werift 0.19.3 → 0.19.5

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

@@ -1 +1 @@
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{ unlink } from \"fs/promises\";\nimport { EventDisposer } from \"rx.mini\";\n\nimport { MediaWriter } from \".\";\nimport { MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n  DepacketizeCallback,\n  JitterBufferCallback,\n  LipsyncCallback,\n  NtpTimeCallback,\n  RtcpSourceCallback,\n  RtpSourceCallback,\n  SupportedCodec,\n  WebmCallback,\n  saveToFileSystem,\n} from \"../../../../../rtp/src/extra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceCallback[] = [];\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 WebmCallback(inputTracks, {\n      duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n    const lipsync = new LipsyncCallback();\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 ntpTime = 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(clockRate);\n\n        rtpSource.pipe(jitterBuffer.input);\n        rtcpSource.pipe(ntpTime.input);\n\n        jitterBuffer.pipe(ntpTime.input);\n        ntpTime.pipe(depacketizer.input);\n        depacketizer.pipe(lipsync.inputVideo);\n        lipsync.pipeVideo(webm.inputVideo);\n      } else {\n        const depacketizer = new DepacketizeCallback(codec);\n\n        rtpSource.pipe(ntpTime.input);\n        rtcpSource.pipe(ntpTime.input);\n\n        ntpTime.pipe(depacketizer.input);\n        depacketizer.pipe(lipsync.inputAudio);\n        lipsync.pipeAudio(webm.inputAudio);\n      }\n\n      return rtpSource;\n    });\n    webm.pipe(saveToFileSystem(this.path));\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n    this.unSubscribers.dispose();\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,0CAAqC;AACrC,qCAAwC;AAExC,wBAAgC;AAChC,gCAA8D;AAC9D,wDAWsC;AAEtC,MAAM,UAAU,GAAG,yDAAyD,CAAC;AAE7E,MAAa,WAAY,SAAQ,cAAW;IAA5C;;QACE;;;;mBAAkC,EAAE;WAAC;QAErC;;;;mBAAgB,IAAI,uBAAa,EAAE;WAAC;IAgHtC,CAAC;IA9GC,KAAK,CAAC,KAAK,CAAC,MAA0B;QACpC,IAAI,IAAI,CAAC,KAAK,CAAC,IAAI,EAAE,CAAC;YACpB,MAAM,IAAA,iBAAM,EAAC,IAAI,CAAC,KAAK,CAAC,IAAI,CAAC,CAAC,KAAK,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC;QAChD,CAAC;QAED,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,CAAC;gBAC3B,MAAM,KAAK,GAAG,CAAC,GAAmB,EAAE;oBAClC,QAAQ,KAAK,CAAC,KAAK,EAAE,IAAI,CAAC,WAAW,EAAyB,EAAE,CAAC;wBAC/D,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;oBACP,CAAC;gBACH,CAAC,CAAC,EAAE,CAAC;gBACL,OAAO;oBACL,IAAI,EAAE,OAAgB;oBACtB,KAAK;oBACL,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,KAAK,EAAE,IAAI,CAAC,KAAK,CAAC,KAAK,IAAI,GAAG;oBAC9B,MAAM,EAAE,IAAI,CAAC,KAAK,CAAC,MAAM,IAAI,GAAG;oBAChC,WAAW;oBACX,KAAK;iBACN,CAAC;YACJ,CAAC;iBAAM,CAAC;gBACN,OAAO;oBACL,IAAI,EAAE,OAAgB;oBACtB,KAAK,EAAE,MAAe;oBACtB,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK;iBACN,CAAC;YACJ,CAAC;QACH,CAAC,CAAC,CAAC;QAEH,MAAM,IAAI,GAAG,IAAI,oBAAY,CAAC,WAAW,EAAE;YACzC,QAAQ,EAAE,IAAI,CAAC,KAAK,CAAC,eAAe,IAAI,IAAI,GAAG,EAAE,GAAG,EAAE,GAAG,EAAE;SAC5D,CAAC,CAAC;QACH,MAAM,OAAO,GAAG,IAAI,uBAAe,EAAE,CAAC;QAEtC,IAAI,CAAC,UAAU,GAAG,WAAW,CAAC,GAAG,CAAC,CAAC,EAAE,KAAK,EAAE,SAAS,EAAE,KAAK,EAAE,EAAE,EAAE;YAChE,MAAM,SAAS,GAAG,IAAI,yBAAiB,EAAE,CAAC;YAC1C,MAAM,UAAU,GAAG,IAAI,0BAAkB,EAAE,CAAC;YAC5C,KAAK,CAAC,YAAY;iBACf,SAAS,CAAC,CAAC,GAAG,EAAE,EAAE;gBACjB,SAAS,CAAC,KAAK,CAAC,GAAG,CAAC,KAAK,EAAE,CAAC,CAAC;YAC/B,CAAC,CAAC;iBACD,QAAQ,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC;YAChC,KAAK,CAAC,aAAa;iBAChB,SAAS,CAAC,CAAC,IAAI,EAAE,EAAE;gBAClB,UAAU,CAAC,KAAK,CAAC,IAAI,CAAC,CAAC;YACzB,CAAC,CAAC;iBACD,QAAQ,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC;YAChC,MAAM,IAAI,GAAG,IAAI,CAAC,KAAK,CAAC,UAAU;gBAChC,CAAC,CAAC,IAAI,uBAAe,CAAC,SAAS,CAAC;gBAChC,CAAC,CAAC,IAAI,uBAAe,CAAC,SAAS,CAAC,CAAC;YAEnC,IAAI,KAAK,CAAC,IAAI,KAAK,OAAO,EAAE,CAAC;gBAC3B,MAAM,YAAY,GAAG,IAAI,2BAAmB,CAAC,KAAK,EAAE;oBAClD,uBAAuB,EAAE,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,MAAM;iBACzC,CAAC,CAAC;gBACH,MAAM,YAAY,GAAG,IAAI,4BAAoB,CAAC,SAAS,CAAC,CAAC;gBAEzD,SAAS,CAAC,IAAI,CAAC,YAAY,CAAC,KAAK,CAAC,CAAC;gBACnC,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC;gBAE5B,YAAY,CAAC,IAAI,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC;gBAC9B,IAAI,CAAC,IAAI,CAAC,YAAY,CAAC,KAAK,CAAC,CAAC;gBAC9B,YAAY,CAAC,IAAI,CAAC,OAAO,CAAC,UAAU,CAAC,CAAC;gBACtC,OAAO,CAAC,SAAS,CAAC,IAAI,CAAC,UAAU,CAAC,CAAC;YACrC,CAAC;iBAAM,CAAC;gBACN,MAAM,YAAY,GAAG,IAAI,2BAAmB,CAAC,KAAK,CAAC,CAAC;gBAEpD,SAAS,CAAC,IAAI,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC;gBAC3B,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC;gBAE5B,IAAI,CAAC,IAAI,CAAC,YAAY,CAAC,KAAK,CAAC,CAAC;gBAC9B,YAAY,CAAC,IAAI,CAAC,OAAO,CAAC,UAAU,CAAC,CAAC;gBACtC,OAAO,CAAC,SAAS,CAAC,IAAI,CAAC,UAAU,CAAC,CAAC;YACrC,CAAC;YAED,OAAO,SAAS,CAAC;QACnB,CAAC,CAAC,CAAC;QACH,IAAI,IAAI,CAAC,KAAK,CAAC,IAAI,EAAE,CAAC;YACpB,IAAI,CAAC,IAAI,CAAC,IAAA,wBAAgB,EAAC,IAAI,CAAC,KAAK,CAAC,IAAI,CAAC,CAAC,CAAC;QAC/C,CAAC;aAAM,IAAI,IAAI,CAAC,KAAK,CAAC,MAAM,EAAE,CAAC;YAC7B,IAAI,CAAC,IAAI,CAAC,KAAK,EAAE,CAAC,EAAE,EAAE;gBACpB,IAAI,CAAC,KAAK,CAAC,MAAM,CAAC,KAAK,CAAC,CAAC,CAAC,CAAC;YAC7B,CAAC,CAAC,CAAC;QACL,CAAC;IACH,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;QACxD,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 { unlink } from \"fs/promises\";\nimport { EventDisposer } from \"rx.mini\";\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  type SupportedCodec,\n  WebmCallback,\n  saveToFileSystem,\n} from \"../../../../../rtp/src/extra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceCallback[] = [];\n\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 = 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.props.width ?? 640,\n          height: this.props.height ?? 360,\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 WebmCallback(inputTracks, {\n      duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n    const lipsync = new LipsyncCallback();\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(clockRate);\n\n        rtpSource.pipe(jitterBuffer.input);\n        rtcpSource.pipe(time.input);\n\n        jitterBuffer.pipe(time.input);\n        time.pipe(depacketizer.input);\n        depacketizer.pipe(lipsync.inputVideo);\n        lipsync.pipeVideo(webm.inputVideo);\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        depacketizer.pipe(lipsync.inputAudio);\n        lipsync.pipeAudio(webm.inputAudio);\n      }\n\n      return rtpSource;\n    });\n    if (this.props.path) {\n      webm.pipe(saveToFileSystem(this.props.path));\n    } else if (this.props.stream) {\n      webm.pipe(async (o) => {\n        this.props.stream.write(o);\n      });\n    }\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\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.d.ts

@@ -1,11 +1,10 @@
-/// <reference types="node" />
-import { ChildProcess } from "child_process";
+import { type ChildProcess } from "child_process";
 import { MediaStreamTrack } from "../media/track";
 export declare const getUserMedia: ({ path, loop, width, height, }: {
     path: string;
-    loop?: boolean | undefined;
-    width?: number | undefined;
-    height?: number | undefined;
+    loop?: boolean;
+    width?: number;
+    height?: number;
 }) => Promise<MediaPlayerMp4 | MediaPlayerWebm>;
 declare abstract class MediaPlayer {
     protected props: {

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

@@ -1 +1 @@
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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

@@ -1,21 +1,21 @@
 import Event from "rx.mini";
-import { Address, CandidatePair, InterfaceAddresses } from ".";
-import { DtlsKeys } from ".";
-import { Message } from "../../ice/src/stun/message";
-import { Protocol } from "../../ice/src/types/model";
+import { type Address, type CandidatePair, type InterfaceAddresses } from ".";
+import { type DtlsKeys } from ".";
+import type { Message } from "../../ice/src/stun/message";
+import type { Protocol } from "../../ice/src/types/model";
 import { RTCDataChannel } from "./dataChannel";
 import { EventTarget } from "./helper";
-import { RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters } from "./media/parameters";
+import { RTCRtpCodecParameters, type RTCRtpHeaderExtensionParameters } from "./media/parameters";
 import { RTCRtpReceiver } from "./media/rtpReceiver";
 import { RTCRtpSender } from "./media/rtpSender";
-import { Direction, RTCRtpTransceiver, TransceiverOptions } from "./media/rtpTransceiver";
-import { MediaStream, MediaStreamTrack } from "./media/track";
+import { type Direction, RTCRtpTransceiver, type TransceiverOptions } from "./media/rtpTransceiver";
+import { MediaStream, type MediaStreamTrack } from "./media/track";
 import { GroupDescription, MediaDescription, SessionDescription } from "./sdp";
 import { RTCDtlsTransport } from "./transport/dtls";
-import { IceGathererState, RTCIceCandidate, RTCIceConnectionState, RTCIceTransport } from "./transport/ice";
+import { type IceGathererState, type RTCIceCandidate, type RTCIceConnectionState, RTCIceTransport } from "./transport/ice";
 import { RTCSctpTransport } from "./transport/sctp";
-import { ConnectionState, Kind, RTCSignalingState } from "./types/domain";
-import { Callback, CallbackWithValue } from "./types/util";
+import type { ConnectionState, Kind, RTCSignalingState } from "./types/domain";
+import type { Callback, CallbackWithValue } from "./types/util";
 export declare class RTCPeerConnection extends EventTarget {
     readonly cname: string;
     sctpTransport?: RTCSctpTransport;

package/lib/webrtc/src/peerConnection.js

@@ -26,7 +26,11 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
     return (mod && mod.__esModule) ? mod : { "default": mod };
 };
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.defaultPeerConfig = exports.findCodecByMimeType = exports.allocateMid = exports.addTransportDescription = exports.createMediaDescriptionForSctp = exports.createMediaDescriptionForTransceiver = exports.RTCPeerConnection = void 0;
+exports.defaultPeerConfig = exports.findCodecByMimeType = exports.RTCPeerConnection = void 0;
+exports.createMediaDescriptionForTransceiver = createMediaDescriptionForTransceiver;
+exports.createMediaDescriptionForSctp = createMediaDescriptionForSctp;
+exports.addTransportDescription = addTransportDescription;
+exports.allocateMid = allocateMid;
 const debug_1 = __importDefault(require("debug"));
 const cloneDeep_1 = __importDefault(require("lodash/cloneDeep"));
 const isEqual_1 = __importDefault(require("lodash/isEqual"));
@@ -594,8 +598,10 @@ class RTCPeerConnection extends helper_1.EventTarget {
             this.updateIceConnectionState();
         });
         iceTransport.iceGather.onIceCandidate = (candidate) => {
-            if (!this.localDescription)
+            if (!this.localDescription) {
+                log("localDescription not found when ice candidate was gathered");
                 return;
+            }
             if (this.config.bundlePolicy === "max-bundle" || this.remoteIsBundled) {
                 candidate.sdpMLineIndex = 0;
                 const media = this._localDescription?.media[0];
@@ -711,13 +717,6 @@ class RTCPeerConnection extends helper_1.EventTarget {
         }
         // for trickle ice
         this.setLocal(description);
-        // connect transports
-        if (description.type === "answer") {
-            this.connect().catch((err) => {
-                log("connect failed", err);
-                this.setConnectionState("failed");
-            });
-        }
         // # gather candidates
         const connected = this.iceTransports.find((transport) => transport.state === "connected");
         if (this.remoteIsBundled && connected) {
@@ -727,6 +726,13 @@ class RTCPeerConnection extends helper_1.EventTarget {
         else {
             await Promise.allSettled(this.iceTransports.map((iceTransport) => iceTransport.iceGather.gather()));
         }
+        // connect transports
+        if (description.type === "answer") {
+            this.connect().catch((err) => {
+                log("connect failed", err);
+                this.setConnectionState("failed");
+            });
+        }
         description.media
             .filter((m) => ["audio", "video"].includes(m.kind))
             .forEach((m, i) => {
@@ -984,14 +990,6 @@ class RTCPeerConnection extends helper_1.EventTarget {
                 this.setConnectionState("failed");
             });
         }
-        const connected = this.iceTransports.find((transport) => transport.state === "connected");
-        if (this.remoteIsBundled && connected) {
-            // no need to gather ice candidates on an existing bundled connection
-            await connected.iceGather.gather();
-        }
-        else {
-            await Promise.allSettled(transports.map((iceTransport) => iceTransport.iceGather.gather()));
-        }
         this.negotiationneeded = false;
         if (this.shouldNegotiationneeded) {
             this.needNegotiation();
@@ -1431,7 +1429,6 @@ function createMediaDescriptionForTransceiver(transceiver, cname, direction) {
     addTransportDescription(media, transceiver.dtlsTransport);
     return media;
 }
-exports.createMediaDescriptionForTransceiver = createMediaDescriptionForTransceiver;
 function createMediaDescriptionForSctp(sctp) {
     const media = new sdp_1.MediaDescription("application", const_1.DISCARD_PORT, "UDP/DTLS/SCTP", ["webrtc-datachannel"]);
     media.sctpPort = sctp.port;
@@ -1440,7 +1437,6 @@ function createMediaDescriptionForSctp(sctp) {
     addTransportDescription(media, sctp.dtlsTransport);
     return media;
 }
-exports.createMediaDescriptionForSctp = createMediaDescriptionForSctp;
 function addTransportDescription(media, dtlsTransport) {
     const iceTransport = dtlsTransport.iceTransport;
     const iceGatherer = iceTransport.iceGather;
@@ -1462,7 +1458,6 @@ function addTransportDescription(media, dtlsTransport) {
         }
     }
 }
-exports.addTransportDescription = addTransportDescription;
 function allocateMid(mids, type) {
     let mid = "";
     for (let i = 0;;) {
@@ -1475,7 +1470,6 @@ function allocateMid(mids, type) {
     mids.add(mid);
     return mid;
 }
-exports.allocateMid = allocateMid;
 const findCodecByMimeType = (codecs, target) => codecs.find((localCodec) => localCodec.mimeType.toLowerCase() === target.mimeType.toLowerCase())
     ? target
     : undefined;