werift 0.22.1 → 0.22.3

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

@@ -1 +1 @@
-{"version":3,"file":"stats.js","sourceRoot":"","sources":["../../../../src/media/stats.ts"],"names":[],"mappings":";AAAA,YAAY","sourcesContent":["// todo impl\n"]}
+{"version":3,"file":"stats.js","sourceRoot":"","sources":["../../../../src/media/stats.ts"],"names":[],"mappings":";AAAA;;;GAGG;;;AA4UH,0CAMC;AAKD,8CAEC;AA9BD;;GAEG;AACH,MAAa,cAAe,SAAQ,GAAqB;IACvD,YAAY,KAAuB;QACjC,KAAK,EAAE,CAAC;QACR,IAAI,KAAK,EAAE,CAAC;YACV,KAAK,MAAM,IAAI,IAAI,KAAK,EAAE,CAAC;gBACzB,IAAI,CAAC,GAAG,CAAC,IAAI,CAAC,EAAE,EAAE,IAAI,CAAC,CAAC;YAC1B,CAAC;QACH,CAAC;IACH,CAAC;CACF;AATD,wCASC;AAED;;GAEG;AACH,SAAgB,eAAe,CAC7B,IAAY,EACZ,GAAG,KAAsC;IAEzC,MAAM,UAAU,GAAG,KAAK,CAAC,MAAM,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,KAAK,SAAS,CAAC,CAAC;IACxD,OAAO,GAAG,IAAI,IAAI,UAAU,CAAC,IAAI,CAAC,GAAG,CAAC,EAAE,CAAC;AAC3C,CAAC;AAED;;GAEG;AACH,SAAgB,iBAAiB;IAC/B,OAAO,WAAW,CAAC,GAAG,EAAE,CAAC;AAC3B,CAAC","sourcesContent":["/**\n * WebRTC Statistics API implementation\n * Based on: https://www.w3.org/TR/webrtc-stats/\n */\n\nexport type RTCStatsType =\n  | \"codec\"\n  | \"inbound-rtp\"\n  | \"outbound-rtp\"\n  | \"remote-inbound-rtp\"\n  | \"remote-outbound-rtp\"\n  | \"media-source\"\n  | \"peer-connection\"\n  | \"data-channel\"\n  | \"transport\"\n  | \"candidate-pair\"\n  | \"local-candidate\"\n  | \"remote-candidate\"\n  | \"certificate\";\n\nexport interface RTCStats {\n  timestamp: number;\n  type: RTCStatsType;\n  id: string;\n}\n\nexport interface RTCRtpStreamStats extends RTCStats {\n  ssrc: number;\n  kind: string;\n  transportId?: string;\n  codecId?: string;\n}\n\nexport interface RTCReceivedRtpStreamStats extends RTCRtpStreamStats {\n  packetsReceived?: number;\n  packetsLost?: number;\n  jitter?: number;\n  packetsDiscarded?: number;\n  packetsRepaired?: number;\n  burstPacketsLost?: number;\n  burstPacketsDiscarded?: number;\n  burstLossCount?: number;\n  burstDiscardCount?: number;\n  burstLossRate?: number;\n  burstDiscardRate?: number;\n  gapLossRate?: number;\n  gapDiscardRate?: number;\n  framesDropped?: number;\n  partialFramesLost?: number;\n  fullFramesLost?: number;\n}\n\nexport interface RTCInboundRtpStreamStats extends RTCReceivedRtpStreamStats {\n  type: \"inbound-rtp\";\n  trackIdentifier?: string;\n  mid?: string;\n  remoteId?: string;\n  framesDecoded?: number;\n  keyFramesDecoded?: number;\n  frameWidth?: number;\n  frameHeight?: number;\n  frameBitDepth?: number;\n  framesPerSecond?: number;\n  qpSum?: number;\n  totalDecodeTime?: number;\n  totalInterFrameDelay?: number;\n  totalSquaredInterFrameDelay?: number;\n  lastPacketReceivedTimestamp?: number;\n  headerBytesReceived?: number;\n  packetsFailedDecryption?: number;\n  packetsDuplicated?: number;\n  perDscpPacketsReceived?: Record<string, number>;\n  nackCount?: number;\n  firCount?: number;\n  pliCount?: number;\n  totalProcessingDelay?: number;\n  estimatedPlayoutTimestamp?: number;\n  jitterBufferDelay?: number;\n  jitterBufferTargetDelay?: number;\n  jitterBufferEmittedCount?: number;\n  jitterBufferMinimumDelay?: number;\n  totalSamplesReceived?: number;\n  concealedSamples?: number;\n  silentConcealedSamples?: number;\n  concealmentEvents?: number;\n  insertedSamplesForDeceleration?: number;\n  removedSamplesForAcceleration?: number;\n  audioLevel?: number;\n  totalAudioEnergy?: number;\n  totalSamplesDuration?: number;\n  framesReceived?: number;\n  decoderImplementation?: string;\n  playoutId?: string;\n  powerEfficientDecoder?: boolean;\n  framesAssembledFromMultiplePackets?: number;\n  totalAssemblyTime?: number;\n  retransmittedPacketsReceived?: number;\n  retransmittedBytesReceived?: number;\n  rtxSsrc?: number;\n  fecSsrc?: number;\n}\n\nexport interface RTCSentRtpStreamStats extends RTCRtpStreamStats {\n  packetsSent?: number;\n  bytesSent?: number;\n}\n\nexport interface RTCOutboundRtpStreamStats extends RTCSentRtpStreamStats {\n  type: \"outbound-rtp\";\n  mid?: string;\n  mediaSourceId?: string;\n  remoteId?: string;\n  rid?: string;\n  headerBytesSent?: number;\n  retransmittedPacketsSent?: number;\n  retransmittedBytesSent?: number;\n  rtxSsrc?: number;\n  targetBitrate?: number;\n  totalEncodedBytesTarget?: number;\n  frameWidth?: number;\n  frameHeight?: number;\n  frameBitDepth?: number;\n  framesPerSecond?: number;\n  framesSent?: number;\n  hugeFramesSent?: number;\n  framesEncoded?: number;\n  keyFramesEncoded?: number;\n  qpSum?: number;\n  totalEncodeTime?: number;\n  totalPacketSendDelay?: number;\n  qualityLimitationReason?: string;\n  qualityLimitationDurations?: Record<string, number>;\n  qualityLimitationResolutionChanges?: number;\n  perDscpPacketsSent?: Record<string, number>;\n  nackCount?: number;\n  firCount?: number;\n  pliCount?: number;\n  encoderImplementation?: string;\n  powerEfficientEncoder?: boolean;\n  active?: boolean;\n  scalabilityMode?: string;\n}\n\nexport interface RTCRemoteInboundRtpStreamStats\n  extends RTCReceivedRtpStreamStats {\n  type: \"remote-inbound-rtp\";\n  localId?: string;\n  roundTripTime?: number;\n  totalRoundTripTime?: number;\n  fractionLost?: number;\n  roundTripTimeMeasurements?: number;\n}\n\nexport interface RTCRemoteOutboundRtpStreamStats extends RTCSentRtpStreamStats {\n  type: \"remote-outbound-rtp\";\n  localId?: string;\n  remoteTimestamp?: number;\n  reportsSent?: number;\n  roundTripTime?: number;\n  totalRoundTripTime?: number;\n  roundTripTimeMeasurements?: number;\n}\n\nexport interface RTCMediaSourceStats extends RTCStats {\n  type: \"media-source\";\n  trackIdentifier: string;\n  kind: string;\n}\n\nexport interface RTCVideoSourceStats extends RTCMediaSourceStats {\n  width?: number;\n  height?: number;\n  frames?: number;\n  framesPerSecond?: number;\n}\n\nexport interface RTCAudioSourceStats extends RTCMediaSourceStats {\n  audioLevel?: number;\n  totalAudioEnergy?: number;\n  totalSamplesDuration?: number;\n  echoReturnLoss?: number;\n  echoReturnLossEnhancement?: number;\n  droppedSamplesDuration?: number;\n  droppedSamplesEvents?: number;\n  totalCaptureDelay?: number;\n  totalSamplesCaptured?: number;\n}\n\nexport interface RTCPeerConnectionStats extends RTCStats {\n  type: \"peer-connection\";\n  dataChannelsOpened?: number;\n  dataChannelsClosed?: number;\n}\n\nexport interface RTCDataChannelStats extends RTCStats {\n  type: \"data-channel\";\n  label?: string;\n  protocol?: string;\n  dataChannelIdentifier?: number;\n  state: RTCDataChannelState;\n  messagesSent?: number;\n  bytesSent?: number;\n  messagesReceived?: number;\n  bytesReceived?: number;\n}\n\nexport interface RTCTransportStats extends RTCStats {\n  type: \"transport\";\n  bytesSent?: number;\n  packetsSent?: number;\n  bytesReceived?: number;\n  packetsReceived?: number;\n  rtcpTransportStatsId?: string;\n  iceRole?: RTCIceRole;\n  iceLocalUsernameFragment?: string;\n  dtlsState: RTCDtlsTransportState;\n  iceState?: RTCIceTransportState;\n  selectedCandidatePairId?: string;\n  localCertificateId?: string;\n  remoteCertificateId?: string;\n  tlsVersion?: string;\n  dtlsCipher?: string;\n  dtlsRole?: RTCDtlsRole;\n  srtpCipher?: string;\n  selectedCandidatePairChanges?: number;\n  iceRestarts?: number;\n}\n\nexport interface RTCIceCandidateStats extends RTCStats {\n  transportId?: string;\n  address?: string;\n  port?: number;\n  protocol?: string;\n  candidateType: RTCIceCandidateType;\n  priority?: number;\n  url?: string;\n  relayProtocol?: string;\n  foundation?: string;\n  relatedAddress?: string;\n  relatedPort?: number;\n  usernameFragment?: string;\n  tcpType?: RTCIceTcpCandidateType;\n}\n\nexport interface RTCIceCandidatePairStats extends RTCStats {\n  type: \"candidate-pair\";\n  transportId?: string;\n  localCandidateId: string;\n  remoteCandidateId: string;\n  state: RTCStatsIceCandidatePairState;\n  nominated?: boolean;\n  packetsSent?: number;\n  packetsReceived?: number;\n  bytesSent?: number;\n  bytesReceived?: number;\n  lastPacketSentTimestamp?: number;\n  lastPacketReceivedTimestamp?: number;\n  totalRoundTripTime?: number;\n  currentRoundTripTime?: number;\n  availableOutgoingBitrate?: number;\n  availableIncomingBitrate?: number;\n  requestsReceived?: number;\n  requestsSent?: number;\n  responsesReceived?: number;\n  responsesSent?: number;\n  retransmissionsReceived?: number;\n  retransmissionsSent?: number;\n  consentRequestsSent?: number;\n  consentExpiredTimestamp?: number;\n  packetsDiscardedOnSend?: number;\n  bytesDiscardedOnSend?: number;\n}\n\nexport interface RTCCertificateStats extends RTCStats {\n  type: \"certificate\";\n  fingerprint: string;\n  fingerprintAlgorithm: string;\n  base64Certificate: string;\n  issuerCertificateId?: string;\n}\n\nexport interface RTCCodecStats extends RTCStats {\n  type: \"codec\";\n  payloadType: number;\n  transportId: string;\n  mimeType: string;\n  clockRate?: number;\n  channels?: number;\n  sdpFmtpLine?: string;\n}\n\n// Type definitions\nexport type RTCDataChannelState = \"connecting\" | \"open\" | \"closing\" | \"closed\";\nexport type RTCDtlsTransportState =\n  | \"new\"\n  | \"connecting\"\n  | \"connected\"\n  | \"closed\"\n  | \"failed\";\nexport type RTCIceTransportState =\n  | \"new\"\n  | \"checking\"\n  | \"connected\"\n  | \"completed\"\n  | \"disconnected\"\n  | \"failed\"\n  | \"closed\";\nexport type RTCIceRole = \"controlling\" | \"controlled\";\nexport type RTCDtlsRole = \"client\" | \"server\";\nexport type RTCIceCandidateType = \"host\" | \"srflx\" | \"prflx\" | \"relay\";\nexport type RTCIceTcpCandidateType = \"active\" | \"passive\" | \"so\";\nexport type RTCStatsIceCandidatePairState =\n  | \"frozen\"\n  | \"waiting\"\n  | \"in-progress\"\n  | \"failed\"\n  | \"succeeded\";\n\n/**\n * RTCStatsReport is a Map-like object that holds WebRTC statistics\n */\nexport class RTCStatsReport extends Map<string, RTCStats> {\n  constructor(stats?: Array<RTCStats>) {\n    super();\n    if (stats) {\n      for (const stat of stats) {\n        this.set(stat.id, stat);\n      }\n    }\n  }\n}\n\n/**\n * Generate a unique ID for a statistics object\n */\nexport function generateStatsId(\n  type: string,\n  ...parts: (string | number | undefined)[]\n): string {\n  const validParts = parts.filter((p) => p !== undefined);\n  return `${type}_${validParts.join(\"_\")}`;\n}\n\n/**\n * Get current timestamp in milliseconds (DOMHighResTimeStamp)\n */\nexport function getStatsTimestamp(): number {\n  return performance.now();\n}\n"]}

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

@@ -1,7 +1,7 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.MediaStream = exports.MediaStreamTrack = void 0;
-const uuid_1 = require("uuid");
+const crypto_1 = require("crypto");
 const common_1 = require("../imports/common");
 const helper_1 = require("../helper");
 const rtp_1 = require("../imports/rtp");
@@ -12,7 +12,7 @@ class MediaStreamTrack extends helper_1.EventTarget {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: (0, uuid_1.v4)()
+            value: (0, crypto_1.randomUUID)().toString()
         });
         /**MediaStream ID*/
         Object.defineProperty(this, "streamId", {
@@ -164,7 +164,7 @@ class MediaStream {
         else {
             Object.assign(this, props);
         }
-        this.id ?? (this.id = (0, uuid_1.v4)());
+        this.id ?? (this.id = (0, crypto_1.randomUUID)().toString());
     }
     addTrack(track) {
         track.streamId = this.id;

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

@@ -1 +1 @@
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RtpPacket.deSerialize(rtp) : rtp;\n    packet.header.payloadType =\n      this.codec?.payloadType ?? packet.header.payloadType;\n    this.onReceiveRtp.execute(packet);\n  };\n}\n\nexport class MediaStream {\n  id!: string;\n  tracks: MediaStreamTrack[] = [];\n\n  constructor(props: Partial<MediaStream> | MediaStreamTrack[]) {\n    if (Array.isArray(props)) {\n      this.tracks = props;\n    } else {\n      Object.assign(this, props);\n    }\n    this.id ??= randomUUID().toString();\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    track.streamId = this.id;\n    this.tracks.push(track);\n  }\n\n  getTracks() {\n    return this.tracks;\n  }\n\n  getAudioTracks() {\n    return this.tracks.filter((track) => track.kind === \"audio\");\n  }\n\n  getVideoTracks() {\n    return this.tracks.filter((track) => track.kind === \"video\");\n  }\n}\n"]}

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

@@ -1 +1 @@
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{ unlink } from \"fs/promises\";\nimport { Event, EventDisposer } from \"../../../imports/common\";\n\nimport { MediaWriter } from \".\";\nimport { type MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n  type ContainerSupportedCodec,\n  DepacketizeCallback,\n  JitterBufferCallback,\n  LipsyncCallback,\n  NtpTimeCallback,\n  RtcpSourceCallback,\n  RtpSourceCallback,\n  RtpTimeCallback,\n  WebmCallback,\n  type WebmTrack,\n  saveToFileSystem,\n} from \"../../../imports/rtpExtra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceCallback[] = [];\n  private onEol = new Event();\n  private ended = false;\n  unSubscribers = new EventDisposer();\n\n  async start(tracks: MediaStreamTrack[]) {\n    if (this.props.path) {\n      await unlink(this.props.path).catch((e) => e);\n    }\n\n    const inputTracks: (WebmTrack & { track: MediaStreamTrack })[] = tracks.map(\n      (track, i) => {\n        const trackNumber = i + 1;\n        const payloadType = track.codec!.payloadType;\n\n        if (track.kind === \"video\") {\n          const codec = ((): ContainerSupportedCodec => {\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.props.width ?? 640,\n            height: this.props.height ?? 360,\n            roll: this.props.roll,\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\n    const webm = new WebmCallback(inputTracks, {\n      duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n    const lipsync = new LipsyncCallback(this.props.lipsync ?? {});\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceCallback();\n      const rtcpSource = new RtcpSourceCallback();\n      track.onReceiveRtp\n        .subscribe((rtp) => {\n          rtpSource.input(rtp.clone());\n        })\n        .disposer(this.unSubscribers);\n      track.onReceiveRtcp\n        .subscribe((rtcp) => {\n          rtcpSource.input(rtcp);\n        })\n        .disposer(this.unSubscribers);\n      const time = this.props.disableNtp\n        ? new RtpTimeCallback(clockRate)\n        : new NtpTimeCallback(clockRate);\n\n      if (track.kind === \"video\") {\n        const depacketizer = new DepacketizeCallback(codec, {\n          isFinalPacketInSequence: (h) => h.marker,\n        });\n        const jitterBuffer = new JitterBufferCallback(\n          clockRate,\n          this.props.jitterBuffer ?? {},\n        );\n\n        rtpSource.pipe(jitterBuffer.input);\n        rtcpSource.pipe(time.input);\n\n        jitterBuffer.pipe(time.input);\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputVideo);\n        } else {\n          depacketizer.pipe(lipsync.inputVideo);\n          lipsync.pipeVideo(webm.inputVideo);\n        }\n      } else {\n        const depacketizer = new DepacketizeCallback(codec);\n\n        rtpSource.pipe(time.input);\n        rtcpSource.pipe(time.input);\n\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputAudio);\n        } else {\n          depacketizer.pipe(lipsync.inputAudio);\n          lipsync.pipeAudio(webm.inputAudio);\n        }\n      }\n\n      return rtpSource;\n    });\n    if (this.props.path) {\n      webm.pipe(async (o) => {\n        const eol = await saveToFileSystem(this.props.path)(o);\n        if (eol) {\n          this.onEol.execute();\n          this.ended = true;\n        }\n      });\n    } else if (this.props.stream) {\n      webm.pipe(async (o) => {\n        this.props.stream.execute(o);\n      });\n    }\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n\n    if (!this.ended) {\n      await this.onEol.asPromise(5000).catch((e) => e);\n    }\n    this.unSubscribers.dispose();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}
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{ unlink } from \"fs/promises\";\nimport { Event, EventDisposer } from \"../../../imports/common\";\n\nimport { MediaWriter } from \".\";\nimport { type MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n  DepacketizeCallback,\n  JitterBufferCallback,\n  LipsyncCallback,\n  NtpTimeCallback,\n  RtcpSourceCallback,\n  RtpSourceCallback,\n  RtpTimeCallback,\n  WebmCallback,\n  type WebmSupportedCodec,\n  type WebmTrack,\n  saveToFileSystem,\n} from \"../../../imports/rtpExtra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceCallback[] = [];\n  private onEol = new Event();\n  private ended = false;\n  unSubscribers = new EventDisposer();\n\n  async start(tracks: MediaStreamTrack[]) {\n    if (this.props.path) {\n      await unlink(this.props.path).catch((e) => e);\n    }\n\n    const inputTracks: (WebmTrack & { track: MediaStreamTrack })[] = tracks.map(\n      (track, i) => {\n        const trackNumber = i + 1;\n        const payloadType = track.codec!.payloadType;\n\n        if (track.kind === \"video\") {\n          const codec = ((): WebmSupportedCodec => {\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.props.width ?? 640,\n            height: this.props.height ?? 360,\n            roll: this.props.roll,\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\n    const webm = new WebmCallback(inputTracks, {\n      duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n    const lipsync = new LipsyncCallback(this.props.lipsync ?? {});\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceCallback();\n      const rtcpSource = new RtcpSourceCallback();\n      track.onReceiveRtp\n        .subscribe((rtp) => {\n          rtpSource.input(rtp.clone());\n        })\n        .disposer(this.unSubscribers);\n      track.onReceiveRtcp\n        .subscribe((rtcp) => {\n          rtcpSource.input(rtcp);\n        })\n        .disposer(this.unSubscribers);\n      const time = this.props.disableNtp\n        ? new RtpTimeCallback(clockRate)\n        : new NtpTimeCallback(clockRate);\n\n      if (track.kind === \"video\") {\n        const depacketizer = new DepacketizeCallback(codec, {\n          isFinalPacketInSequence: (h) => h.marker,\n        });\n        const jitterBuffer = new JitterBufferCallback(\n          clockRate,\n          this.props.jitterBuffer ?? {},\n        );\n\n        rtpSource.pipe(jitterBuffer.input);\n        rtcpSource.pipe(time.input);\n\n        jitterBuffer.pipe(time.input);\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputVideo);\n        } else {\n          depacketizer.pipe(lipsync.inputVideo);\n          lipsync.pipeVideo(webm.inputVideo);\n        }\n      } else {\n        const depacketizer = new DepacketizeCallback(codec);\n\n        rtpSource.pipe(time.input);\n        rtcpSource.pipe(time.input);\n\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputAudio);\n        } else {\n          depacketizer.pipe(lipsync.inputAudio);\n          lipsync.pipeAudio(webm.inputAudio);\n        }\n      }\n\n      return rtpSource;\n    });\n    if (this.props.path) {\n      webm.pipe(async (o) => {\n        const eol = await saveToFileSystem(this.props.path)(o);\n        if (eol) {\n          this.onEol.execute();\n          this.ended = true;\n        }\n      });\n    } else if (this.props.stream) {\n      webm.pipe(async (o) => {\n        this.props.stream.execute(o);\n      });\n    }\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n\n    if (!this.ended) {\n      await this.onEol.asPromise(5000).catch((e) => e);\n    }\n    this.unSubscribers.dispose();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}

package/lib/webrtc/src/nonstandard/userMedia.js

@@ -2,9 +2,9 @@
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.MediaPlayerWebm = exports.MediaPlayerMp4 = exports.getUserMedia = void 0;
 const child_process_1 = require("child_process");
+const crypto_1 = require("crypto");
 const dgram_1 = require("dgram");
 const promises_1 = require("timers/promises");
-const uuid_1 = require("uuid");
 const common_1 = require("../imports/common");
 const rtp_1 = require("../imports/rtp");
 const track_1 = require("../media/track");
@@ -45,7 +45,7 @@ class MediaPlayer {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: (0, uuid_1.v4)()
+            value: (0, crypto_1.randomUUID)().toString()
         });
         Object.defineProperty(this, "audio", {
             enumerable: true,

package/lib/webrtc/src/nonstandard/userMedia.js.map

@@ -1 +1 @@
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{ type ChildProcess, exec } from \"child_process\";\nimport { randomUUID } from \"crypto\";\nimport { createSocket } from \"dgram\";\nimport { setImmediate } from \"timers/promises\";\n\nimport { randomPort } from \"../imports/common\";\nimport { RtpPacket } from \"../imports/rtp\";\nimport { MediaStreamTrack } from \"../media/track\";\n\nexport const getUserMedia = async ({\n  path,\n  loop,\n  width,\n  height,\n}: {\n  path: string;\n  loop?: boolean;\n  width?: number;\n  height?: number;\n}) => {\n  const audioPort = await randomPort();\n  const videoPort = await randomPort();\n\n  if (path.endsWith(\".mp4\")) {\n    return new MediaPlayerMp4({\n      audioPort,\n      videoPort,\n      path,\n      loop,\n      width,\n      height,\n    });\n  } else {\n    return new MediaPlayerWebm({\n      audioPort,\n      videoPort,\n      path,\n      loop,\n      width,\n      height,\n    });\n  }\n};\n\nabstract class MediaPlayer {\n  protected streamId = randomUUID().toString();\n  audio = new MediaStreamTrack({ kind: \"audio\", streamId: this.streamId });\n  video = new MediaStreamTrack({ kind: \"video\", streamId: this.streamId });\n  protected process!: ChildProcess;\n  stopped = false;\n\n  constructor(\n    protected props: {\n      videoPort: number;\n      audioPort: number;\n      path: string;\n      loop?: boolean;\n      width?: number;\n      height?: number;\n    },\n  ) {\n    this.setupTrack(props.audioPort, this.audio);\n    this.setupTrack(props.videoPort, this.video);\n  }\n\n  private setupTrack = (port: number, track: MediaStreamTrack) => {\n    let payloadType = 0;\n\n    const socket = createSocket(\"udp4\");\n    socket.bind(port);\n    socket.on(\"message\", async (buf) => {\n      const rtp = RtpPacket.deSerialize(buf);\n      if (!payloadType) {\n        payloadType = rtp.header.payloadType;\n      }\n\n      // detect gStreamer restarted\n      if (payloadType !== rtp.header.payloadType) {\n        payloadType = rtp.header.payloadType;\n        track.onSourceChanged.execute(rtp.header);\n      }\n\n      track.writeRtp(buf);\n    });\n  };\n\n  stop() {\n    this.stopped = true;\n    this.process.kill(\"SIGINT\");\n  }\n}\n\nexport class MediaPlayerMp4 extends MediaPlayer {\n  async start() {\n    let payloadType = 96;\n    const run = async () => {\n      if (payloadType > 100) payloadType = 96;\n\n      let cmd = \"\";\n      if (this.props.width && this.props.height) {\n        cmd = `gst-launch-1.0 filesrc location= ${this.props.path} ! \\\ndecodebin ! videoscale ! video/x-raw,width=${this.props.width},height=${\n          this.props.height\n        } ! x264enc ! \\\nh264parse ! rtph264pay config-interval=10 pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort}`;\n      } else {\n        cmd = `gst-launch-1.0 filesrc location= ${this.props.path} ! \\\nqtdemux name=d ! \\\nqueue ! h264parse ! rtph264pay config-interval=10 pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort} d. ! \\\nqueue ! aacparse ! avdec_aac ! audioresample ! audioconvert ! opusenc ! rtpopuspay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.audioPort}`;\n      }\n      console.log(cmd);\n      this.process = exec(cmd);\n      this.process.on(\"error\", (e) => console.error(\"gst error\", e));\n\n      if (this.props.loop) {\n        await new Promise((r) => this.process.on(\"close\", r));\n        if (!this.stopped) {\n          run();\n        }\n      }\n    };\n    await setImmediate();\n    run();\n  }\n}\n\nexport class MediaPlayerWebm extends MediaPlayer {\n  async start() {\n    let payloadType = 96;\n    const run = async () => {\n      if (payloadType > 100) payloadType = 96;\n\n      const cmd = `gst-launch-1.0 filesrc location=${this.props.path} ! matroskademux name=d \\\nd.video_0 ! queue ! rtpvp8pay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort} \\\nd.audio_0 ! queue ! rtpopuspay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.audioPort}`;\n      this.process = exec(cmd);\n\n      if (this.props.loop) {\n        await new Promise((r) => this.process.on(\"close\", r));\n        if (!this.stopped) {\n          run();\n        }\n      }\n    };\n    await setImmediate();\n    run();\n  }\n}\n"]}

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

@@ -1,27 +1,28 @@
-import { RTCDataChannel } from "./dataChannel";
+import type { RTCDataChannel } from "./dataChannel";
 import { EventTarget } from "./helper";
 import { type Address, Event, type InterfaceAddresses } from "./imports/common";
 import type { CandidatePair, Message, Protocol } from "./imports/ice";
-import { type Direction, MediaStream, type MediaStreamTrack, type RTCRtpCodecParameters, type RTCRtpHeaderExtensionParameters, RTCRtpReceiver, RTCRtpSender, RTCRtpTransceiver, type TransceiverOptions } from "./media";
-import { GroupDescription, MediaDescription, SessionDescription } from "./sdp";
-import { type DtlsKeys, RTCDtlsTransport } from "./transport/dtls";
-import { type IceGathererState, type RTCIceCandidate, type RTCIceCandidateInit, type RTCIceConnectionState, RTCIceTransport } from "./transport/ice";
-import { RTCSctpTransport } from "./transport/sctp";
-import type { ConnectionState, Kind, RTCSignalingState } from "./types/domain";
+import { type MediaStream, type MediaStreamTrack, type RTCRtpCodecParameters, type RTCRtpHeaderExtensionParameters, type RTCRtpReceiver, type RTCRtpSender, type RTCRtpTransceiver, type TransceiverOptions } from "./media";
+import { RTCStatsReport } from "./media/stats";
+import { type BundlePolicy, SessionDescription } from "./sdp";
+import { type RTCSessionDescriptionInit } from "./sdpManager";
+import type { DtlsKeys, RTCDtlsTransport } from "./transport/dtls";
+import type { RTCIceCandidate, RTCIceCandidateInit, RTCIceTransport } from "./transport/ice";
+import type { Kind, RTCSignalingState } from "./types/domain";
 import type { Callback, CallbackWithValue } from "./types/util";
 export declare class RTCPeerConnection extends EventTarget {
     readonly cname: string;
-    sctpTransport?: RTCSctpTransport;
     config: Required<PeerConfig>;
-    connectionState: ConnectionState;
-    iceConnectionState: RTCIceConnectionState;
-    iceGatheringState: IceGathererState;
     signalingState: RTCSignalingState;
     negotiationneeded: boolean;
-    private readonly transceivers;
-    private pushTransceiver;
-    private replaceTransceiver;
     needRestart: boolean;
+    private readonly router;
+    private readonly sdpManager;
+    private readonly transceiverManager;
+    private readonly sctpManager;
+    private readonly secureManager;
+    private isClosed;
+    private shouldNegotiationneeded;
     readonly iceGatheringStateChange: Event<["complete" | "new" | "gathering"]>;
     readonly iceConnectionStateChange: Event<["disconnected" | "closed" | "completed" | "new" | "connected" | "failed" | "checking"]>;
     readonly signalingStateChange: Event<["closed" | "stable" | "have-local-offer" | "have-remote-offer" | "have-local-pranswer" | "have-remote-pranswer"]>;
@@ -34,43 +35,42 @@ export declare class RTCPeerConnection extends EventTarget {
     readonly onTrack: Event<[MediaStreamTrack]>;
     ondatachannel?: CallbackWithValue<RTCDataChannelEvent>;
     onicecandidate?: CallbackWithValue<RTCPeerConnectionIceEvent>;
+    onicecandidateerror?: CallbackWithValue<any>;
+    onicegatheringstatechange?: CallbackWithValue<any>;
     onnegotiationneeded?: CallbackWithValue<any>;
     onsignalingstatechange?: CallbackWithValue<any>;
     ontrack?: CallbackWithValue<RTCTrackEvent>;
     onconnectionstatechange?: Callback;
     oniceconnectionstatechange?: Callback;
-    private readonly router;
-    private certificate?;
-    sctpRemotePort?: number;
-    private seenMid;
-    private currentLocalDescription?;
-    private currentRemoteDescription?;
-    private pendingLocalDescription?;
-    private pendingRemoteDescription?;
-    private isClosed;
-    private shouldNegotiationneeded;
+    constructor(config?: Partial<PeerConfig>);
+    get connectionState(): "disconnected" | "closed" | "new" | "connected" | "connecting" | "failed";
+    get iceConnectionState(): "disconnected" | "closed" | "completed" | "new" | "connected" | "failed" | "checking";
+    get iceGathererState(): "complete" | "new" | "gathering";
+    get iceGatheringState(): "complete" | "new" | "gathering";
     get dtlsTransports(): RTCDtlsTransport[];
+    get sctpTransport(): import(".").RTCSctpTransport | undefined;
+    get sctpRemotePort(): number | undefined;
     get iceTransports(): RTCIceTransport[];
     get extIdUriMap(): {
         [id: number]: string;
     };
     get iceGeneration(): number;
-    constructor(config?: Partial<PeerConfig>);
-    setConfiguration(config: Partial<PeerConfig>): void;
     get localDescription(): import("./sdp").RTCSessionDescription | undefined;
     get remoteDescription(): import("./sdp").RTCSessionDescription | undefined;
+    get remoteIsBundled(): import("./sdp").GroupDescription | undefined;
     /**@private */
     get _localDescription(): SessionDescription | undefined;
     /**@private */
     get _remoteDescription(): SessionDescription | undefined;
-    private getTransceiverByMid;
-    private getTransceiverByMLineIndex;
+    getTransceivers(): RTCRtpTransceiver[];
+    getSenders(): RTCRtpSender[];
+    getReceivers(): RTCRtpReceiver[];
+    setConfiguration(config: Partial<PeerConfig>): void;
     getConfiguration(): Required<PeerConfig>;
     createOffer({ iceRestart }?: {
         iceRestart?: boolean;
     }): Promise<import("./sdp").RTCSessionDescription>;
-    private assignTransceiverCodecs;
-    buildOfferSdp(): SessionDescription;
+    private createSctpTransport;
     createDataChannel(label: string, options?: Partial<{
         maxPacketLifeTime?: number;
         protocol: string;
@@ -81,51 +81,27 @@ export declare class RTCPeerConnection extends EventTarget {
     }>): RTCDataChannel;
     removeTrack(sender: RTCRtpSender): void;
     private needNegotiation;
-    private createTransport;
-    private createSctpTransport;
-    setLocalDescription(sessionDescription: {
+    private findOrCreateTransport;
+    setLocalDescription(sessionDescription?: {
         type: "offer" | "answer";
         sdp: string;
     }): Promise<SessionDescription>;
     private gatherCandidates;
-    private setLocal;
-    private getTransportByMid;
-    private getTransportByMLineIndex;
     addIceCandidate(candidateMessage: RTCIceCandidate | RTCIceCandidateInit): Promise<void>;
     private connect;
-    private getLocalRtpParams;
-    private getRemoteRtpParams;
-    get remoteIsBundled(): GroupDescription | undefined;
     restartIce(): void;
     setRemoteDescription(sessionDescription: RTCSessionDescriptionInit): Promise<void>;
-    private setRemoteRTP;
-    private setRemoteSCTP;
-    private validateDescription;
-    private fireOnTrack;
     addTransceiver(trackOrKind: Kind | MediaStreamTrack, options?: Partial<TransceiverOptions>): RTCRtpTransceiver;
-    private _addTransceiver;
-    getTransceivers(): RTCRtpTransceiver[];
-    getSenders(): RTCRtpSender[];
-    getReceivers(): RTCRtpReceiver[];
     addTrack(track: MediaStreamTrack, 
     /**todo impl */
     ms?: MediaStream): RTCRtpSender;
-    private ensureCerts;
     createAnswer(): Promise<import("./sdp").RTCSessionDescription>;
-    private buildAnswer;
-    close(): Promise<void>;
     private assertNotClosed;
-    private updateIceGatheringState;
-    private updateIceConnectionState;
     private setSignalingState;
-    private setConnectionState;
-    private dispose;
+    private createPeerConnectionStats;
+    getStats(selector?: MediaStreamTrack | null): Promise<RTCStatsReport>;
+    close(): Promise<void>;
 }
-export declare function createMediaDescriptionForTransceiver(transceiver: RTCRtpTransceiver, cname: string, direction: Direction): MediaDescription;
-export declare function createMediaDescriptionForSctp(sctp: RTCSctpTransport): MediaDescription;
-export declare function addTransportDescription(media: MediaDescription, dtlsTransport: RTCDtlsTransport): void;
-export declare function allocateMid(mids: Set<string>, type: "dc" | "av" | ""): string;
-export type BundlePolicy = "max-compat" | "max-bundle" | "disable";
 export interface PeerConfig {
     codecs: Partial<{
         /**
@@ -193,8 +169,3 @@ export interface RTCDataChannelEvent {
 export interface RTCPeerConnectionIceEvent {
     candidate?: RTCIceCandidate;
 }
-export interface RTCSessionDescriptionInit {
-    sdp?: string;
-    type: RTCSdpType;
-}
-export type RTCSdpType = "answer" | "offer" | "pranswer" | "rollback";