werift 0.17.6 → 0.18.0

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

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

@@ -66,36 +66,270 @@ const RTP_HISTORY_SIZE = 128;
 const RTT_ALPHA = 0.85;
 class RTCRtpSender {
     constructor(trackOrKind) {
-        this.trackOrKind = trackOrKind;
-        this.type = "sender";
-        this.kind = typeof this.trackOrKind === "string"
-            ? this.trackOrKind
-            : this.trackOrKind.kind;
-        this.ssrc = jspack_1.jspack.Unpack("!L", (0, crypto_1.randomBytes)(4))[0];
-        this.rtxSsrc = jspack_1.jspack.Unpack("!L", (0, crypto_1.randomBytes)(4))[0];
-        this.streamId = uuid.v4();
-        this.trackId = uuid.v4();
-        this.onReady = new rx_mini_1.default();
-        this.onRtcp = new rx_mini_1.default();
-        this.onPictureLossIndication = new rx_mini_1.default();
-        this.onGenericNack = new rx_mini_1.default();
-        this.senderBWE = new senderBWE_1.SenderBandwidthEstimator();
-        this.rtxSequenceNumber = (0, src_1.random16)();
-        this._redDistance = 2;
-        this.redEncoder = new src_2.RedEncoder(this._redDistance);
-        this.headerExtensions = [];
-        this.ntpTimestamp = 0n;
-        this.rtpTimestamp = 0;
-        this.octetCount = 0;
-        this.packetCount = 0;
-        this.receiverEstimatedMaxBitrate = 0n;
-        this.timestampOffset = 0;
-        this.seqOffset = 0;
-        this.rtpCache = [];
-        this.dtlsDisposer = [];
-        this.stopped = false;
-        this.rtcpRunning = false;
-        this.rtcpCancel = new AbortController();
+        Object.defineProperty(this, "trackOrKind", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: trackOrKind
+        });
+        Object.defineProperty(this, "type", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: "sender"
+        });
+        Object.defineProperty(this, "kind", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "ssrc", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: jspack_1.jspack.Unpack("!L", (0, crypto_1.randomBytes)(4))[0]
+        });
+        Object.defineProperty(this, "rtxSsrc", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: jspack_1.jspack.Unpack("!L", (0, crypto_1.randomBytes)(4))[0]
+        });
+        Object.defineProperty(this, "streamId", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: uuid.v4()
+        });
+        Object.defineProperty(this, "trackId", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: uuid.v4()
+        });
+        Object.defineProperty(this, "onReady", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new rx_mini_1.default()
+        });
+        Object.defineProperty(this, "onRtcp", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new rx_mini_1.default()
+        });
+        Object.defineProperty(this, "onPictureLossIndication", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new rx_mini_1.default()
+        });
+        Object.defineProperty(this, "onGenericNack", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new rx_mini_1.default()
+        });
+        Object.defineProperty(this, "senderBWE", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new senderBWE_1.SenderBandwidthEstimator()
+        });
+        Object.defineProperty(this, "cname", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "mid", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "rtpStreamId", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "repairedRtpStreamId", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "rtxPayloadType", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "rtxSequenceNumber", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: (0, src_1.random16)()
+        });
+        Object.defineProperty(this, "redRedundantPayloadType", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "_redDistance", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 2
+        });
+        Object.defineProperty(this, "redEncoder", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new src_2.RedEncoder(this._redDistance)
+        });
+        Object.defineProperty(this, "headerExtensions", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: []
+        });
+        Object.defineProperty(this, "disposeTrack", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        // # stats
+        Object.defineProperty(this, "lastSRtimestamp", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "lastSentSRTimestamp", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "ntpTimestamp", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 0n
+        });
+        Object.defineProperty(this, "rtpTimestamp", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 0
+        });
+        Object.defineProperty(this, "octetCount", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 0
+        });
+        Object.defineProperty(this, "packetCount", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 0
+        });
+        Object.defineProperty(this, "rtt", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "receiverEstimatedMaxBitrate", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 0n
+        });
+        // rtp
+        Object.defineProperty(this, "sequenceNumber", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "timestamp", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "timestampOffset", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 0
+        });
+        Object.defineProperty(this, "seqOffset", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 0
+        });
+        Object.defineProperty(this, "rtpCache", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: []
+        });
+        Object.defineProperty(this, "codec", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "dtlsTransport", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "dtlsDisposer", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: []
+        });
+        Object.defineProperty(this, "track", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "stopped", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: false
+        });
+        Object.defineProperty(this, "rtcpRunning", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: false
+        });
+        Object.defineProperty(this, "rtcpCancel", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new AbortController()
+        });
+        this.kind =
+            typeof this.trackOrKind === "string"
+                ? this.trackOrKind
+                : this.trackOrKind.kind;
         if (trackOrKind instanceof track_1.MediaStreamTrack) {
             if (trackOrKind.streamId) {
                 this.streamId = trackOrKind.streamId;
@@ -252,13 +486,14 @@ class RTCRtpSender {
             return;
         }
         rtp = Buffer.isBuffer(rtp) ? src_2.RtpPacket.deSerialize(rtp) : rtp;
-        const header = rtp.header;
+        const { header, payload } = rtp;
         header.ssrc = this.ssrc;
         header.payloadType = this.codec.payloadType;
         header.timestamp = (0, src_1.uint32Add)(header.timestamp, this.timestampOffset);
         header.sequenceNumber = (0, src_1.uint16Add)(header.sequenceNumber, this.seqOffset);
         this.timestamp = header.timestamp;
         this.sequenceNumber = header.sequenceNumber;
+        const ntptime = (0, utils_1.ntpTime)();
         header.extensions = this.headerExtensions
             .map((extension) => {
             const payload = (() => {
@@ -285,7 +520,7 @@ class RTCRtpSender {
                         return (0, src_1.bufferWriter)([2], [this.dtlsTransport.transportSequenceNumber]);
                     case rtpExtension_1.RTP_EXTENSION_URI.absSendTime:
                         const buf = Buffer.alloc(3);
-                        const time = ((0, utils_1.ntpTime)() >> 14n) & 0x00ffffffn;
+                        const time = (ntptime >> 14n) & 0x00ffffffn;
                         buf.writeUIntBE(Number(time), 0, 3);
                         return buf;
                 }
@@ -294,14 +529,12 @@ class RTCRtpSender {
                 return { id: extension.id, payload };
         })
             .filter((v) => v);
-        this.ntpTimestamp = (0, utils_1.ntpTime)();
-        this.rtpTimestamp = rtp.header.timestamp;
-        this.octetCount += rtp.payload.length;
+        this.ntpTimestamp = ntptime;
+        this.rtpTimestamp = header.timestamp;
+        this.octetCount += payload.length;
         this.packetCount = (0, src_1.uint32Add)(this.packetCount, 1);
-        rtp.header = header;
-        this.rtpCache.push(rtp);
-        this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);
-        let rtpPayload = rtp.payload;
+        this.rtpCache[header.sequenceNumber % RTP_HISTORY_SIZE] = rtp;
+        let rtpPayload = payload;
         if (this.redRedundantPayloadType) {
             this.redEncoder.push({
                 block: rtpPayload,
@@ -313,11 +546,12 @@ class RTCRtpSender {
         }
         const size = await this.dtlsTransport.sendRtp(rtpPayload, header);
         this.runRtcp();
+        const millitime = (0, utils_1.milliTime)();
         const sentInfo = {
             wideSeq: this.dtlsTransport.transportSequenceNumber,
             size,
-            sendingAtMs: (0, utils_1.milliTime)(),
-            sentAtMs: (0, utils_1.milliTime)(),
+            sendingAtMs: millitime,
+            sentAtMs: millitime,
         };
         this.senderBWE.rtpPacketSent(sentInfo);
     }
@@ -360,7 +594,10 @@ class RTCRtpSender {
                             {
                                 const feedback = packet.feedback;
                                 feedback.lost.forEach(async (seqNum) => {
-                                    let packet = this.rtpCache.find((rtp) => rtp.header.sequenceNumber === seqNum);
+                                    let packet = this.rtpCache[seqNum % RTP_HISTORY_SIZE];
+                                    if (packet && packet.header.sequenceNumber !== seqNum) {
+                                        packet = undefined;
+                                    }
                                     if (packet) {
                                         if (this.rtxPayloadType != undefined) {
                                             packet = wrapRtx(packet, this.rtxPayloadType, this.rtxSequenceNumber, this.rtxSsrc);

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\";\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, timestampSeconds } 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 lastSRtimestamp?: number;\n  private lastSentSRTimestamp?: number;\n  private ntpTimestamp = 0n;\n  private rtpTimestamp = 0;\n  private octetCount = 0;\n  private packetCount = 0;\n  private rtt?: number;\n  receiverEstimatedMaxBitrate: 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 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}\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 = 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.lastSRtimestamp === report.lsr && report.dlsr) {\n                if (this.lastSentSRTimestamp) {\n                  const rtt =\n                    timestampSeconds() -\n                    this.lastSentSRTimestamp -\n                    report.dlsr / 65536;\n                  if (this.rtt === undefined) {\n                    this.rtt = rtt;\n                  } else {\n                    this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;\n                  }\n                }\n              }\n            });\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpTransportLayerFeedback;\n          switch (packet.feedback.count) {\n            case TransportWideCC.count:\n              {\n                const feedback = packet.feedback as TransportWideCC;\n                this.senderBWE.receiveTWCC(feedback);\n              }\n              break;\n            case GenericNack.count:\n              {\n                const feedback = packet.feedback as GenericNack;\n                feedback.lost.forEach(async (seqNum) => {\n                  let packet = 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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  [10 Nov 1995 11:33:25.125 UTC]       [10 Nov 1995 11:33:36.5 UTC]\n   n                 SR(n)              A=b710:8000 (46864.500 s)\n   ---------------------------------------------------------------->\n                      v                 ^\n   ntp_sec =0xb44db705 v               ^ dlsr=0x0005:4000 (    5.250s)\n   ntp_frac=0x20000000  v             ^  lsr =0xb705:2000 (46853.125s)\n     (3024992005.125 s)  v           ^\n   r                      v         ^ RR(n)\n   ---------------------------------------------------------------->\n                          |<-DLSR->|\n                           (5.250 s)\n        \n   A     0xb710:8000 (46864.500 s)\n   DLSR -0x0005:4000 (    5.250 s)\n   LSR  -0xb705:2000 (46853.125 s)\n   -------------------------------\n   delay 0x0006:2000 (    6.125 s)\n        \nFigure 2: Example for round-trip time computation\n */\n\nimport { randomBytes } from \"crypto\";\nimport 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, timestampSeconds } 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: Kind;\n  readonly ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  readonly rtxSsrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  streamId = uuid.v4();\n  readonly trackId = uuid.v4();\n  readonly onReady = new Event();\n  readonly onRtcp = new Event<[RtcpPacket]>();\n  readonly onPictureLossIndication = new Event<[]>();\n  readonly onGenericNack = new Event<[GenericNack]>();\n  readonly senderBWE = new SenderBandwidthEstimator();\n\n  private cname?: string;\n  private mid?: string;\n  private rtpStreamId?: string;\n  private repairedRtpStreamId?: string;\n  private rtxPayloadType?: number;\n  private rtxSequenceNumber = random16();\n  redRedundantPayloadType?: number;\n  private _redDistance = 2;\n  redEncoder = new RedEncoder(this._redDistance);\n  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lastSRtimestamp?: number;\n  private lastSentSRTimestamp?: number;\n  private ntpTimestamp = 0n;\n  private rtpTimestamp = 0;\n  private octetCount = 0;\n  private packetCount = 0;\n  private rtt?: number;\n  receiverEstimatedMaxBitrate: 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    this.kind =\n      typeof this.trackOrKind === \"string\"\n        ? this.trackOrKind\n        : this.trackOrKind.kind;\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    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n    this.track = undefined;\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const packets: RtcpPacket[] = [\n          new RtcpSrPacket({\n            ssrc: this.ssrc,\n            senderInfo: new RtcpSenderInfo({\n              ntpTimestamp: this.ntpTimestamp,\n              rtpTimestamp: this.rtpTimestamp,\n              packetCount: this.packetCount,\n              octetCount: this.octetCount,\n            }),\n          }),\n        ];\n        this.lastSRtimestamp = compactNtp(this.ntpTimestamp);\n        this.lastSentSRTimestamp = timestampSeconds();\n\n        if (this.cname) {\n          packets.push(\n            new RtcpSourceDescriptionPacket({\n              chunks: [\n                new SourceDescriptionChunk({\n                  source: this.ssrc,\n                  items: [\n                    new SourceDescriptionItem({ type: 1, text: this.cname }),\n                  ],\n                }),\n              ],\n            })\n          );\n        }\n\n        try {\n          await this.dtlsTransport.sendRtcp(packets);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  replaceRTP(\n    {\n      sequenceNumber,\n      timestamp,\n    }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">,\n    discontinuity = false\n  ) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n      if (discontinuity) {\n        this.seqOffset = uint16Add(this.seqOffset, 2);\n      }\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = uint32Add(this.timestamp, -timestamp);\n      if (discontinuity) {\n        this.timestampOffset = uint16Add(this.timestampOffset, 1);\n      }\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  async sendRtp(rtp: Buffer | RtpPacket) {\n    if (this.dtlsTransport.state !== \"connected\" || !this.codec) {\n      return;\n    }\n\n    rtp = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp;\n\n    const { header, payload } = rtp;\n    header.ssrc = this.ssrc;\n    header.payloadType = this.codec.payloadType;\n    header.timestamp = uint32Add(header.timestamp, this.timestampOffset);\n    header.sequenceNumber = uint16Add(header.sequenceNumber, this.seqOffset);\n    this.timestamp = header.timestamp;\n    this.sequenceNumber = header.sequenceNumber;\n\n    const ntptime = ntpTime();\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 = header.timestamp;\n    this.octetCount += payload.length;\n    this.packetCount = uint32Add(this.packetCount, 1);\n\n    this.rtpCache[header.sequenceNumber % RTP_HISTORY_SIZE] = rtp;\n\n    let rtpPayload = payload;\n\n    if (this.redRedundantPayloadType) {\n      this.redEncoder.push({\n        block: rtpPayload,\n        timestamp: header.timestamp,\n        blockPT: this.redRedundantPayloadType,\n      });\n      const red = this.redEncoder.build();\n      rtpPayload = red.serialize();\n    }\n\n    const size = await this.dtlsTransport.sendRtp(rtpPayload, header);\n\n    this.runRtcp();\n    const millitime = milliTime();\n    const sentInfo: SentInfo = {\n      wideSeq: this.dtlsTransport.transportSequenceNumber,\n      size,\n      sendingAtMs: millitime,\n      sentAtMs: millitime,\n    };\n    this.senderBWE.rtpPacketSent(sentInfo);\n  }\n\n  handleRtcpPacket(rtcpPacket: RtcpPacket) {\n    switch (rtcpPacket.type) {\n      case RtcpSrPacket.type:\n      case RtcpRrPacket.type:\n        {\n          const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;\n          packet.reports\n            .filter((report) => report.ssrc === this.ssrc)\n            .forEach((report) => {\n              if (this.lastSRtimestamp === report.lsr && report.dlsr) {\n                if (this.lastSentSRTimestamp) {\n                  const rtt =\n                    timestampSeconds() -\n                    this.lastSentSRTimestamp -\n                    report.dlsr / 65536;\n                  if (this.rtt === undefined) {\n                    this.rtt = rtt;\n                  } else {\n                    this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;\n                  }\n                }\n              }\n            });\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpTransportLayerFeedback;\n          switch (packet.feedback.count) {\n            case TransportWideCC.count:\n              {\n                const feedback = packet.feedback as TransportWideCC;\n                this.senderBWE.receiveTWCC(feedback);\n              }\n              break;\n            case GenericNack.count:\n              {\n                const feedback = packet.feedback as GenericNack;\n                feedback.lost.forEach(async (seqNum) => {\n                  let packet: RtpPacket | undefined =\n                    this.rtpCache[seqNum % RTP_HISTORY_SIZE];\n                  if (packet && packet.header.sequenceNumber !== seqNum) {\n                    packet = undefined;\n                  }\n                  if (packet) {\n                    if (this.rtxPayloadType != undefined) {\n                      packet = wrapRtx(\n                        packet,\n                        this.rtxPayloadType,\n                        this.rtxSequenceNumber,\n                        this.rtxSsrc\n                      );\n                      this.rtxSequenceNumber = uint16Add(\n                        this.rtxSequenceNumber,\n                        1\n                      );\n                    }\n                    await this.dtlsTransport.sendRtp(\n                      packet.payload,\n                      packet.header\n                    );\n                  }\n                });\n                this.onGenericNack.execute(feedback);\n              }\n              break;\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpPayloadSpecificFeedback;\n          switch (packet.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              {\n                const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;\n                this.receiverEstimatedMaxBitrate = feedback.bitrate;\n              }\n              break;\n            case PictureLossIndication.count:\n              {\n                this.onPictureLossIndication.execute();\n              }\n              break;\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(rtcpPacket);\n  }\n}\n\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

@@ -47,8 +47,8 @@ export declare const Sendonly = "sendonly";
 export declare const Recvonly = "recvonly";
 export declare const Sendrecv = "sendrecv";
 export declare const Directions: readonly ["inactive", "sendonly", "recvonly", "sendrecv"];
-export declare type Direction = typeof Directions[number];
-declare type SimulcastDirection = "send" | "recv";
+export type Direction = typeof Directions[number];
+type SimulcastDirection = "send" | "recv";
 export interface TransceiverOptions {
     direction: Direction;
     simulcast: {