werift 0.15.2 → 0.15.5

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

@@ -1 +1 @@
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{ debug } from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  Red,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { Nack } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { RedHandler } from \"./receiver/red\";\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 readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new Nack(this);\n  private readonly redHandler = 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 senderReport\n  readonly lsr: { [ssrc: number]: number } = {};\n  readonly lsrTime: { [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  supportTWCC = false;\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(public kind: Kind, public rtcpSsrc: number) {}\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  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    this.supportTWCC = !!Object.values(this.codecs).find((codec) =>\n      codec.rtcpFeedback.find((v) => v.type === \"transport-cc\")\n    );\n    log(\"twcc support\", this.supportTWCC);\n\n    if (this.supportTWCC && mediaSourceSsrc) {\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) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\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 lsr = 0,\n              dlsr = 0;\n            if (this.lsr[ssrc]) {\n              lsr = this.lsr[ssrc];\n              const delay = Date.now() / 1000 - this.lsrTime[ssrc];\n              if (delay > 0 && delay < 65536) {\n                dlsr = int(delay * 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,\n              dlsr,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\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    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.lsr[sr.ssrc] = compactNtp(sr.senderInfo.ntpTimestamp);\n          this.lsrTime[sr.ssrc] = Date.now() / 1000;\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) return;\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 number;\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.supportTWCC) {\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    }\n\n    // todo fix select use or not use nack\n    if (track?.kind === \"video\") {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        const payloads = this.redHandler.push(red, packet);\n        for (const packet of payloads) {\n          track.onReceiveRtp.execute(packet.clone());\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.slice(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.slice(2)\n  );\n  return packet;\n}\n"]}
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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 readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new Nack(this);\n  private readonly redHandler = 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 senderReport\n  readonly lsr: { [ssrc: number]: number } = {};\n  readonly lsrTime: { [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  supportTWCC = false;\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(public kind: Kind, public rtcpSsrc: number) {}\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  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    this.supportTWCC = !!Object.values(this.codecs).find((codec) =>\n      codec.rtcpFeedback.find((v) => v.type === \"transport-cc\")\n    );\n    log(\"twcc support\", this.supportTWCC);\n\n    if (this.supportTWCC && mediaSourceSsrc) {\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) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\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 lsr = 0,\n              dlsr = 0;\n            if (this.lsr[ssrc]) {\n              lsr = this.lsr[ssrc];\n              const delay = Date.now() / 1000 - this.lsrTime[ssrc];\n              if (delay > 0 && delay < 65536) {\n                dlsr = int(delay * 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,\n              dlsr,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\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    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.lsr[sr.ssrc] = compactNtp(sr.senderInfo.ntpTimestamp);\n          this.lsrTime[sr.ssrc] = Date.now() / 1000;\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) return;\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 number;\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.supportTWCC) {\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    }\n\n    // todo fix select use or not use nack\n    if (track?.kind === \"video\") {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        const payloads = this.redHandler.push(red, packet);\n        for (const packet of payloads) {\n          track.onReceiveRtp.execute(packet.clone());\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.slice(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.slice(2)\n  );\n  return packet;\n}\n"]}

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

@@ -1,7 +1,11 @@
 "use strict";
 var __createBinding = (this && this.__createBinding) || (Object.create ? (function(o, m, k, k2) {
     if (k2 === undefined) k2 = k;
-    Object.defineProperty(o, k2, { enumerable: true, get: function() { return m[k]; } });
+    var desc = Object.getOwnPropertyDescriptor(m, k);
+    if (!desc || ("get" in desc ? !m.__esModule : desc.writable || desc.configurable)) {
+      desc = { enumerable: true, get: function() { return m[k]; } };
+    }
+    Object.defineProperty(o, k2, desc);
 }) : (function(o, m, k, k2) {
     if (k2 === undefined) k2 = k;
     o[k2] = m[k];

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

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{ randomBytes } from \"crypto\";\nimport debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport * as uuid from \"uuid\";\n\nimport {\n  bufferWriter,\n  random16,\n  uint16Add,\n  uint32Add,\n} from \"../../../common/src\";\nimport {\n  Extension,\n  GenericNack,\n  PictureLossIndication,\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} from \"../../../rtp/src\";\nimport { codecParametersFromString } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\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 =\n    typeof this.trackOrKind === \"string\"\n      ? this.trackOrKind\n      : this.trackOrKind.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 lsr?: number;\n  private lsrTime: number = Date.now() / 1000;\n  private ntpTimestamp = 0n;\n  private rtpTimestamp = 0;\n  private octetCount = 0;\n  private packetCount = 0;\n  private rtt?: number;\n  receiverEstimatedMaxBitrate: bigint = 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    if (trackOrKind instanceof MediaStreamTrack) {\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\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.lsr = compactNtp(this.ntpTimestamp);\n        this.lsrTime = Date.now() / 1000;\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    sequenceNumber,\n    timestamp,\n  }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = uint32Add(this.timestamp, -timestamp);\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 = rtp.header;\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    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 Buffer.from(this.mid);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.sdesRTPStreamID:\n              if (this.rtpStreamId) {\n                return Buffer.from(this.rtpStreamId);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.repairedRtpStreamId:\n              if (this.repairedRtpStreamId) {\n                return Buffer.from(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 bufferWriter(\n                [2],\n                [this.dtlsTransport.transportSequenceNumber]\n              );\n            case RTP_EXTENSION_URI.absSendTime:\n              const buf = Buffer.alloc(3);\n              const time = (ntpTime() >> 14n) & 0x00ffffffn;\n              buf.writeUIntBE(Number(time), 0, 3);\n              return buf;\n          }\n        })();\n\n        if (payload) return { id: extension.id, payload };\n      })\n      .filter((v) => v) as Extension[];\n\n    this.ntpTimestamp = ntpTime();\n    this.rtpTimestamp = rtp.header.timestamp;\n    this.octetCount += rtp.payload.length;\n    this.packetCount = uint32Add(this.packetCount, 1);\n\n    rtp.header = header;\n\n    this.rtpCache.push(rtp);\n    this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);\n\n    let rtpPayload = rtp.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 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.lsr === report.lsr && report.dlsr) {\n                const rtt =\n                  Date.now() / 1000 - this.lsrTime - 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        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 = this.rtpCache.find(\n                    (rtp) => rtp.header.sequenceNumber === seqNum\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\nexport function wrapRtx(\n  packet: RtpPacket,\n  payloadType: number,\n  sequenceNumber: number,\n  ssrc: number\n) {\n  const rtx = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: packet.header.marker,\n      sequenceNumber,\n      timestamp: packet.header.timestamp,\n      ssrc,\n      csrc: packet.header.csrc,\n      extensions: packet.header.extensions,\n    }),\n    Buffer.concat([\n      Buffer.from(jspack.Pack(\"!H\", [packet.header.sequenceNumber])),\n      packet.payload,\n    ])\n  );\n  return rtx;\n}\n"]}
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{ randomBytes } from \"crypto\";\nimport debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport * as uuid from \"uuid\";\n\nimport {\n  bufferWriter,\n  random16,\n  uint16Add,\n  uint32Add,\n} from \"../../../common/src\";\nimport {\n  Extension,\n  GenericNack,\n  PictureLossIndication,\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} from \"../../../rtp/src\";\nimport { codecParametersFromString } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\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 =\n    typeof this.trackOrKind === \"string\"\n      ? this.trackOrKind\n      : this.trackOrKind.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 lsr?: number;\n  private lsrTime: number = Date.now() / 1000;\n  private ntpTimestamp = 0n;\n  private rtpTimestamp = 0;\n  private octetCount = 0;\n  private packetCount = 0;\n  private rtt?: number;\n  receiverEstimatedMaxBitrate: bigint = 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    if (trackOrKind instanceof MediaStreamTrack) {\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\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.lsr = compactNtp(this.ntpTimestamp);\n        this.lsrTime = Date.now() / 1000;\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    sequenceNumber,\n    timestamp,\n  }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = uint32Add(this.timestamp, -timestamp);\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 = rtp.header;\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    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 Buffer.from(this.mid);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.sdesRTPStreamID:\n              if (this.rtpStreamId) {\n                return Buffer.from(this.rtpStreamId);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.repairedRtpStreamId:\n              if (this.repairedRtpStreamId) {\n                return Buffer.from(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 bufferWriter(\n                [2],\n                [this.dtlsTransport.transportSequenceNumber]\n              );\n            case RTP_EXTENSION_URI.absSendTime:\n              const buf = Buffer.alloc(3);\n              const time = (ntpTime() >> 14n) & 0x00ffffffn;\n              buf.writeUIntBE(Number(time), 0, 3);\n              return buf;\n          }\n        })();\n\n        if (payload) return { id: extension.id, payload };\n      })\n      .filter((v) => v) as Extension[];\n\n    this.ntpTimestamp = ntpTime();\n    this.rtpTimestamp = rtp.header.timestamp;\n    this.octetCount += rtp.payload.length;\n    this.packetCount = uint32Add(this.packetCount, 1);\n\n    rtp.header = header;\n\n    this.rtpCache.push(rtp);\n    this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);\n\n    let rtpPayload = rtp.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 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.lsr === report.lsr && report.dlsr) {\n                const rtt =\n                  Date.now() / 1000 - this.lsrTime - 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        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 = this.rtpCache.find(\n                    (rtp) => rtp.header.sequenceNumber === seqNum\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\nexport function wrapRtx(\n  packet: RtpPacket,\n  payloadType: number,\n  sequenceNumber: number,\n  ssrc: number\n) {\n  const rtx = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: packet.header.marker,\n      sequenceNumber,\n      timestamp: packet.header.timestamp,\n      ssrc,\n      csrc: packet.header.csrc,\n      extensions: packet.header.extensions,\n    }),\n    Buffer.concat([\n      Buffer.from(jspack.Pack(\"!H\", [packet.header.sequenceNumber])),\n      packet.payload,\n    ])\n  );\n  return rtx;\n}\n"]}

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

@@ -21,7 +21,9 @@ export declare class RTCRtpTransceiver {
     /**RFC 8829 4.2.5. last negotiated direction */
     get currentDirection(): Direction | "stopped" | undefined;
     offerDirection: Direction;
-    codecs: RTCRtpCodecParameters[];
+    _codecs: RTCRtpCodecParameters[];
+    set codecs(codecs: RTCRtpCodecParameters[]);
+    get codecs(): RTCRtpCodecParameters[];
     headerExtensions: RTCRtpHeaderExtensionParameters[];
     options: Partial<TransceiverOptions>;
     stopping: boolean;

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

@@ -1,7 +1,11 @@
 "use strict";
 var __createBinding = (this && this.__createBinding) || (Object.create ? (function(o, m, k, k2) {
     if (k2 === undefined) k2 = k;
-    Object.defineProperty(o, k2, { enumerable: true, get: function() { return m[k]; } });
+    var desc = Object.getOwnPropertyDescriptor(m, k);
+    if (!desc || ("get" in desc ? !m.__esModule : desc.writable || desc.configurable)) {
+      desc = { enumerable: true, get: function() { return m[k]; } };
+    }
+    Object.defineProperty(o, k2, desc);
 }) : (function(o, m, k, k2) {
     if (k2 === undefined) k2 = k;
     o[k2] = m[k];
@@ -37,7 +41,7 @@ class RTCRtpTransceiver {
         this.uuid = uuid.v4();
         this.onTrack = new rx_mini_1.default();
         this.usedForSender = false;
-        this.codecs = [];
+        this._codecs = [];
         this.headerExtensions = [];
         this.options = {};
         this.stopping = false;
@@ -54,6 +58,12 @@ class RTCRtpTransceiver {
     get currentDirection() {
         return this._currentDirection;
     }
+    set codecs(codecs) {
+        this._codecs = codecs;
+    }
+    get codecs() {
+        return this._codecs;
+    }
     get dtlsTransport() {
         return this.receiver.dtlsTransport;
     }

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

@@ -1 +1 @@
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+{"version":3,"file":"rtpTransceiver.js","sourceRoot":"","sources":["../../../../src/media/rtpTransceiver.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;;;;;;;;AAAA,sDAA4B;AAC5B,2CAA6B;AAG7B,oCAA4C;AAU5C,MAAa,iBAAiB;IA+B5B,YACkB,IAAU,EAC1B,aAA+B,EACxB,QAAwB,EACxB,MAAoB;IAC3B,qEAAqE;IAC9D,SAAoB;QALX,SAAI,GAAJ,IAAI,CAAM;QAEnB,aAAQ,GAAR,QAAQ,CAAgB;QACxB,WAAM,GAAN,MAAM,CAAc;QAEpB,cAAS,GAAT,SAAS,CAAW;QApCpB,SAAI,GAAG,IAAI,CAAC,EAAE,EAAE,CAAC;QACjB,YAAO,GAAG,IAAI,iBAAK,EAAsB,CAAC;QAGnD,kBAAa,GAAG,KAAK,CAAC;QActB,YAAO,GAA4B,EAAE,CAAC;QAOtC,qBAAgB,GAAsC,EAAE,CAAC;QACzD,YAAO,GAAgC,EAAE,CAAC;QAC1C,aAAQ,GAAG,KAAK,CAAC;QACjB,YAAO,GAAG,KAAK,CAAC;QAUd,IAAI,CAAC,gBAAgB,CAAC,aAAa,CAAC,CAAC;IACvC,CAAC;IAjCD,IAAI,gBAAgB,CAAC,SAA4C;QAC/D,IAAI,CAAC,iBAAiB,GAAG,SAAS,CAAC;QACnC,IAAI,wBAAgB,CAAC,QAAQ,CAAC,IAAI,CAAC,iBAAiB,IAAI,EAAE,CAAC,EAAE;YAC3D,IAAI,CAAC,aAAa,GAAG,IAAI,CAAC;SAC3B;IACH,CAAC;IACD,+CAA+C;IAC/C,IAAI,gBAAgB;QAClB,OAAO,IAAI,CAAC,iBAAiB,CAAC;IAChC,CAAC;IAID,IAAI,MAAM,CAAC,MAA+B;QACxC,IAAI,CAAC,OAAO,GAAG,MAAM,CAAC;IACxB,CAAC;IACD,IAAI,MAAM;QACR,OAAO,IAAI,CAAC,OAAO,CAAC;IACtB,CAAC;IAiBD,IAAI,aAAa;QACf,OAAO,IAAI,CAAC,QAAQ,CAAC,aAAa,CAAC;IACrC,CAAC;IAED,gBAAgB,CAAC,IAAsB;QACrC,IAAI,CAAC,QAAQ,CAAC,gBAAgB,CAAC,IAAI,CAAC,CAAC;QACrC,IAAI,CAAC,MAAM,CAAC,gBAAgB,CAAC,IAAI,CAAC,CAAC;IACrC,CAAC;IAED,IAAI,IAAI;QACN,OAAO,GAAG,IAAI,CAAC,MAAM,CAAC,QAAQ,IAAI,IAAI,CAAC,MAAM,CAAC,OAAO,EAAE,CAAC;IAC1D,CAAC;IAED,QAAQ,CAAC,KAAuB;QAC9B,MAAM,GAAG,GAAG,IAAI,CAAC,QAAQ,CAAC,QAAQ,CAAC,KAAK,CAAC,CAAC;QAC1C,IAAI,GAAG;YAAE,IAAI,CAAC,OAAO,CAAC,OAAO,CAAC,KAAK,CAAC,CAAC;IACvC,CAAC;IAED,YAAY;IACZ,0CAA0C;IAC1C,IAAI;QACF,IAAI,IAAI,CAAC,QAAQ;YAAE,OAAO;QAE1B,oDAAoD;QAEpD,IAAI,CAAC,QAAQ,GAAG,IAAI,CAAC;IACvB,CAAC;IAED,cAAc,CAAC,QAAgB;QAC7B,OAAO,IAAI,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC,KAAK,EAAE,EAAE,CAChC,KAAK,CAAC,QAAQ,CAAC,WAAW,EAAE,CAAC,QAAQ,CAAC,QAAQ,CAAC,WAAW,EAAE,CAAC,CAC9D,EAAE,WAAW,CAAC;IACjB,CAAC;CACF;AA3ED,8CA2EC;AAEY,QAAA,UAAU,GAAG;IACxB,UAAU;IACV,UAAU;IACV,UAAU;IACV,UAAU;CACF,CAAC","sourcesContent":["import Event from \"rx.mini\";\nimport * as uuid from \"uuid\";\n\nimport { RTCDtlsTransport } from \"..\";\nimport { SenderDirections } from \"../const\";\nimport { Kind } from \"../types/domain\";\nimport {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport { RTCRtpSender } from \"./rtpSender\";\nimport { MediaStreamTrack } from \"./track\";\n\nexport class RTCRtpTransceiver {\n  readonly uuid = uuid.v4();\n  readonly onTrack = new Event<[MediaStreamTrack]>();\n  mid?: string;\n  mLineIndex?: number;\n  usedForSender = false;\n  private _currentDirection?: Direction | \"stopped\";\n  set currentDirection(direction: Direction | \"stopped\" | undefined) {\n    this._currentDirection = direction;\n    if (SenderDirections.includes(this._currentDirection || \"\")) {\n      this.usedForSender = true;\n    }\n  }\n  /**RFC 8829 4.2.5. last negotiated direction */\n  get currentDirection(): Direction | \"stopped\" | undefined {\n    return this._currentDirection;\n  }\n\n  offerDirection!: Direction;\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,\n    public receiver: RTCRtpReceiver,\n    public sender: RTCRtpSender,\n    /**RFC 8829 4.2.4.  direction the transceiver was initialized with */\n    public direction: Direction\n  ) {\n    this.setDtlsTransport(dtlsTransport);\n  }\n\n  get dtlsTransport() {\n    return this.receiver.dtlsTransport;\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) this.onTrack.execute(track);\n  }\n\n  // todo impl\n  // https://www.w3.org/TR/webrtc/#methods-8\n  stop() {\n    if (this.stopping) return;\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 Directions = [\n  \"inactive\",\n  \"sendonly\",\n  \"recvonly\",\n  \"sendrecv\",\n] 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/nonstandard/recorder/writer/webm.js

@@ -1,7 +1,11 @@
 "use strict";
 var __createBinding = (this && this.__createBinding) || (Object.create ? (function(o, m, k, k2) {
     if (k2 === undefined) k2 = k;
-    Object.defineProperty(o, k2, { enumerable: true, get: function() { return m[k]; } });
+    var desc = Object.getOwnPropertyDescriptor(m, k);
+    if (!desc || ("get" in desc ? !m.__esModule : desc.writable || desc.configurable)) {
+      desc = { enumerable: true, get: function() { return m[k]; } };
+    }
+    Object.defineProperty(o, k2, desc);
 }) : (function(o, m, k, k2) {
     if (k2 === undefined) k2 = k;
     o[k2] = m[k];

package/lib/webrtc/src/nonstandard/recorder/writer/webm.js.map

@@ -1 +1 @@
-{"version":3,"file":"webm.js","sourceRoot":"","sources":["../../../../../../src/nonstandard/recorder/writer/webm.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;AAAA,gDAAkC;AAGlC,gCAKkB;AAClB,wBAAgC;AAEhC,MAAa,WAAY,SAAQ,cAAW;IAG1C,KAAK,CAAC,MAA0B;QAC9B,IAAI,CAAC,IAAI,GAAG,IAAI,cAAU,CACxB,EAAE,EACF,IAAI,CAAC,IAAI,EACT,MAAM,CAAC,GAAG,CAAC,CAAC,KAAK,EAAE,CAAC,EAAE,EAAE;YACtB,MAAM,WAAW,GAAG,CAAC,GAAG,CAAC,CAAC;YAC1B,MAAM,WAAW,GAAG,KAAK,CAAC,KAAM,CAAC,WAAW,CAAC;YAE7C,IAAI,KAAK,CAAC,IAAI,KAAK,OAAO,EAAE;gBAC1B,MAAM,KAAK,GAAG,CAAC,GAAmB,EAAE;oBAClC,QAAQ,KAAK,CAAC,KAAK,EAAE,IAAI,CAAC,WAAW,EAAyB,EAAE;wBAC9D,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,MAAM;4BACT,OAAO,eAAe,CAAC;wBACzB,KAAK,MAAM;4BACT,OAAO,KAAK,CAAC;wBACf;4BACE,MAAM,IAAI,KAAK,EAAE,CAAC;qBACrB;gBACH,CAAC,CAAC,EAAE,CAAC;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK;oBACL,KAAK,EAAE,IAAI,CAAC,OAAO,CAAC,KAAK;oBACzB,MAAM,EAAE,IAAI,CAAC,OAAO,CAAC,MAAM;iBAC5B,CAAC;aACH;iBAAM;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK,EAAE,MAAM;iBACd,CAAC;aACH;QACH,CAAC,CAAC,CACH,CAAC;QAEF,MAAM,CAAC,OAAO,CAAC,CAAC,KAAK,EAAE,EAAE;YACvB,MAAM,aAAa,GACjB,KAAK,CAAC,IAAI,KAAK,OAAO;gBACpB,CAAC,CAAC,IAAI,iBAAa,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,MAAM,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC;gBACvD,CAAC,CAAC,IAAI,iBAAa,CAAC,GAAG,EAAE,CAAC,IAAI,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC,CAAC;YACrD,IAAI,gBAAY,CAAC;gBACf,SAAS,EAAE,KAAK,CAAC,YAAY;gBAC7B,UAAU,EAAE,KAAK,CAAC,aAAa;aAChC,CAAC,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC;QACzB,CAAC,CAAC,CAAC;IACL,CAAC;IAED,KAAK,CAAC,IAAI;QACR,MAAM,IAAI,CAAC,IAAK,CAAC,IAAI,EAAE,CAAC;IAC1B,CAAC;CACF;AA9DD,kCA8DC;AAED,MAAM,oBAAoB,GAAG,CAAC,MAAM,EAAE,KAAK,EAAE,KAAK,EAAE,MAAM,CAAU,CAAC","sourcesContent":["import * as fs from \"fs/promises\";\n\nimport { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n  JitterBuffer,\n  MediaStreamTrack,\n  SampleBuilder,\n  WebmOutput,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nexport class WebmFactory extends MediaWriter {\n  webm?: WebmOutput;\n\n  start(tracks: MediaStreamTrack[]) {\n    this.webm = new WebmOutput(\n      fs,\n      this.path,\n      tracks.map((track, i) => {\n        const trackNumber = i + 1;\n        const payloadType = track.codec!.payloadType;\n\n        if (track.kind === \"video\") {\n          const codec = ((): SupportedCodec => {\n            switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n              case \"vp8\":\n                return \"VP8\";\n              case \"vp9\":\n                return \"VP9\";\n              case \"h264\":\n                return \"MPEG4/ISO/AVC\";\n              case \"av1x\":\n                return \"AV1\";\n              default:\n                throw new Error();\n            }\n          })();\n          return {\n            kind: \"video\",\n            clockRate: 90000,\n            payloadType,\n            trackNumber,\n            codec,\n            width: this.options.width,\n            height: this.options.height,\n          };\n        } else {\n          return {\n            kind: \"audio\",\n            clockRate: 48000,\n            payloadType,\n            trackNumber,\n            codec: \"OPUS\",\n          };\n        }\n      })\n    );\n\n    tracks.forEach((track) => {\n      const sampleBuilder =\n        track.kind === \"video\"\n          ? new SampleBuilder((h) => !!h.marker).pipe(this.webm!)\n          : new SampleBuilder(() => true).pipe(this.webm!);\n      new JitterBuffer({\n        rtpStream: track.onReceiveRtp,\n        rtcpStream: track.onReceiveRtcp,\n      }).pipe(sampleBuilder);\n    });\n  }\n\n  async stop() {\n    await this.webm!.stop();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}
+{"version":3,"file":"webm.js","sourceRoot":"","sources":["../../../../../../src/nonstandard/recorder/writer/webm.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;;;;;AAAA,gDAAkC;AAGlC,gCAKkB;AAClB,wBAAgC;AAEhC,MAAa,WAAY,SAAQ,cAAW;IAG1C,KAAK,CAAC,MAA0B;QAC9B,IAAI,CAAC,IAAI,GAAG,IAAI,cAAU,CACxB,EAAE,EACF,IAAI,CAAC,IAAI,EACT,MAAM,CAAC,GAAG,CAAC,CAAC,KAAK,EAAE,CAAC,EAAE,EAAE;YACtB,MAAM,WAAW,GAAG,CAAC,GAAG,CAAC,CAAC;YAC1B,MAAM,WAAW,GAAG,KAAK,CAAC,KAAM,CAAC,WAAW,CAAC;YAE7C,IAAI,KAAK,CAAC,IAAI,KAAK,OAAO,EAAE;gBAC1B,MAAM,KAAK,GAAG,CAAC,GAAmB,EAAE;oBAClC,QAAQ,KAAK,CAAC,KAAK,EAAE,IAAI,CAAC,WAAW,EAAyB,EAAE;wBAC9D,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,MAAM;4BACT,OAAO,eAAe,CAAC;wBACzB,KAAK,MAAM;4BACT,OAAO,KAAK,CAAC;wBACf;4BACE,MAAM,IAAI,KAAK,EAAE,CAAC;qBACrB;gBACH,CAAC,CAAC,EAAE,CAAC;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK;oBACL,KAAK,EAAE,IAAI,CAAC,OAAO,CAAC,KAAK;oBACzB,MAAM,EAAE,IAAI,CAAC,OAAO,CAAC,MAAM;iBAC5B,CAAC;aACH;iBAAM;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK,EAAE,MAAM;iBACd,CAAC;aACH;QACH,CAAC,CAAC,CACH,CAAC;QAEF,MAAM,CAAC,OAAO,CAAC,CAAC,KAAK,EAAE,EAAE;YACvB,MAAM,aAAa,GACjB,KAAK,CAAC,IAAI,KAAK,OAAO;gBACpB,CAAC,CAAC,IAAI,iBAAa,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,MAAM,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC;gBACvD,CAAC,CAAC,IAAI,iBAAa,CAAC,GAAG,EAAE,CAAC,IAAI,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC,CAAC;YACrD,IAAI,gBAAY,CAAC;gBACf,SAAS,EAAE,KAAK,CAAC,YAAY;gBAC7B,UAAU,EAAE,KAAK,CAAC,aAAa;aAChC,CAAC,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC;QACzB,CAAC,CAAC,CAAC;IACL,CAAC;IAED,KAAK,CAAC,IAAI;QACR,MAAM,IAAI,CAAC,IAAK,CAAC,IAAI,EAAE,CAAC;IAC1B,CAAC;CACF;AA9DD,kCA8DC;AAED,MAAM,oBAAoB,GAAG,CAAC,MAAM,EAAE,KAAK,EAAE,KAAK,EAAE,MAAM,CAAU,CAAC","sourcesContent":["import * as fs from \"fs/promises\";\n\nimport { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n  JitterBuffer,\n  MediaStreamTrack,\n  SampleBuilder,\n  WebmOutput,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nexport class WebmFactory extends MediaWriter {\n  webm?: WebmOutput;\n\n  start(tracks: MediaStreamTrack[]) {\n    this.webm = new WebmOutput(\n      fs,\n      this.path,\n      tracks.map((track, i) => {\n        const trackNumber = i + 1;\n        const payloadType = track.codec!.payloadType;\n\n        if (track.kind === \"video\") {\n          const codec = ((): SupportedCodec => {\n            switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n              case \"vp8\":\n                return \"VP8\";\n              case \"vp9\":\n                return \"VP9\";\n              case \"h264\":\n                return \"MPEG4/ISO/AVC\";\n              case \"av1x\":\n                return \"AV1\";\n              default:\n                throw new Error();\n            }\n          })();\n          return {\n            kind: \"video\",\n            clockRate: 90000,\n            payloadType,\n            trackNumber,\n            codec,\n            width: this.options.width,\n            height: this.options.height,\n          };\n        } else {\n          return {\n            kind: \"audio\",\n            clockRate: 48000,\n            payloadType,\n            trackNumber,\n            codec: \"OPUS\",\n          };\n        }\n      })\n    );\n\n    tracks.forEach((track) => {\n      const sampleBuilder =\n        track.kind === \"video\"\n          ? new SampleBuilder((h) => !!h.marker).pipe(this.webm!)\n          : new SampleBuilder(() => true).pipe(this.webm!);\n      new JitterBuffer({\n        rtpStream: track.onReceiveRtp,\n        rtcpStream: track.onReceiveRtcp,\n      }).pipe(sampleBuilder);\n    });\n  }\n\n  async stop() {\n    await this.webm!.stop();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}

package/lib/webrtc/src/peerConnection.d.ts

@@ -59,6 +59,7 @@ export declare class RTCPeerConnection extends EventTarget {
     private getTransceiverByMid;
     private getTransceiverByMLineIndex;
     createOffer(): Promise<import("./sdp").RTCSessionDescription>;
+    buildOfferSdp(): SessionDescription;
     createDataChannel(label: string, options?: Partial<{
         maxPacketLifeTime?: number;
         protocol: string;
@@ -76,6 +77,8 @@ export declare class RTCPeerConnection extends EventTarget {
         sdp: string;
     }): Promise<SessionDescription>;
     private setLocal;
+    private getTransportByMid;
+    private getTransportByMLineIndex;
     addIceCandidate(candidateMessage: RTCIceCandidate): Promise<void>;
     private connect;
     private getLocalRtpParams;
@@ -97,6 +100,7 @@ export declare class RTCPeerConnection extends EventTarget {
     ms?: MediaStream): RTCRtpSender;
     private ensureCerts;
     createAnswer(): Promise<import("./sdp").RTCSessionDescription>;
+    private buildAnswer;
     close(): Promise<void>;
     private assertNotClosed;
     private updateIceGatheringState;
@@ -105,8 +109,8 @@ export declare class RTCPeerConnection extends EventTarget {
     private setConnectionState;
     private dispose;
 }
-export declare function createMediaDescriptionForTransceiver(transceiver: RTCRtpTransceiver, cname: string, direction: Direction, mid: string): MediaDescription;
-export declare function createMediaDescriptionForSctp(sctp: RTCSctpTransport, mid: string): MediaDescription;
+export declare function createMediaDescriptionForTransceiver(transceiver: RTCRtpTransceiver, cname: string, direction: Direction): MediaDescription;
+export declare function createMediaDescriptionForSctp(sctp: RTCSctpTransport): MediaDescription;
 export declare function addTransportDescription(media: MediaDescription, dtlsTransport: RTCDtlsTransport): void;
 export declare function allocateMid(mids: Set<string>): string;
 export declare type BundlePolicy = "max-compat" | "max-bundle" | "disable";