werift 0.20.0 → 0.20.1

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 { 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    if (inputTracks.length === 1 || this.props.disableNtp) {\n      this.props.disableLipSync = true;\n    }\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          bufferSize: this.props.jitterBufferSize,\n          latency: this.props.jitterBufferLatency,\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(saveToFileSystem(this.props.path));\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    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 { 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  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    if (inputTracks.length === 1 || this.props.disableNtp) {\n      this.props.disableLipSync = true;\n    }\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          bufferSize: this.props.jitterBufferSize,\n          latency: this.props.jitterBufferLatency,\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(saveToFileSystem(this.props.path));\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    this.unSubscribers.dispose();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}

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

@@ -1,4 +1,4 @@
-import Event from "rx.mini";
+import { Event } from "./imports/common";
 import { type Address, type CandidatePair, type InterfaceAddresses } from ".";
 import { type DtlsKeys } from ".";
 import type { Message } from "../../ice/src/stun/message";
@@ -30,7 +30,7 @@ export declare class RTCPeerConnection extends EventTarget {
     private pushTransceiver;
     private replaceTransceiver;
     candidatesSent: Set<string>;
-    readonly iceGatheringStateChange: Event<["new" | "complete" | "gathering"]>;
+    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"]>;
     readonly connectionStateChange: Event<["disconnected" | "closed" | "new" | "connected" | "connecting" | "failed"]>;
@@ -151,6 +151,7 @@ export interface PeerConfig {
     iceAdditionalHostAddresses: string[] | undefined;
     iceUseIpv4: boolean;
     iceUseIpv6: boolean;
+    forceTurnTCP: boolean;
     /** If provided, is called on each STUN request.
      * Return `true` if a STUN response should be sent, false if it should be skipped. */
     iceFilterStunResponse: ((message: Message, addr: Address, protocol: Protocol) => boolean) | undefined;

package/lib/webrtc/src/peerConnection.js

@@ -34,8 +34,8 @@ exports.allocateMid = allocateMid;
 const debug_1 = __importDefault(require("debug"));
 const cloneDeep_1 = __importDefault(require("lodash/cloneDeep"));
 const isEqual_1 = __importDefault(require("lodash/isEqual"));
-const rx_mini_1 = __importDefault(require("rx.mini"));
 const uuid = __importStar(require("uuid"));
+const common_1 = require("./imports/common");
 const _1 = require(".");
 const _2 = require(".");
 const const_1 = require("./const");
@@ -150,61 +150,61 @@ class RTCPeerConnection extends helper_1.EventTarget {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "iceConnectionStateChange", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "signalingStateChange", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "connectionStateChange", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "onDataChannel", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "onRemoteTransceiverAdded", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "onTransceiverAdded", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "onIceCandidate", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "onNegotiationneeded", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "onTrack", {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: new rx_mini_1.default()
+            value: new common_1.Event()
         });
         Object.defineProperty(this, "ondatachannel", {
             enumerable: true,
@@ -586,6 +586,7 @@ class RTCPeerConnection extends helper_1.EventTarget {
             filterCandidatePair: this.config.iceFilterCandidatePair,
             useIpv4: this.config.iceUseIpv4,
             useIpv6: this.config.iceUseIpv6,
+            turnTransport: this.config.forceTurnTCP === true ? "tcp" : "udp",
         });
         if (existing) {
             iceGatherer.connection.localUserName = existing.connection.localUserName;
@@ -1525,5 +1526,6 @@ exports.defaultPeerConfig = {
     bundlePolicy: "max-compat",
     debug: {},
     midSuffix: false,
+    forceTurnTCP: false,
 };
 //# sourceMappingURL=peerConnection.js.map