werift 0.19.0 → 0.19.2

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

@@ -1 +1 @@
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{ ChildProcess, exec } from \"child_process\";\nimport { createSocket } from \"dgram\";\nimport { setImmediate } from \"timers/promises\";\nimport { v4 } from \"uuid\";\n\nimport { randomPort } from \"../../../common/src\";\nimport { RtpPacket } from \"../../../rtp/src\";\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 = v4();\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

@@ -1,8 +1,8 @@
 import Event from "rx.mini";
-import { Message } from "../../ice/src/stun/message";
-import { Protocol } from "../../ice/src/types/model";
 import { Address, CandidatePair, InterfaceAddresses } from ".";
 import { DtlsKeys } from ".";
+import { Message } from "../../ice/src/stun/message";
+import { Protocol } from "../../ice/src/types/model";
 import { RTCDataChannel } from "./dataChannel";
 import { EventTarget } from "./helper";
 import { RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters } from "./media/parameters";
@@ -68,6 +68,7 @@ export declare class RTCPeerConnection extends EventTarget {
     private get _remoteDescription();
     private getTransceiverByMid;
     private getTransceiverByMLineIndex;
+    getConfiguration(): Required<PeerConfig>;
     createOffer(): Promise<import("./sdp").RTCSessionDescription>;
     private assignTransceiverCodecs;
     buildOfferSdp(): SessionDescription;

package/lib/webrtc/src/peerConnection.js

@@ -372,13 +372,15 @@ class RTCPeerConnection extends helper_1.EventTarget {
         });
     }
     get localDescription() {
-        if (!this._localDescription)
-            return;
+        if (!this._localDescription) {
+            return undefined;
+        }
         return this._localDescription.toJSON();
     }
     get remoteDescription() {
-        if (!this._remoteDescription)
-            return;
+        if (!this._remoteDescription) {
+            return undefined;
+        }
         return this._remoteDescription.toJSON();
     }
     get _localDescription() {
@@ -393,6 +395,9 @@ class RTCPeerConnection extends helper_1.EventTarget {
     getTransceiverByMLineIndex(index) {
         return this.transceivers.find((transceiver) => transceiver.mLineIndex === index);
     }
+    getConfiguration() {
+        return this.config;
+    }
     async createOffer() {
         await this.ensureCerts();
         const description = this.buildOfferSdp();
@@ -630,7 +635,7 @@ class RTCPeerConnection extends helper_1.EventTarget {
     }
     createSctpTransport() {
         const dtlsTransport = this.createTransport([
-            const_1.SRTP_PROFILE.SRTP_AEAD_AES_128_GCM,
+            const_1.SRTP_PROFILE.SRTP_AEAD_AES_128_GCM, // prefer
             const_1.SRTP_PROFILE.SRTP_AES128_CM_HMAC_SHA1_80,
         ]);
         const sctp = new sctp_1.RTCSctpTransport();
@@ -852,8 +857,9 @@ class RTCPeerConnection extends helper_1.EventTarget {
     }
     get remoteIsBundled() {
         const remoteSdp = this._remoteDescription;
-        if (!remoteSdp)
-            return;
+        if (!remoteSdp) {
+            return undefined;
+        }
         const bundle = remoteSdp.group.find((g) => g.semantic === "BUNDLE" && this.config.bundlePolicy !== "disable");
         return bundle;
     }
@@ -1128,7 +1134,7 @@ class RTCPeerConnection extends helper_1.EventTarget {
         const kind = typeof trackOrKind === "string" ? trackOrKind : trackOrKind.kind;
         const direction = options.direction || "sendrecv";
         const dtlsTransport = this.createTransport([
-            const_1.SRTP_PROFILE.SRTP_AEAD_AES_128_GCM,
+            const_1.SRTP_PROFILE.SRTP_AEAD_AES_128_GCM, // prefer
             const_1.SRTP_PROFILE.SRTP_AES128_CM_HMAC_SHA1_80,
         ]);
         const sender = new rtpSender_1.RTCRtpSender(trackOrKind);