werift 0.17.5 → 0.17.6

package/lib/common/src/network.js

@@ -31,10 +31,13 @@ async function findPort(min, max, protocol = "udp4", interfaceAddresses) {
             port: i,
             address: (0, exports.interfaceAddress)(protocol, interfaceAddresses),
         }));
-        await new Promise((r) => {
-            socket.once("error", r);
-            socket.once("listening", r);
+        const err = await new Promise((r) => {
+            socket.once("error", (e) => r(e));
+            socket.once("listening", () => r());
         });
+        if (err) {
+            continue;
+        }
         port = socket.address()?.port;
         await new Promise((r) => socket.close(() => r()));
         if (min <= port && port <= max) {

package/lib/common/src/network.js.map

@@ -1 +1 @@
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+{"version":3,"file":"network.js","sourceRoot":"","sources":["../../../../common/src/network.ts"],"names":[],"mappings":";;;AAAA,iCAAiD;AAM1C,MAAM,gBAAgB,GAAG,CAC9B,IAAgB,EAChB,kBAAkD,EAClD,EAAE,CAAC,CAAC,kBAAkB,CAAC,CAAC,CAAC,kBAAkB,CAAC,IAAI,CAAC,CAAC,CAAC,CAAC,SAAS,CAAC,CAAC;AAHpD,QAAA,gBAAgB,oBAGoC;AAE1D,KAAK,UAAU,UAAU,CAC9B,WAAuB,MAAM,EAC7B,kBAAuC;IAEvC,MAAM,MAAM,GAAG,IAAA,oBAAY,EAAC,QAAQ,CAAC,CAAC;IAEtC,YAAY,CAAC,GAAG,EAAE,CAChB,MAAM,CAAC,IAAI,CAAC;QACV,IAAI,EAAE,CAAC;QACP,OAAO,EAAE,IAAA,wBAAgB,EAAC,QAAQ,EAAE,kBAAkB,CAAC;KACxD,CAAC,CACH,CAAC;IAEF,MAAM,IAAI,OAAO,CAAO,CAAC,CAAC,EAAE,EAAE;QAC5B,MAAM,CAAC,IAAI,CAAC,OAAO,EAAE,CAAC,CAAC,CAAC;QACxB,MAAM,CAAC,IAAI,CAAC,WAAW,EAAE,CAAC,CAAC,CAAC;IAC9B,CAAC,CAAC,CAAC;IAEH,MAAM,IAAI,GAAG,MAAM,CAAC,OAAO,EAAE,EAAE,IAAI,CAAC;IACpC,MAAM,IAAI,OAAO,CAAO,CAAC,CAAC,EAAE,EAAE,CAAC,MAAM,CAAC,KAAK,CAAC,GAAG,EAAE,CAAC,CAAC,EAAE,CAAC,CAAC,CAAC;IACxD,OAAO,IAAI,CAAC;AACd,CAAC;AArBD,gCAqBC;AAEM,KAAK,UAAU,WAAW,CAC/B,GAAW,EACX,WAAuB,MAAM,EAC7B,kBAAuC;IAEvC,OAAO,OAAO,CAAC,GAAG,CAChB,CAAC,GAAG,KAAK,CAAC,GAAG,CAAC,CAAC,CAAC,GAAG,CAAC,GAAG,EAAE,CAAC,UAAU,CAAC,QAAQ,EAAE,kBAAkB,CAAC,CAAC,CACpE,CAAC;AACJ,CAAC;AARD,kCAQC;AAEM,KAAK,UAAU,QAAQ,CAC5B,GAAW,EACX,GAAW,EACX,WAAuB,MAAM,EAC7B,kBAAuC;IAEvC,IAAI,IAAwB,CAAC;IAE7B,KAAK,IAAI,CAAC,GAAG,GAAG,EAAE,CAAC,IAAI,GAAG,EAAE,CAAC,EAAE,EAAE;QAC/B,MAAM,MAAM,GAAG,IAAA,oBAAY,EAAC,QAAQ,CAAC,CAAC;QAEtC,YAAY,CAAC,GAAG,EAAE,CAChB,MAAM,CAAC,IAAI,CAAC;YACV,IAAI,EAAE,CAAC;YACP,OAAO,EAAE,IAAA,wBAAgB,EAAC,QAAQ,EAAE,kBAAkB,CAAC;SACxD,CAAC,CACH,CAAC;QAEF,MAAM,GAAG,GAAG,MAAM,IAAI,OAAO,CAAe,CAAC,CAAC,EAAE,EAAE;YAChD,MAAM,CAAC,IAAI,CAAC,OAAO,EAAE,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;YAClC,MAAM,CAAC,IAAI,CAAC,WAAW,EAAE,GAAG,EAAE,CAAC,CAAC,EAAE,CAAC,CAAC;QACtC,CAAC,CAAC,CAAC;QACH,IAAI,GAAG,EAAE;YACP,SAAS;SACV;QAED,IAAI,GAAG,MAAM,CAAC,OAAO,EAAE,EAAE,IAAI,CAAC;QAC9B,MAAM,IAAI,OAAO,CAAO,CAAC,CAAC,EAAE,EAAE,CAAC,MAAM,CAAC,KAAK,CAAC,GAAG,EAAE,CAAC,CAAC,EAAE,CAAC,CAAC,CAAC;QACxD,IAAI,GAAG,IAAI,IAAI,IAAI,IAAI,IAAI,GAAG,EAAE;YAC9B,MAAM;SACP;KACF;IAED,IAAI,CAAC,IAAI;QAAE,MAAM,IAAI,KAAK,CAAC,gBAAgB,CAAC,CAAC;IAE7C,OAAO,IAAI,CAAC;AACd,CAAC;AApCD,4BAoCC","sourcesContent":["import { createSocket, SocketType } from \"dgram\";\n\nexport type InterfaceAddresses = {\n  [K in SocketType]?: string;\n};\n\nexport const interfaceAddress = (\n  type: SocketType,\n  interfaceAddresses: InterfaceAddresses | undefined\n) => (interfaceAddresses ? interfaceAddresses[type] : undefined);\n\nexport async function randomPort(\n  protocol: SocketType = \"udp4\",\n  interfaceAddresses?: InterfaceAddresses\n) {\n  const socket = createSocket(protocol);\n\n  setImmediate(() =>\n    socket.bind({\n      port: 0,\n      address: interfaceAddress(protocol, interfaceAddresses),\n    })\n  );\n\n  await new Promise<void>((r) => {\n    socket.once(\"error\", r);\n    socket.once(\"listening\", r);\n  });\n\n  const port = socket.address()?.port;\n  await new Promise<void>((r) => socket.close(() => r()));\n  return port;\n}\n\nexport async function randomPorts(\n  num: number,\n  protocol: SocketType = \"udp4\",\n  interfaceAddresses?: InterfaceAddresses\n) {\n  return Promise.all(\n    [...Array(num)].map(() => randomPort(protocol, interfaceAddresses))\n  );\n}\n\nexport async function findPort(\n  min: number,\n  max: number,\n  protocol: SocketType = \"udp4\",\n  interfaceAddresses?: InterfaceAddresses\n) {\n  let port: number | undefined;\n\n  for (let i = min; i <= max; i++) {\n    const socket = createSocket(protocol);\n\n    setImmediate(() =>\n      socket.bind({\n        port: i,\n        address: interfaceAddress(protocol, interfaceAddresses),\n      })\n    );\n\n    const err = await new Promise<Error | void>((r) => {\n      socket.once(\"error\", (e) => r(e));\n      socket.once(\"listening\", () => r());\n    });\n    if (err) {\n      continue;\n    }\n\n    port = socket.address()?.port;\n    await new Promise<void>((r) => socket.close(() => r()));\n    if (min <= port && port <= max) {\n      break;\n    }\n  }\n\n  if (!port) throw new Error(\"port not found\");\n\n  return port;\n}\n"]}

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

@@ -181,6 +181,9 @@ class RTCRtpSender {
         this.stopped = true;
         this.rtcpRunning = false;
         this.rtcpCancel.abort();
+        if (this.disposeTrack) {
+            this.disposeTrack();
+        }
         this.track = undefined;
     }
     async runRtcp() {

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 = 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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 =\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"]}

package/lib/webrtc/src/nonstandard/recorder/writer/webm.d.ts

@@ -1,7 +1,9 @@
+import { EventDisposer } from "rx.mini";
 import { MediaStreamTrack, RtpSourceStream } from "../../..";
 import { MediaWriter } from ".";
 export declare class WebmFactory extends MediaWriter {
     rtpSources: RtpSourceStream[];
+    unSubscribers: EventDisposer;
     start(tracks: MediaStreamTrack[]): Promise<void>;
     stop(): Promise<void>;
 }

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

@@ -2,6 +2,7 @@
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.WebmFactory = void 0;
 const promises_1 = require("fs/promises");
+const rx_mini_1 = require("rx.mini");
 const __1 = require("../../..");
 const _1 = require(".");
 const sourcePath = "packages/webrtc/src/nonstandard/recorder/writer/webm.ts";
@@ -9,6 +10,7 @@ class WebmFactory extends _1.MediaWriter {
     constructor() {
         super(...arguments);
         this.rtpSources = [];
+        this.unSubscribers = new rx_mini_1.EventDisposer();
     }
     async start(tracks) {
         await (0, promises_1.unlink)(this.path).catch((e) => e);
@@ -60,14 +62,18 @@ class WebmFactory extends _1.MediaWriter {
         });
         this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {
             const rtpSource = new __1.RtpSourceStream();
-            track.onReceiveRtp.subscribe((r) => rtpSource.push(r));
-            const jitterBuffer = (0, __1.jitterBufferTransformer)(clockRate, {
-                latency: this.options.jitterBufferLatency,
-                bufferSize: this.options.jitterBufferSize,
-            });
+            track.onReceiveRtp
+                .subscribe((rtp) => {
+                rtpSource.push(rtp.clone());
+            })
+                .disposer(this.unSubscribers);
+            // const jitterBuffer = jitterBufferTransformer(clockRate, {
+            //   latency: this.options.jitterBufferLatency,
+            //   bufferSize: this.options.jitterBufferSize,
+            // });
             if (track.kind === "video") {
                 rtpSource.readable
-                    .pipeThrough(jitterBuffer)
+                    // .pipeThrough(jitterBuffer)
                     .pipeThrough((0, __1.depacketizeTransformer)(codec, {
                     waitForKeyframe: this.options.waitForKeyframe,
                     isFinalPacketInSequence: (h) => h.marker,
@@ -76,7 +82,7 @@ class WebmFactory extends _1.MediaWriter {
             }
             else {
                 rtpSource.readable
-                    .pipeThrough(jitterBuffer)
+                    // .pipeThrough(jitterBuffer)
                     .pipeThrough((0, __1.depacketizeTransformer)(codec))
                     .pipeTo(webm.audioStream);
             }
@@ -101,6 +107,7 @@ class WebmFactory extends _1.MediaWriter {
     }
     async stop() {
         await Promise.all(this.rtpSources.map((r) => r.stop()));
+        this.unSubscribers.dispose();
     }
 }
 exports.WebmFactory = WebmFactory;

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

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{ appendFile, open, unlink } from \"fs/promises\";\nimport { ReadableStreamDefaultReadResult } from \"stream/web\";\n\nimport { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n  depacketizeTransformer,\n  jitterBufferTransformer,\n  MediaStreamTrack,\n  RtpSourceStream,\n  WebmStream,\n  WebmStreamOutput,\n  WeriftError,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceStream[] = [];\n\n  async start(tracks: MediaStreamTrack[]) {\n    await unlink(this.path).catch((e) => e);\n\n    const inputTracks = tracks.map((track, i) => {\n      const trackNumber = i + 1;\n      const payloadType = track.codec!.payloadType;\n\n      if (track.kind === \"video\") {\n        const codec = ((): SupportedCodec => {\n          switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n            case \"vp8\":\n              return \"VP8\";\n            case \"vp9\":\n              return \"VP9\";\n            case \"h264\":\n              return \"MPEG4/ISO/AVC\";\n            case \"av1x\":\n              return \"AV1\";\n            default:\n              throw new WeriftError({\n                message: \"unsupported codec\",\n                payload: { track, path: sourcePath },\n              });\n          }\n        })();\n        return {\n          kind: \"video\" as const,\n          codec,\n          clockRate: 90000,\n          trackNumber,\n          width: this.options.width,\n          height: this.options.height,\n          payloadType,\n          track,\n        };\n      } else {\n        return {\n          kind: \"audio\" as const,\n          codec: \"OPUS\" as const,\n          clockRate: 48000,\n          trackNumber,\n          payloadType,\n          track,\n        };\n      }\n    });\n\n    const webm = new WebmStream(inputTracks, {\n      duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceStream();\n      track.onReceiveRtp.subscribe((r) => rtpSource.push(r));\n\n      const jitterBuffer = jitterBufferTransformer(clockRate, {\n        latency: this.options.jitterBufferLatency,\n        bufferSize: this.options.jitterBufferSize,\n      });\n\n      if (track.kind === \"video\") {\n        rtpSource.readable\n          .pipeThrough(jitterBuffer)\n          .pipeThrough(\n            depacketizeTransformer(codec, {\n              waitForKeyframe: this.options.waitForKeyframe,\n              isFinalPacketInSequence: (h) => h.marker,\n            })\n          )\n          .pipeTo(webm.videoStream);\n      } else {\n        rtpSource.readable\n          .pipeThrough(jitterBuffer)\n          .pipeThrough(depacketizeTransformer(codec))\n          .pipeTo(webm.audioStream);\n      }\n\n      return rtpSource;\n    });\n\n    const reader = webm.webmStream.getReader();\n    const readChunk = async ({\n      value,\n      done,\n    }: ReadableStreamDefaultReadResult<WebmStreamOutput>) => {\n      if (done) return;\n\n      if (value.saveToFile) {\n        await appendFile(this.path, value.saveToFile);\n      } else if (value.eol) {\n        const { durationElement } = value.eol;\n        const handler = await open(this.path, \"r+\");\n        await handler.write(durationElement, 0, durationElement.length, 83);\n        await handler.close();\n      }\n      reader.read().then(readChunk);\n    };\n    reader.read().then(readChunk);\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}
+{"version":3,"file":"webm.js","sourceRoot":"","sources":["../../../../../../src/nonstandard/recorder/writer/webm.ts"],"names":[],"mappings":";;;AAAA,0CAAuD;AACvD,qCAAwC;AAIxC,gCAQkB;AAClB,wBAAgC;AAEhC,MAAM,UAAU,GAAG,yDAAyD,CAAC;AAE7E,MAAa,WAAY,SAAQ,cAAW;IAA5C;;QACE,eAAU,GAAsB,EAAE,CAAC;QAEnC,kBAAa,GAAG,IAAI,uBAAa,EAAE,CAAC;IAgHtC,CAAC;IA9GC,KAAK,CAAC,KAAK,CAAC,MAA0B;QACpC,MAAM,IAAA,iBAAM,EAAC,IAAI,CAAC,IAAI,CAAC,CAAC,KAAK,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC;QAExC,MAAM,WAAW,GAAG,MAAM,CAAC,GAAG,CAAC,CAAC,KAAK,EAAE,CAAC,EAAE,EAAE;YAC1C,MAAM,WAAW,GAAG,CAAC,GAAG,CAAC,CAAC;YAC1B,MAAM,WAAW,GAAG,KAAK,CAAC,KAAM,CAAC,WAAW,CAAC;YAE7C,IAAI,KAAK,CAAC,IAAI,KAAK,OAAO,EAAE;gBAC1B,MAAM,KAAK,GAAG,CAAC,GAAmB,EAAE;oBAClC,QAAQ,KAAK,CAAC,KAAK,EAAE,IAAI,CAAC,WAAW,EAAyB,EAAE;wBAC9D,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,MAAM;4BACT,OAAO,eAAe,CAAC;wBACzB,KAAK,MAAM;4BACT,OAAO,KAAK,CAAC;wBACf;4BACE,MAAM,IAAI,eAAW,CAAC;gCACpB,OAAO,EAAE,mBAAmB;gCAC5B,OAAO,EAAE,EAAE,KAAK,EAAE,IAAI,EAAE,UAAU,EAAE;6BACrC,CAAC,CAAC;qBACN;gBACH,CAAC,CAAC,EAAE,CAAC;gBACL,OAAO;oBACL,IAAI,EAAE,OAAgB;oBACtB,KAAK;oBACL,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,KAAK,EAAE,IAAI,CAAC,OAAO,CAAC,KAAK;oBACzB,MAAM,EAAE,IAAI,CAAC,OAAO,CAAC,MAAM;oBAC3B,WAAW;oBACX,KAAK;iBACN,CAAC;aACH;iBAAM;gBACL,OAAO;oBACL,IAAI,EAAE,OAAgB;oBACtB,KAAK,EAAE,MAAe;oBACtB,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK;iBACN,CAAC;aACH;QACH,CAAC,CAAC,CAAC;QAEH,MAAM,IAAI,GAAG,IAAI,cAAU,CAAC,WAAW,EAAE;YACvC,QAAQ,EAAE,IAAI,CAAC,OAAO,CAAC,eAAe,IAAI,IAAI,GAAG,EAAE,GAAG,EAAE,GAAG,EAAE;SAC9D,CAAC,CAAC;QAEH,IAAI,CAAC,UAAU,GAAG,WAAW,CAAC,GAAG,CAAC,CAAC,EAAE,KAAK,EAAE,SAAS,EAAE,KAAK,EAAE,EAAE,EAAE;YAChE,MAAM,SAAS,GAAG,IAAI,mBAAe,EAAE,CAAC;YAExC,KAAK,CAAC,YAAY;iBACf,SAAS,CAAC,CAAC,GAAG,EAAE,EAAE;gBACjB,SAAS,CAAC,IAAI,CAAC,GAAG,CAAC,KAAK,EAAE,CAAC,CAAC;YAC9B,CAAC,CAAC;iBACD,QAAQ,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC;YAEhC,4DAA4D;YAC5D,+CAA+C;YAC/C,+CAA+C;YAC/C,MAAM;YAEN,IAAI,KAAK,CAAC,IAAI,KAAK,OAAO,EAAE;gBAC1B,SAAS,CAAC,QAAQ;oBAChB,6BAA6B;qBAC5B,WAAW,CACV,IAAA,0BAAsB,EAAC,KAAK,EAAE;oBAC5B,eAAe,EAAE,IAAI,CAAC,OAAO,CAAC,eAAe;oBAC7C,uBAAuB,EAAE,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,MAAM;iBACzC,CAAC,CACH;qBACA,MAAM,CAAC,IAAI,CAAC,WAAW,CAAC,CAAC;aAC7B;iBAAM;gBACL,SAAS,CAAC,QAAQ;oBAChB,6BAA6B;qBAC5B,WAAW,CAAC,IAAA,0BAAsB,EAAC,KAAK,CAAC,CAAC;qBAC1C,MAAM,CAAC,IAAI,CAAC,WAAW,CAAC,CAAC;aAC7B;YAED,OAAO,SAAS,CAAC;QACnB,CAAC,CAAC,CAAC;QAEH,MAAM,MAAM,GAAG,IAAI,CAAC,UAAU,CAAC,SAAS,EAAE,CAAC;QAC3C,MAAM,SAAS,GAAG,KAAK,EAAE,EACvB,KAAK,EACL,IAAI,GAC8C,EAAE,EAAE;YACtD,IAAI,IAAI;gBAAE,OAAO;YAEjB,IAAI,KAAK,CAAC,UAAU,EAAE;gBACpB,MAAM,IAAA,qBAAU,EAAC,IAAI,CAAC,IAAI,EAAE,KAAK,CAAC,UAAU,CAAC,CAAC;aAC/C;iBAAM,IAAI,KAAK,CAAC,GAAG,EAAE;gBACpB,MAAM,EAAE,eAAe,EAAE,GAAG,KAAK,CAAC,GAAG,CAAC;gBACtC,MAAM,OAAO,GAAG,MAAM,IAAA,eAAI,EAAC,IAAI,CAAC,IAAI,EAAE,IAAI,CAAC,CAAC;gBAC5C,MAAM,OAAO,CAAC,KAAK,CAAC,eAAe,EAAE,CAAC,EAAE,eAAe,CAAC,MAAM,EAAE,EAAE,CAAC,CAAC;gBACpE,MAAM,OAAO,CAAC,KAAK,EAAE,CAAC;aACvB;YACD,MAAM,CAAC,IAAI,EAAE,CAAC,IAAI,CAAC,SAAS,CAAC,CAAC;QAChC,CAAC,CAAC;QACF,MAAM,CAAC,IAAI,EAAE,CAAC,IAAI,CAAC,SAAS,CAAC,CAAC;IAChC,CAAC;IAED,KAAK,CAAC,IAAI;QACR,MAAM,OAAO,CAAC,GAAG,CAAC,IAAI,CAAC,UAAU,CAAC,GAAG,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,IAAI,EAAE,CAAC,CAAC,CAAC;QAExD,IAAI,CAAC,aAAa,CAAC,OAAO,EAAE,CAAC;IAC/B,CAAC;CACF;AAnHD,kCAmHC;AAED,MAAM,oBAAoB,GAAG,CAAC,MAAM,EAAE,KAAK,EAAE,KAAK,EAAE,MAAM,CAAU,CAAC","sourcesContent":["import { appendFile, open, unlink } from \"fs/promises\";\nimport { EventDisposer } from \"rx.mini\";\nimport { ReadableStreamDefaultReadResult } from \"stream/web\";\n\nimport { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n  depacketizeTransformer,\n  jitterBufferTransformer,\n  MediaStreamTrack,\n  RtpSourceStream,\n  WebmStream,\n  WebmStreamOutput,\n  WeriftError,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceStream[] = [];\n\n  unSubscribers = new EventDisposer();\n\n  async start(tracks: MediaStreamTrack[]) {\n    await unlink(this.path).catch((e) => e);\n\n    const inputTracks = tracks.map((track, i) => {\n      const trackNumber = i + 1;\n      const payloadType = track.codec!.payloadType;\n\n      if (track.kind === \"video\") {\n        const codec = ((): SupportedCodec => {\n          switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n            case \"vp8\":\n              return \"VP8\";\n            case \"vp9\":\n              return \"VP9\";\n            case \"h264\":\n              return \"MPEG4/ISO/AVC\";\n            case \"av1x\":\n              return \"AV1\";\n            default:\n              throw new WeriftError({\n                message: \"unsupported codec\",\n                payload: { track, path: sourcePath },\n              });\n          }\n        })();\n        return {\n          kind: \"video\" as const,\n          codec,\n          clockRate: 90000,\n          trackNumber,\n          width: this.options.width,\n          height: this.options.height,\n          payloadType,\n          track,\n        };\n      } else {\n        return {\n          kind: \"audio\" as const,\n          codec: \"OPUS\" as const,\n          clockRate: 48000,\n          trackNumber,\n          payloadType,\n          track,\n        };\n      }\n    });\n\n    const webm = new WebmStream(inputTracks, {\n      duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceStream();\n\n      track.onReceiveRtp\n        .subscribe((rtp) => {\n          rtpSource.push(rtp.clone());\n        })\n        .disposer(this.unSubscribers);\n\n      // const jitterBuffer = jitterBufferTransformer(clockRate, {\n      //   latency: this.options.jitterBufferLatency,\n      //   bufferSize: this.options.jitterBufferSize,\n      // });\n\n      if (track.kind === \"video\") {\n        rtpSource.readable\n          // .pipeThrough(jitterBuffer)\n          .pipeThrough(\n            depacketizeTransformer(codec, {\n              waitForKeyframe: this.options.waitForKeyframe,\n              isFinalPacketInSequence: (h) => h.marker,\n            })\n          )\n          .pipeTo(webm.videoStream);\n      } else {\n        rtpSource.readable\n          // .pipeThrough(jitterBuffer)\n          .pipeThrough(depacketizeTransformer(codec))\n          .pipeTo(webm.audioStream);\n      }\n\n      return rtpSource;\n    });\n\n    const reader = webm.webmStream.getReader();\n    const readChunk = async ({\n      value,\n      done,\n    }: ReadableStreamDefaultReadResult<WebmStreamOutput>) => {\n      if (done) return;\n\n      if (value.saveToFile) {\n        await appendFile(this.path, value.saveToFile);\n      } else if (value.eol) {\n        const { durationElement } = value.eol;\n        const handler = await open(this.path, \"r+\");\n        await handler.write(durationElement, 0, durationElement.length, 83);\n        await handler.close();\n      }\n      reader.read().then(readChunk);\n    };\n    reader.read().then(readChunk);\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n\n    this.unSubscribers.dispose();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "werift",
-  "version": "0.17.5",
+  "version": "0.17.6",
   "description": "WebRTC Implementation for TypeScript (Node.js)",
   "keywords": [
     "WebRTC",

package/src/media/rtpSender.ts

@@ -230,7 +230,9 @@ export class RTCRtpSender {
     this.stopped = true;
     this.rtcpRunning = false;
     this.rtcpCancel.abort();
-
+    if (this.disposeTrack) {
+      this.disposeTrack();
+    }
     this.track = undefined;
   }
 

package/src/nonstandard/recorder/writer/webm.ts

@@ -1,4 +1,5 @@
 import { appendFile, open, unlink } from "fs/promises";
+import { EventDisposer } from "rx.mini";
 import { ReadableStreamDefaultReadResult } from "stream/web";
 
 import { SupportedCodec } from "../../../../../rtp/src/container/webm";
@@ -18,6 +19,8 @@ const sourcePath = "packages/webrtc/src/nonstandard/recorder/writer/webm.ts";
 export class WebmFactory extends MediaWriter {
   rtpSources: RtpSourceStream[] = [];
 
+  unSubscribers = new EventDisposer();
+
   async start(tracks: MediaStreamTrack[]) {
     await unlink(this.path).catch((e) => e);
 
@@ -71,16 +74,21 @@ export class WebmFactory extends MediaWriter {
 
     this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {
       const rtpSource = new RtpSourceStream();
-      track.onReceiveRtp.subscribe((r) => rtpSource.push(r));
 
-      const jitterBuffer = jitterBufferTransformer(clockRate, {
-        latency: this.options.jitterBufferLatency,
-        bufferSize: this.options.jitterBufferSize,
-      });
+      track.onReceiveRtp
+        .subscribe((rtp) => {
+          rtpSource.push(rtp.clone());
+        })
+        .disposer(this.unSubscribers);
+
+      // const jitterBuffer = jitterBufferTransformer(clockRate, {
+      //   latency: this.options.jitterBufferLatency,
+      //   bufferSize: this.options.jitterBufferSize,
+      // });
 
       if (track.kind === "video") {
         rtpSource.readable
-          .pipeThrough(jitterBuffer)
+          // .pipeThrough(jitterBuffer)
           .pipeThrough(
             depacketizeTransformer(codec, {
               waitForKeyframe: this.options.waitForKeyframe,
@@ -90,7 +98,7 @@ export class WebmFactory extends MediaWriter {
           .pipeTo(webm.videoStream);
       } else {
         rtpSource.readable
-          .pipeThrough(jitterBuffer)
+          // .pipeThrough(jitterBuffer)
           .pipeThrough(depacketizeTransformer(codec))
           .pipeTo(webm.audioStream);
       }
@@ -120,6 +128,8 @@ export class WebmFactory extends MediaWriter {
 
   async stop() {
     await Promise.all(this.rtpSources.map((r) => r.stop()));
+
+    this.unSubscribers.dispose();
   }
 }