: true,
             writable: true,
             value: false
         });
+        /**
+         * 共通 codec がない / remote port 0 のため answer で拒否することが確定した。
+         * `inactive` や app の `stop()` とは区別し、確定後は `stopped` も true になる。
+         */
+        Object.defineProperty(this, "rejected", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: false
+        });
+        /**
+         * remote offer の m-line を拒否予定 (answer 未確定)。
+         * 確定するまで既存の RTP pipeline / track は維持し、rollback で false に戻す。
+         */
+        Object.defineProperty(this, "pendingRejection", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: false
+        });
+        /**@private remote から受信中として track event を通知済みか */
+        Object.defineProperty(this, "firedReceiving", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: false
+        });
+        /**@private app の stop() 要求を manager に伝える */
+        Object.defineProperty(this, "onStopRequested", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new common_1.Event()
+        });
+        /**@private 停止の確定または stop() でメディア資源を解放する際に通知する */
+        Object.defineProperty(this, "onRelease", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new common_1.Event()
+        });
         if (dtlsTransport) {
             this.setDtlsTransport(dtlsTransport);
         }
@@ -140,6 +227,13 @@ class RTCRtpTransceiver {
     }
     setCurrentDirection(direction) {
         this._currentDirection = direction;
+        if (direction &&
+            direction !== "stopped" &&
+            const_1.SenderDirections.includes(direction) &&
+            this.sender.track) {
+            // 実際に送信へ使われた sender は addTrack の自動再使用対象から外す
+            this.usedForSender = true;
+        }
     }
     setDtlsTransport(dtls) {
         this.receiver.setDtlsTransport(dtls);
@@ -157,17 +251,53 @@ class RTCRtpTransceiver {
             this.onTrack.execute(track, this);
         }
     }
-    // todo impl
-    // https://www.w3.org/TR/webrtc/#methods-8
+    /**m-line (MID / index) と関連付け済みか */
+    get associated() {
+        return this.mid != null && this.mLineIndex != undefined;
+    }
+    /**
+     * https://www.w3.org/TR/webrtc/#dom-rtcrtptransceiver-stop
+     * 送受信をただちに止めて資源を解放し、次の自分の offer で port 0 を交渉する。
+     * 冪等で、2 回目以降は何もしない。
+     */
     stop() {
         if (this.stopping) {
             return;
         }
-        // todo Stop sending and receiving with transceiver.
         this.stopping = true;
+        this.releaseMedia();
+        if (!this.associated) {
+            // m-line と未関連付けなら交渉対象の m-line を作らずに停止を確定する
+            this.markStopped();
+        }
+        this.onStopRequested.execute();
+    }
+    /**
+     * @private
+     * port 0 の交渉確定 (自分の stop の answer / remote による拒否) を反映する。
+     */
+    commitStopped({ rejected }) {
+        if (rejected && !this.stopping) {
+            this.rejected = true;
+        }
+        this.pendingRejection = false;
+        this.stopping = true;
+        this.releaseMedia();
+        this.markStopped();
+    }
+    markStopped() {
+        this.stopped = true;
+        this.setCurrentDirection("stopped");
+    }
+    releaseMedia() {
+        // W3C の stop() は sender.track を null にしないため参照は維持する
+        this.sender.stop({ keepTrack: true });
+        this.receiver.stop();
+        this.receiver.endTracks();
+        this.onRelease.execute();
     }
     forceStop() {
-        if (this.stopped) {
+        if (this.stopped && this.sender.stopped && this.receiver.stopped) {
             return;
         }
         this.stopping = true;
@@ -175,6 +305,7 @@ class RTCRtpTransceiver {
         this.setCurrentDirection("stopped");
         this.receiver.stop();
         this.sender.stop();
+        this.onRelease.execute();
     }
     getPayloadType(mimeType) {
         return this.codecs.find((codec) => codec.mimeType.toLowerCase().includes(mimeType.toLowerCase()))?.payloadType;
@@ -208,6 +339,28 @@ class RTCRtpTransceiver {
     }
 }
 exports.RTCRtpTransceiver = RTCRtpTransceiver;
+function sameCodecPreferences(left, right) {
+    if (left === right)
+        return true;
+    if (left == undefined || right == undefined || left.length !== right.length) {
+        return false;
+    }
+    return stableCodecValue(left) === stableCodecValue(right);
+}
+function stableCodecValue(value) {
+    if (Array.isArray(value)) {
+        return `[${value.map(stableCodecValue).join(",")}]`;
+    }
+    if (value != undefined && typeof value === "object") {
+        const entries = Object.entries(value)
+            .filter(([, item]) => item !== undefined)
+            .sort(([left], [right]) => left.localeCompare(right));
+        return `{${entries
+            .map(([key, item]) => `${JSON.stringify(key)}:${stableCodecValue(item)}`)
+            .join(",")}}`;
+    }
+    return JSON.stringify(value) ?? "undefined";
+}
 exports.Inactive = "inactive";
 exports.Sendonly = "sendonly";
 exports.Recvonly = "recvonly";

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

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{ randomUUID } from \"crypto\";\nimport { Event } from \"../imports/common\";\n\nimport type { RTCDtlsTransport } from \"..\";\nimport { SenderDirections } from \"../const\";\nimport type { Kind } from \"../types/domain\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n} from \"./parameters\";\nimport type { RTCRtpReceiver } from \"./rtpReceiver\";\nimport type { RTCRtpSender } from \"./rtpSender\";\nimport {\n  type RTCCodecStats,\n  type RTCStats,\n  generateCodecStatsId,\n  generateStatsId,\n  getStatsTimestamp,\n} from \"./stats\";\nimport type { MediaStream, MediaStreamTrack } from \"./track\";\n\nexport class RTCRtpTransceiver {\n  readonly id = randomUUID().toString();\n  readonly onTrack = new Event<[MediaStreamTrack, RTCRtpTransceiver]>();\n  mid: string | null = null;\n  mLineIndex?: number;\n  /**should not be reused because it has been used for sending before. */\n  usedForSender = false;\n  private _currentDirection?: CurrentDirection;\n  offerDirection!: MediaDirection;\n  _codecs: RTCRtpCodecParameters[] = [];\n  set codecs(codecs: RTCRtpCodecParameters[]) {\n    this._codecs = codecs;\n  }\n  get codecs() {\n    return this._codecs;\n  }\n  headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  options: Partial<TransceiverOptions> = {};\n  stopping = false;\n  stopped = false;\n\n  constructor(\n    public readonly kind: Kind,\n    dtlsTransport: RTCDtlsTransport | undefined,\n    public receiver: RTCRtpReceiver,\n    public sender: RTCRtpSender,\n    /**RFC 8829 4.2.4.  direction the transceiver was initialized with */\n    private _direction: MediaDirection,\n  ) {\n    if (dtlsTransport) {\n      this.setDtlsTransport(dtlsTransport);\n    }\n  }\n\n  get dtlsTransport() {\n    return this.receiver.dtlsTransport;\n  }\n\n  /**RFC 8829 4.2.4. setDirectionに渡された最後の値を示します */\n  get direction() {\n    return this._direction;\n  }\n\n  set direction(direction: MediaDirection) {\n    this.setDirection(direction);\n  }\n\n  setDirection(direction: MediaDirection) {\n    this._direction = direction;\n    if (\n      this._currentDirection &&\n      this._currentDirection !== \"stopped\" &&\n      SenderDirections.includes(this._currentDirection)\n    ) {\n      this.usedForSender = true;\n    }\n  }\n\n  /**RFC 8829 4.2.5. last negotiated direction */\n  get currentDirection(): CurrentDirection | null {\n    return this._currentDirection ?? null;\n  }\n\n  setCurrentDirection(direction: CurrentDirection | 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.msids[0];\n  }\n\n  get msids() {\n    return this.sender.streamIds.map(\n      (streamId) => `${streamId} ${this.sender.trackId}`,\n    );\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  forceStop() {\n    if (this.stopped) {\n      return;\n    }\n\n    this.stopping = true;\n    this.stopped = true;\n    this.setCurrentDirection(\"stopped\");\n    this.receiver.stop();\n    this.sender.stop();\n  }\n\n  getPayloadType(mimeType: string) {\n    return this.codecs.find((codec) =>\n      codec.mimeType.toLowerCase().includes(mimeType.toLowerCase()),\n    )?.payloadType;\n  }\n\n  getCodecStats(): RTCStats[] {\n    const timestamp = getStatsTimestamp();\n    return this.collectCodecStats(timestamp);\n  }\n\n  collectCodecStats(timestamp: number): RTCStats[] {\n    const stats: RTCStats[] = [];\n\n    if (!this.dtlsTransport) {\n      return stats;\n    }\n\n    const transportId = generateStatsId(\"transport\", this.dtlsTransport.id);\n\n    // Add codec stats for each codec\n    for (const codec of this.codecs) {\n      const codecStats: RTCCodecStats = {\n        type: \"codec\",\n        id: generateCodecStatsId(transportId, codec.payloadType, this.id),\n        timestamp,\n        payloadType: codec.payloadType,\n        transportId,\n        mimeType: codec.mimeType,\n        clockRate: codec.clockRate,\n        channels: codec.channels,\n        sdpFmtpLine: codec.parameters,\n      };\n      stats.push(codecStats);\n    }\n\n    return stats;\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 MediaDirection = (typeof Directions)[number];\nexport type CurrentDirection = MediaDirection | \"stopped\";\n\ntype SimulcastDirection = \"send\" | \"recv\";\n\nexport interface RTCRtpEncodingParameters {\n  active?: boolean;\n}\n\nexport interface TransceiverOptions {\n  direction: MediaDirection;\n  sendEncodings: RTCRtpEncodingParameters[];\n  simulcast: { direction: SimulcastDirection; rid: string }[];\n  streams: MediaStream[];\n}\n"]}
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{ randomUUID } from \"crypto\";\nimport { Event } from \"../imports/common\";\n\nimport type { RTCDtlsTransport } from \"..\";\nimport { SenderDirections } from \"../const\";\nimport { createWebRtcTypeError } from \"../errors\";\nimport type { Kind } from \"../types/domain\";\nimport { cloneCodecParameters } from \"./codec\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n} from \"./parameters\";\nimport type { RTCRtpReceiver } from \"./rtpReceiver\";\nimport type { RTCRtpSender } from \"./rtpSender\";\nimport {\n  type RTCCodecStats,\n  type RTCStats,\n  generateCodecStatsId,\n  generateStatsId,\n  getStatsTimestamp,\n} from \"./stats\";\nimport type { MediaStream, MediaStreamTrack } from \"./track\";\n\nexport class RTCRtpTransceiver {\n  readonly id = randomUUID().toString();\n  readonly onTrack = new Event<[MediaStreamTrack, RTCRtpTransceiver]>();\n  mid: string | null = null;\n  mLineIndex?: number;\n  /**should not be reused because it has been used 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(sameCodecPreferences(this._codecPreferences, next)) return;\n    this._codecPreferences = next;\n    // 現在の交渉結果とは別に、次回 offer/answer 用の再解決要求を保持する。\n    // pending offer への answer が現在の codec を書き戻しても、この要求は\n    // 次回の createOffer() まで失われない。\n    this.codecPreferencesNeedResolution = true;\n    this._codecs = [];\n    this.onCodecPreferencesChanged.execute();\n  }\n  options: Partial<TransceiverOptions> = {};\n  /**stop() 済み、または停止が確定した transceiver */\n  stopping = false;\n  /**port 0 の交渉が確定し、m-line が停止した transceiver */\n  stopped = false;\n  /**\n   * 共通 codec がない / remote port 0 のため answer で拒否することが確定した。\n   * `inactive` や app の `stop()` とは区別し、確定後は `stopped` も true になる。\n   */\n  rejected = false;\n  /**\n   * remote offer の m-line を拒否予定 (answer 未確定)。\n   * 確定するまで既存の RTP pipeline / track は維持し、rollback で false に戻す。\n   */\n  pendingRejection = false;\n  /**@private remote から受信中として track event を通知済みか */\n  firedReceiving = false;\n  /**@private app の stop() 要求を manager に伝える */\n  readonly onStopRequested = new Event<[]>();\n  /**@private 停止の確定または stop() でメディア資源を解放する際に通知する */\n  readonly onRelease = new Event<[]>();\n\n  constructor(\n    public readonly kind: Kind,\n    dtlsTransport: RTCDtlsTransport | undefined,\n    public receiver: RTCRtpReceiver,\n    public sender: RTCRtpSender,\n    /**RFC 8829 4.2.4.  direction the transceiver was initialized with */\n    private _direction: MediaDirection,\n  ) {\n    if (dtlsTransport) {\n      this.setDtlsTransport(dtlsTransport);\n    }\n  }\n\n  get dtlsTransport() {\n    return this.receiver.dtlsTransport;\n  }\n\n  /**RFC 8829 4.2.4. setDirectionに渡された最後の値を示します */\n  get direction() {\n    return this._direction;\n  }\n\n  set direction(direction: MediaDirection) {\n    this.setDirection(direction);\n  }\n\n  setDirection(direction: MediaDirection) {\n    this._direction = direction;\n    if (\n      this._currentDirection &&\n      this._currentDirection !== \"stopped\" &&\n      SenderDirections.includes(this._currentDirection)\n    ) {\n      this.usedForSender = true;\n    }\n  }\n\n  /**RFC 8829 4.2.5. last negotiated direction */\n  get currentDirection(): CurrentDirection | null {\n    return this._currentDirection ?? null;\n  }\n\n  setCurrentDirection(direction: CurrentDirection | undefined) {\n    this._currentDirection = direction;\n    if (\n      direction &&\n      direction !== \"stopped\" &&\n      SenderDirections.includes(direction) &&\n      this.sender.track\n    ) {\n      // 実際に送信へ使われた sender は addTrack の自動再使用対象から外す\n      this.usedForSender = true;\n    }\n  }\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.receiver.setDtlsTransport(dtls);\n    this.sender.setDtlsTransport(dtls);\n  }\n\n  get msid() {\n    return this.msids[0];\n  }\n\n  get msids() {\n    return this.sender.streamIds.map(\n      (streamId) => `${streamId} ${this.sender.trackId}`,\n    );\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  /**m-line (MID / index) と関連付け済みか */\n  get associated() {\n    return this.mid != null && this.mLineIndex != undefined;\n  }\n\n  /**\n   * https://www.w3.org/TR/webrtc/#dom-rtcrtptransceiver-stop\n   * 送受信をただちに止めて資源を解放し、次の自分の offer で port 0 を交渉する。\n   * 冪等で、2 回目以降は何もしない。\n   */\n  stop() {\n    if (this.stopping) {\n      return;\n    }\n\n    this.stopping = true;\n    this.releaseMedia();\n    if (!this.associated) {\n      // m-line と未関連付けなら交渉対象の m-line を作らずに停止を確定する\n      this.markStopped();\n    }\n    this.onStopRequested.execute();\n  }\n\n  /**\n   * @private\n   * port 0 の交渉確定 (自分の stop の answer / remote による拒否) を反映する。\n   */\n  commitStopped({ rejected }: { rejected: boolean }) {\n    if (rejected && !this.stopping) {\n      this.rejected = true;\n    }\n    this.pendingRejection = false;\n    this.stopping = true;\n    this.releaseMedia();\n    this.markStopped();\n  }\n\n  private markStopped() {\n    this.stopped = true;\n    this.setCurrentDirection(\"stopped\");\n  }\n\n  private releaseMedia() {\n    // W3C の stop() は sender.track を null にしないため参照は維持する\n    this.sender.stop({ keepTrack: true });\n    this.receiver.stop();\n    this.receiver.endTracks();\n    this.onRelease.execute();\n  }\n\n  forceStop() {\n    if (this.stopped && this.sender.stopped && this.receiver.stopped) {\n      return;\n    }\n\n    this.stopping = true;\n    this.stopped = true;\n    this.setCurrentDirection(\"stopped\");\n    this.receiver.stop();\n    this.sender.stop();\n    this.onRelease.execute();\n  }\n\n  getPayloadType(mimeType: string) {\n    return this.codecs.find((codec) =>\n      codec.mimeType.toLowerCase().includes(mimeType.toLowerCase()),\n    )?.payloadType;\n  }\n\n  getCodecStats(): RTCStats[] {\n    const timestamp = getStatsTimestamp();\n    return this.collectCodecStats(timestamp);\n  }\n\n  collectCodecStats(timestamp: number): RTCStats[] {\n    const stats: RTCStats[] = [];\n\n    if (!this.dtlsTransport) {\n      return stats;\n    }\n\n    const transportId = generateStatsId(\"transport\", this.dtlsTransport.id);\n\n    // Add codec stats for each codec\n    for (const codec of this.codecs) {\n      const codecStats: RTCCodecStats = {\n        type: \"codec\",\n        id: generateCodecStatsId(transportId, codec.payloadType, this.id),\n        timestamp,\n        payloadType: codec.payloadType,\n        transportId,\n        mimeType: codec.mimeType,\n        clockRate: codec.clockRate,\n        channels: codec.channels,\n        sdpFmtpLine: codec.parameters,\n      };\n      stats.push(codecStats);\n    }\n\n    return stats;\n  }\n}\n\nfunction sameCodecPreferences(\n  left: readonly RTCRtpCodecParameters[] | undefined,\n  right: readonly RTCRtpCodecParameters[] | undefined,\n) {\n  if (left === right) return true;\n  if (left == undefined || right == undefined || left.length !== right.length) {\n    return false;\n  }\n  return stableCodecValue(left) === stableCodecValue(right);\n}\n\nfunction stableCodecValue(value: unknown): string {\n  if (Array.isArray(value)) {\n    return `[${value.map(stableCodecValue).join(\",\")}]`;\n  }\n  if (value != undefined && typeof value === \"object\") {\n    const entries = Object.entries(value as Record<string, unknown>)\n      .filter(([, item]) => item !== undefined)\n      .sort(([left], [right]) => left.localeCompare(right));\n    return `{${entries\n      .map(([key, item]) => `${JSON.stringify(key)}:${stableCodecValue(item)}`)\n      .join(\",\")}}`;\n  }\n  return JSON.stringify(value) ?? \"undefined\";\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 MediaDirection = (typeof Directions)[number];\nexport type CurrentDirection = MediaDirection | \"stopped\";\n\ntype SimulcastDirection = \"send\" | \"recv\";\n\nexport interface RTCRtpEncodingParameters {\n  active?: boolean;\n  rid?: string;\n  maxBitrate?: number;\n}\n\nexport interface TransceiverOptions {\n  direction: MediaDirection;\n  sendEncodings: RTCRtpEncodingParameters[];\n  simulcast: { direction: SimulcastDirection; rid: string }[];\n  streams: MediaStream[];\n}\n"]}

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

@@ -3,6 +3,20 @@ import { EventTarget } from "../helper";
 import { type Extensions, type RtcpPacket, type RtpHeader, RtpPacket } from "../imports/rtp";
 import type { Kind } from "../types/domain";
 import type { RTCRtpCodecParameters } from "./parameters";
+export declare function setTrackSourceCodecs(track: MediaStreamTrack, codecs: readonly RTCRtpCodecParameters[] | undefined): void;
+export declare function getTrackSourceCodecs(track: MediaStreamTrack | undefined | null): readonly RTCRtpCodecParameters[] | undefined;
+export declare function captureTrackSourceCodecs(track: MediaStreamTrack): void;
+declare class TrackBroadcastSource {
+    private readonly tracks;
+    private readonly upstreamStops;
+    attach(track: MediaStreamTrack): void;
+    detach(track: MediaStreamTrack): void;
+    addUpstreamStop(stop: () => void): void;
+    deliverRtp(packet: RtpPacket, extensions?: Extensions): void;
+    deliverRtcp(packet: RtcpPacket): void;
+    deliverSourceChanged(header: Pick<RtpHeader, "sequenceNumber" | "timestamp">): void;
+    stopAllTracks(): void;
+}
 export declare class MediaStreamTrack extends EventTarget {
     readonly uuid: string;
     /**MediaStream ID*/
@@ -23,9 +37,19 @@ export declare class MediaStreamTrack extends EventTarget {
     readonly onSourceChanged: Event<[Pick<RtpHeader, "sequenceNumber" | "timestamp">]>;
     stopped: boolean;
     muted: boolean;
-    constructor(props: Partial<MediaStreamTrack> & Pick<MediaStreamTrack, "kind">);
+    private broadcastSource;
+    constructor(props: Partial<MediaStreamTrack> & Pick<MediaStreamTrack, "kind"> & {
+        broadcastSource?: TrackBroadcastSource;
+    });
+    get readyState(): "live" | "ended";
     stop: () => void;
     writeRtp: (rtp: RtpPacket | Buffer) => void;
+    writeRtcp: (rtcp: RtcpPacket) => void;
+    notifySourceChanged(header: Pick<RtpHeader, "sequenceNumber" | "timestamp">): void;
+    applyIncomingRtp(packet: RtpPacket, extensions?: Extensions): void;
+    bindUpstreamStop(stop: () => void): void;
+    stopMediaSource(): void;
+    clone(): MediaStreamTrack;
 }
 export declare class MediaStream {
     id: string;
@@ -36,4 +60,8 @@ export declare class MediaStream {
     getTracks(): MediaStreamTrack[];
     getAudioTracks(): MediaStreamTrack[];
     getVideoTracks(): MediaStreamTrack[];
+    getTrackById(id: string): MediaStreamTrack | undefined;
+    get active(): boolean;
+    clone(): MediaStream;
 }
+export {};