werift 0.22.1 → 0.22.3

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

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

@@ -4,7 +4,8 @@ import type { Kind } from "../types/domain";
 import type { RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters } from "./parameters";
 import type { RTCRtpReceiver } from "./rtpReceiver";
 import type { RTCRtpSender } from "./rtpSender";
-import type { MediaStreamTrack } from "./track";
+import { type RTCStats } from "./stats";
+import type { MediaStream, MediaStreamTrack } from "./track";
 export declare class RTCRtpTransceiver {
     readonly kind: Kind;
     receiver: RTCRtpReceiver;
@@ -18,7 +19,7 @@ export declare class RTCRtpTransceiver {
     /**should not be reused because it has been used for sending before. */
     usedForSender: boolean;
     private _currentDirection?;
-    offerDirection: Direction;
+    offerDirection: MediaDirection;
     _codecs: RTCRtpCodecParameters[];
     set codecs(codecs: RTCRtpCodecParameters[]);
     get codecs(): RTCRtpCodecParameters[];
@@ -26,34 +27,36 @@ export declare class RTCRtpTransceiver {
     options: Partial<TransceiverOptions>;
     stopping: boolean;
     stopped: boolean;
-    constructor(kind: Kind, dtlsTransport: RTCDtlsTransport, receiver: RTCRtpReceiver, sender: RTCRtpSender, 
+    constructor(kind: Kind, dtlsTransport: RTCDtlsTransport | undefined, receiver: RTCRtpReceiver, sender: RTCRtpSender, 
     /**RFC 8829 4.2.4.  direction the transceiver was initialized with */
-    _direction: Direction);
+    _direction: MediaDirection);
     get dtlsTransport(): RTCDtlsTransport;
     /**RFC 8829 4.2.4. setDirectionに渡された最後の値を示します */
     get direction(): "inactive" | "sendonly" | "recvonly" | "sendrecv";
-    setDirection(direction: Direction): void;
+    setDirection(direction: MediaDirection): void;
     /**RFC 8829 4.2.5. last negotiated direction */
-    get currentDirection(): Direction | undefined;
-    setCurrentDirection(direction: Direction | undefined): void;
+    get currentDirection(): MediaDirection | undefined;
+    setCurrentDirection(direction: MediaDirection | undefined): void;
     setDtlsTransport(dtls: RTCDtlsTransport): void;
     get msid(): string;
     addTrack(track: MediaStreamTrack): void;
     stop(): void;
     getPayloadType(mimeType: string): number | undefined;
+    getCodecStats(): RTCStats[];
 }
 export declare const Inactive = "inactive";
 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 type Direction = (typeof Directions)[number];
+export type MediaDirection = (typeof Directions)[number];
 type SimulcastDirection = "send" | "recv";
 export interface TransceiverOptions {
-    direction: Direction;
+    direction: MediaDirection;
     simulcast: {
         direction: SimulcastDirection;
         rid: string;
     }[];
+    streams: MediaStream[];
 }
 export {};

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

@@ -1,32 +1,10 @@
 "use strict";
-var __createBinding = (this && this.__createBinding) || (Object.create ? (function(o, m, k, k2) {
-    if (k2 === undefined) k2 = k;
-    var desc = Object.getOwnPropertyDescriptor(m, k);
-    if (!desc || ("get" in desc ? !m.__esModule : desc.writable || desc.configurable)) {
-      desc = { enumerable: true, get: function() { return m[k]; } };
-    }
-    Object.defineProperty(o, k2, desc);
-}) : (function(o, m, k, k2) {
-    if (k2 === undefined) k2 = k;
-    o[k2] = m[k];
-}));
-var __setModuleDefault = (this && this.__setModuleDefault) || (Object.create ? (function(o, v) {
-    Object.defineProperty(o, "default", { enumerable: true, value: v });
-}) : function(o, v) {
-    o["default"] = v;
-});
-var __importStar = (this && this.__importStar) || function (mod) {
-    if (mod && mod.__esModule) return mod;
-    var result = {};
-    if (mod != null) for (var k in mod) if (k !== "default" && Object.prototype.hasOwnProperty.call(mod, k)) __createBinding(result, mod, k);
-    __setModuleDefault(result, mod);
-    return result;
-};
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.Directions = exports.Sendrecv = exports.Recvonly = exports.Sendonly = exports.Inactive = exports.RTCRtpTransceiver = void 0;
-const uuid = __importStar(require("uuid"));
+const crypto_1 = require("crypto");
 const common_1 = require("../imports/common");
 const const_1 = require("../const");
+const stats_1 = require("./stats");
 class RTCRtpTransceiver {
     set codecs(codecs) {
         this._codecs = codecs;
@@ -65,7 +43,7 @@ class RTCRtpTransceiver {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: uuid.v4()
+            value: (0, crypto_1.randomUUID)().toString()
         });
         Object.defineProperty(this, "onTrack", {
             enumerable: true,
@@ -134,7 +112,9 @@ class RTCRtpTransceiver {
             writable: true,
             value: false
         });
-        this.setDtlsTransport(dtlsTransport);
+        if (dtlsTransport) {
+            this.setDtlsTransport(dtlsTransport);
+        }
     }
     get dtlsTransport() {
         return this.receiver.dtlsTransport;
@@ -181,6 +161,30 @@ class RTCRtpTransceiver {
     getPayloadType(mimeType) {
         return this.codecs.find((codec) => codec.mimeType.toLowerCase().includes(mimeType.toLowerCase()))?.payloadType;
     }
+    getCodecStats() {
+        const timestamp = (0, stats_1.getStatsTimestamp)();
+        const stats = [];
+        if (!this.dtlsTransport) {
+            return stats;
+        }
+        const transportId = (0, stats_1.generateStatsId)("transport", this.dtlsTransport.id);
+        // Add codec stats for each codec
+        for (const codec of this.codecs) {
+            const codecStats = {
+                type: "codec",
+                id: (0, stats_1.generateStatsId)("codec", codec.payloadType, transportId),
+                timestamp,
+                payloadType: codec.payloadType,
+                transportId,
+                mimeType: codec.mimeType,
+                clockRate: codec.clockRate,
+                channels: codec.channels,
+                sdpFmtpLine: codec.parameters,
+            };
+            stats.push(codecStats);
+        }
+        return stats;
+    }
 }
 exports.RTCRtpTransceiver = RTCRtpTransceiver;
 exports.Inactive = "inactive";

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

@@ -1 +1 @@
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\"\")) {\n      this.usedForSender = true;\n    }\n  }\n\n  /**RFC 8829 4.2.5. last negotiated direction */\n  get currentDirection(): Direction | undefined {\n    return this._currentDirection;\n  }\n\n  setCurrentDirection(direction: Direction | undefined) {\n    this._currentDirection = direction;\n  }\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.receiver.setDtlsTransport(dtls);\n    this.sender.setDtlsTransport(dtls);\n  }\n\n  get msid() {\n    return `${this.sender.streamId} ${this.sender.trackId}`;\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const res = this.receiver.addTrack(track);\n    if (res) {\n      this.onTrack.execute(track, this);\n    }\n  }\n\n  // todo impl\n  // https://www.w3.org/TR/webrtc/#methods-8\n  stop() {\n    if (this.stopping) {\n      return;\n    }\n\n    // todo Stop sending and receiving with transceiver.\n\n    this.stopping = true;\n  }\n\n  getPayloadType(mimeType: string) {\n    return this.codecs.find((codec) =>\n      codec.mimeType.toLowerCase().includes(mimeType.toLowerCase()),\n    )?.payloadType;\n  }\n}\n\nexport const Inactive = \"inactive\";\nexport const Sendonly = \"sendonly\";\nexport const Recvonly = \"recvonly\";\nexport const Sendrecv = \"sendrecv\";\n\nexport const Directions = [Inactive, Sendonly, Recvonly, Sendrecv] as const;\n\nexport type Direction = (typeof Directions)[number];\n\ntype SimulcastDirection = \"send\" | \"recv\";\n\nexport interface TransceiverOptions {\n  direction: Direction;\n  simulcast: { direction: SimulcastDirection; rid: string }[];\n}\n"]}
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{ randomUUID } from \"crypto\";\nimport { Event } from \"../imports/common\";\n\nimport type { RTCDtlsTransport } from \"..\";\nimport { SenderDirections } from \"../const\";\nimport type { Kind } from \"../types/domain\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n} from \"./parameters\";\nimport type { RTCRtpReceiver } from \"./rtpReceiver\";\nimport type { RTCRtpSender } from \"./rtpSender\";\nimport {\n  type RTCCodecStats,\n  type RTCStats,\n  generateStatsId,\n  getStatsTimestamp,\n} from \"./stats\";\nimport type { MediaStream, MediaStreamTrack } from \"./track\";\n\nexport class RTCRtpTransceiver {\n  readonly id = randomUUID().toString();\n  readonly onTrack = new Event<[MediaStreamTrack, RTCRtpTransceiver]>();\n  mid?: string;\n  mLineIndex?: number;\n  /**should not be reused because it has been used for sending before. */\n  usedForSender = false;\n  private _currentDirection?: MediaDirection;\n  offerDirection!: MediaDirection;\n  _codecs: RTCRtpCodecParameters[] = [];\n  set codecs(codecs: RTCRtpCodecParameters[]) {\n    this._codecs = codecs;\n  }\n  get codecs() {\n    return this._codecs;\n  }\n  headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  options: Partial<TransceiverOptions> = {};\n  stopping = false;\n  stopped = false;\n\n  constructor(\n    public readonly kind: Kind,\n    dtlsTransport: RTCDtlsTransport | undefined,\n    public receiver: RTCRtpReceiver,\n    public sender: RTCRtpSender,\n    /**RFC 8829 4.2.4.  direction the transceiver was initialized with */\n    private _direction: MediaDirection,\n  ) {\n    if (dtlsTransport) {\n      this.setDtlsTransport(dtlsTransport);\n    }\n  }\n\n  get dtlsTransport() {\n    return this.receiver.dtlsTransport;\n  }\n\n  /**RFC 8829 4.2.4. setDirectionに渡された最後の値を示します */\n  get direction() {\n    return this._direction;\n  }\n\n  setDirection(direction: MediaDirection) {\n    this._direction = direction;\n    if (SenderDirections.includes(this._currentDirection ?? \"\")) {\n      this.usedForSender = true;\n    }\n  }\n\n  /**RFC 8829 4.2.5. last negotiated direction */\n  get currentDirection(): MediaDirection | undefined {\n    return this._currentDirection;\n  }\n\n  setCurrentDirection(direction: MediaDirection | undefined) {\n    this._currentDirection = direction;\n  }\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.receiver.setDtlsTransport(dtls);\n    this.sender.setDtlsTransport(dtls);\n  }\n\n  get msid() {\n    return `${this.sender.streamId} ${this.sender.trackId}`;\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const res = this.receiver.addTrack(track);\n    if (res) {\n      this.onTrack.execute(track, this);\n    }\n  }\n\n  // todo impl\n  // https://www.w3.org/TR/webrtc/#methods-8\n  stop() {\n    if (this.stopping) {\n      return;\n    }\n\n    // todo Stop sending and receiving with transceiver.\n\n    this.stopping = true;\n  }\n\n  getPayloadType(mimeType: string) {\n    return this.codecs.find((codec) =>\n      codec.mimeType.toLowerCase().includes(mimeType.toLowerCase()),\n    )?.payloadType;\n  }\n\n  getCodecStats(): RTCStats[] {\n    const timestamp = getStatsTimestamp();\n    const stats: RTCStats[] = [];\n\n    if (!this.dtlsTransport) {\n      return stats;\n    }\n\n    const transportId = generateStatsId(\"transport\", this.dtlsTransport.id);\n\n    // Add codec stats for each codec\n    for (const codec of this.codecs) {\n      const codecStats: RTCCodecStats = {\n        type: \"codec\",\n        id: generateStatsId(\"codec\", codec.payloadType, transportId),\n        timestamp,\n        payloadType: codec.payloadType,\n        transportId,\n        mimeType: codec.mimeType,\n        clockRate: codec.clockRate,\n        channels: codec.channels,\n        sdpFmtpLine: codec.parameters,\n      };\n      stats.push(codecStats);\n    }\n\n    return stats;\n  }\n}\n\nexport const Inactive = \"inactive\";\nexport const Sendonly = \"sendonly\";\nexport const Recvonly = \"recvonly\";\nexport const Sendrecv = \"sendrecv\";\n\nexport const Directions = [Inactive, Sendonly, Recvonly, Sendrecv] as const;\n\nexport type MediaDirection = (typeof Directions)[number];\n\ntype SimulcastDirection = \"send\" | \"recv\";\n\nexport interface TransceiverOptions {\n  direction: MediaDirection;\n  simulcast: { direction: SimulcastDirection; rid: string }[];\n  streams: MediaStream[];\n}\n"]}

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

@@ -0,0 +1,279 @@
+/**
+ * WebRTC Statistics API implementation
+ * Based on: https://www.w3.org/TR/webrtc-stats/
+ */
+export type RTCStatsType = "codec" | "inbound-rtp" | "outbound-rtp" | "remote-inbound-rtp" | "remote-outbound-rtp" | "media-source" | "peer-connection" | "data-channel" | "transport" | "candidate-pair" | "local-candidate" | "remote-candidate" | "certificate";
+export interface RTCStats {
+    timestamp: number;
+    type: RTCStatsType;
+    id: string;
+}
+export interface RTCRtpStreamStats extends RTCStats {
+    ssrc: number;
+    kind: string;
+    transportId?: string;
+    codecId?: string;
+}
+export interface RTCReceivedRtpStreamStats extends RTCRtpStreamStats {
+    packetsReceived?: number;
+    packetsLost?: number;
+    jitter?: number;
+    packetsDiscarded?: number;
+    packetsRepaired?: number;
+    burstPacketsLost?: number;
+    burstPacketsDiscarded?: number;
+    burstLossCount?: number;
+    burstDiscardCount?: number;
+    burstLossRate?: number;
+    burstDiscardRate?: number;
+    gapLossRate?: number;
+    gapDiscardRate?: number;
+    framesDropped?: number;
+    partialFramesLost?: number;
+    fullFramesLost?: number;
+}
+export interface RTCInboundRtpStreamStats extends RTCReceivedRtpStreamStats {
+    type: "inbound-rtp";
+    trackIdentifier?: string;
+    mid?: string;
+    remoteId?: string;
+    framesDecoded?: number;
+    keyFramesDecoded?: number;
+    frameWidth?: number;
+    frameHeight?: number;
+    frameBitDepth?: number;
+    framesPerSecond?: number;
+    qpSum?: number;
+    totalDecodeTime?: number;
+    totalInterFrameDelay?: number;
+    totalSquaredInterFrameDelay?: number;
+    lastPacketReceivedTimestamp?: number;
+    headerBytesReceived?: number;
+    packetsFailedDecryption?: number;
+    packetsDuplicated?: number;
+    perDscpPacketsReceived?: Record<string, number>;
+    nackCount?: number;
+    firCount?: number;
+    pliCount?: number;
+    totalProcessingDelay?: number;
+    estimatedPlayoutTimestamp?: number;
+    jitterBufferDelay?: number;
+    jitterBufferTargetDelay?: number;
+    jitterBufferEmittedCount?: number;
+    jitterBufferMinimumDelay?: number;
+    totalSamplesReceived?: number;
+    concealedSamples?: number;
+    silentConcealedSamples?: number;
+    concealmentEvents?: number;
+    insertedSamplesForDeceleration?: number;
+    removedSamplesForAcceleration?: number;
+    audioLevel?: number;
+    totalAudioEnergy?: number;
+    totalSamplesDuration?: number;
+    framesReceived?: number;
+    decoderImplementation?: string;
+    playoutId?: string;
+    powerEfficientDecoder?: boolean;
+    framesAssembledFromMultiplePackets?: number;
+    totalAssemblyTime?: number;
+    retransmittedPacketsReceived?: number;
+    retransmittedBytesReceived?: number;
+    rtxSsrc?: number;
+    fecSsrc?: number;
+}
+export interface RTCSentRtpStreamStats extends RTCRtpStreamStats {
+    packetsSent?: number;
+    bytesSent?: number;
+}
+export interface RTCOutboundRtpStreamStats extends RTCSentRtpStreamStats {
+    type: "outbound-rtp";
+    mid?: string;
+    mediaSourceId?: string;
+    remoteId?: string;
+    rid?: string;
+    headerBytesSent?: number;
+    retransmittedPacketsSent?: number;
+    retransmittedBytesSent?: number;
+    rtxSsrc?: number;
+    targetBitrate?: number;
+    totalEncodedBytesTarget?: number;
+    frameWidth?: number;
+    frameHeight?: number;
+    frameBitDepth?: number;
+    framesPerSecond?: number;
+    framesSent?: number;
+    hugeFramesSent?: number;
+    framesEncoded?: number;
+    keyFramesEncoded?: number;
+    qpSum?: number;
+    totalEncodeTime?: number;
+    totalPacketSendDelay?: number;
+    qualityLimitationReason?: string;
+    qualityLimitationDurations?: Record<string, number>;
+    qualityLimitationResolutionChanges?: number;
+    perDscpPacketsSent?: Record<string, number>;
+    nackCount?: number;
+    firCount?: number;
+    pliCount?: number;
+    encoderImplementation?: string;
+    powerEfficientEncoder?: boolean;
+    active?: boolean;
+    scalabilityMode?: string;
+}
+export interface RTCRemoteInboundRtpStreamStats extends RTCReceivedRtpStreamStats {
+    type: "remote-inbound-rtp";
+    localId?: string;
+    roundTripTime?: number;
+    totalRoundTripTime?: number;
+    fractionLost?: number;
+    roundTripTimeMeasurements?: number;
+}
+export interface RTCRemoteOutboundRtpStreamStats extends RTCSentRtpStreamStats {
+    type: "remote-outbound-rtp";
+    localId?: string;
+    remoteTimestamp?: number;
+    reportsSent?: number;
+    roundTripTime?: number;
+    totalRoundTripTime?: number;
+    roundTripTimeMeasurements?: number;
+}
+export interface RTCMediaSourceStats extends RTCStats {
+    type: "media-source";
+    trackIdentifier: string;
+    kind: string;
+}
+export interface RTCVideoSourceStats extends RTCMediaSourceStats {
+    width?: number;
+    height?: number;
+    frames?: number;
+    framesPerSecond?: number;
+}
+export interface RTCAudioSourceStats extends RTCMediaSourceStats {
+    audioLevel?: number;
+    totalAudioEnergy?: number;
+    totalSamplesDuration?: number;
+    echoReturnLoss?: number;
+    echoReturnLossEnhancement?: number;
+    droppedSamplesDuration?: number;
+    droppedSamplesEvents?: number;
+    totalCaptureDelay?: number;
+    totalSamplesCaptured?: number;
+}
+export interface RTCPeerConnectionStats extends RTCStats {
+    type: "peer-connection";
+    dataChannelsOpened?: number;
+    dataChannelsClosed?: number;
+}
+export interface RTCDataChannelStats extends RTCStats {
+    type: "data-channel";
+    label?: string;
+    protocol?: string;
+    dataChannelIdentifier?: number;
+    state: RTCDataChannelState;
+    messagesSent?: number;
+    bytesSent?: number;
+    messagesReceived?: number;
+    bytesReceived?: number;
+}
+export interface RTCTransportStats extends RTCStats {
+    type: "transport";
+    bytesSent?: number;
+    packetsSent?: number;
+    bytesReceived?: number;
+    packetsReceived?: number;
+    rtcpTransportStatsId?: string;
+    iceRole?: RTCIceRole;
+    iceLocalUsernameFragment?: string;
+    dtlsState: RTCDtlsTransportState;
+    iceState?: RTCIceTransportState;
+    selectedCandidatePairId?: string;
+    localCertificateId?: string;
+    remoteCertificateId?: string;
+    tlsVersion?: string;
+    dtlsCipher?: string;
+    dtlsRole?: RTCDtlsRole;
+    srtpCipher?: string;
+    selectedCandidatePairChanges?: number;
+    iceRestarts?: number;
+}
+export interface RTCIceCandidateStats extends RTCStats {
+    transportId?: string;
+    address?: string;
+    port?: number;
+    protocol?: string;
+    candidateType: RTCIceCandidateType;
+    priority?: number;
+    url?: string;
+    relayProtocol?: string;
+    foundation?: string;
+    relatedAddress?: string;
+    relatedPort?: number;
+    usernameFragment?: string;
+    tcpType?: RTCIceTcpCandidateType;
+}
+export interface RTCIceCandidatePairStats extends RTCStats {
+    type: "candidate-pair";
+    transportId?: string;
+    localCandidateId: string;
+    remoteCandidateId: string;
+    state: RTCStatsIceCandidatePairState;
+    nominated?: boolean;
+    packetsSent?: number;
+    packetsReceived?: number;
+    bytesSent?: number;
+    bytesReceived?: number;
+    lastPacketSentTimestamp?: number;
+    lastPacketReceivedTimestamp?: number;
+    totalRoundTripTime?: number;
+    currentRoundTripTime?: number;
+    availableOutgoingBitrate?: number;
+    availableIncomingBitrate?: number;
+    requestsReceived?: number;
+    requestsSent?: number;
+    responsesReceived?: number;
+    responsesSent?: number;
+    retransmissionsReceived?: number;
+    retransmissionsSent?: number;
+    consentRequestsSent?: number;
+    consentExpiredTimestamp?: number;
+    packetsDiscardedOnSend?: number;
+    bytesDiscardedOnSend?: number;
+}
+export interface RTCCertificateStats extends RTCStats {
+    type: "certificate";
+    fingerprint: string;
+    fingerprintAlgorithm: string;
+    base64Certificate: string;
+    issuerCertificateId?: string;
+}
+export interface RTCCodecStats extends RTCStats {
+    type: "codec";
+    payloadType: number;
+    transportId: string;
+    mimeType: string;
+    clockRate?: number;
+    channels?: number;
+    sdpFmtpLine?: string;
+}
+export type RTCDataChannelState = "connecting" | "open" | "closing" | "closed";
+export type RTCDtlsTransportState = "new" | "connecting" | "connected" | "closed" | "failed";
+export type RTCIceTransportState = "new" | "checking" | "connected" | "completed" | "disconnected" | "failed" | "closed";
+export type RTCIceRole = "controlling" | "controlled";
+export type RTCDtlsRole = "client" | "server";
+export type RTCIceCandidateType = "host" | "srflx" | "prflx" | "relay";
+export type RTCIceTcpCandidateType = "active" | "passive" | "so";
+export type RTCStatsIceCandidatePairState = "frozen" | "waiting" | "in-progress" | "failed" | "succeeded";
+/**
+ * RTCStatsReport is a Map-like object that holds WebRTC statistics
+ */
+export declare class RTCStatsReport extends Map<string, RTCStats> {
+    constructor(stats?: Array<RTCStats>);
+}
+/**
+ * Generate a unique ID for a statistics object
+ */
+export declare function generateStatsId(type: string, ...parts: (string | number | undefined)[]): string;
+/**
+ * Get current timestamp in milliseconds (DOMHighResTimeStamp)
+ */
+export declare function getStatsTimestamp(): number;

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

@@ -1,3 +1,37 @@
 "use strict";
-// todo impl
+/**
+ * WebRTC Statistics API implementation
+ * Based on: https://www.w3.org/TR/webrtc-stats/
+ */
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.RTCStatsReport = void 0;
+exports.generateStatsId = generateStatsId;
+exports.getStatsTimestamp = getStatsTimestamp;
+/**
+ * RTCStatsReport is a Map-like object that holds WebRTC statistics
+ */
+class RTCStatsReport extends Map {
+    constructor(stats) {
+        super();
+        if (stats) {
+            for (const stat of stats) {
+                this.set(stat.id, stat);
+            }
+        }
+    }
+}
+exports.RTCStatsReport = RTCStatsReport;
+/**
+ * Generate a unique ID for a statistics object
+ */
+function generateStatsId(type, ...parts) {
+    const validParts = parts.filter((p) => p !== undefined);
+    return `${type}_${validParts.join("_")}`;
+}
+/**
+ * Get current timestamp in milliseconds (DOMHighResTimeStamp)
+ */
+function getStatsTimestamp() {
+    return performance.now();
+}
 //# sourceMappingURL=stats.js.map