werift 0.12.8 → 0.13.2

package/lib/webrtc/src/index.js

@@ -13,12 +13,16 @@ Object.defineProperty(exports, "__esModule", { value: true });
 __exportStar(require("../../ice/src"), exports);
 __exportStar(require("../../rtp/src"), exports);
 __exportStar(require("./dataChannel"), exports);
-__exportStar(require("./extension/rtcpFeedback"), exports);
-__exportStar(require("./extension/rtpExtension"), exports);
 __exportStar(require("./helper"), exports);
+__exportStar(require("./media/extension/rtcpFeedback"), exports);
+__exportStar(require("./media/extension/rtpExtension"), exports);
 __exportStar(require("./media/parameters"), exports);
 __exportStar(require("./media/rtpTransceiver"), exports);
 __exportStar(require("./media/track"), exports);
+__exportStar(require("./nonstandard/lipsync"), exports);
+__exportStar(require("./nonstandard/recorder"), exports);
+__exportStar(require("./nonstandard/sampleBuilder"), exports);
+__exportStar(require("./nonstandard/userMedia"), exports);
 __exportStar(require("./peerConnection"), exports);
 __exportStar(require("./sdp"), exports);
 __exportStar(require("./transport/dtls"), exports);

package/lib/webrtc/src/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../src/index.ts"],"names":[],"mappings":";;;;;;;;;;;;AAAA,gDAA8B;AAC9B,gDAA8B;AAC9B,gDAA8B;AAC9B,2DAAyC;AACzC,2DAAyC;AACzC,2CAAyB;AACzB,qDAAmC;AACnC,yDAAuC;AACvC,gDAA8B;AAC9B,mDAAiC;AACjC,wCAAsB;AACtB,mDAAiC;AACjC,kDAAgC;AAChC,mDAAiC;AACjC,iDAA+B;AAC/B,0CAAwB","sourcesContent":["export * from \"../../ice/src\";\nexport * from \"../../rtp/src\";\nexport * from \"./dataChannel\";\nexport * from \"./extension/rtcpFeedback\";\nexport * from \"./extension/rtpExtension\";\nexport * from \"./helper\";\nexport * from \"./media/parameters\";\nexport * from \"./media/rtpTransceiver\";\nexport * from \"./media/track\";\nexport * from \"./peerConnection\";\nexport * from \"./sdp\";\nexport * from \"./transport/dtls\";\nexport * from \"./transport/ice\";\nexport * from \"./transport/sctp\";\nexport * from \"./types/domain\";\nexport * from \"./utils\";\n"]}
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../src/index.ts"],"names":[],"mappings":";;;;;;;;;;;;AAAA,gDAA8B;AAC9B,gDAA8B;AAC9B,gDAA8B;AAC9B,2CAAyB;AACzB,iEAA+C;AAC/C,iEAA+C;AAC/C,qDAAmC;AACnC,yDAAuC;AACvC,gDAA8B;AAC9B,wDAAsC;AACtC,yDAAuC;AACvC,8DAA4C;AAC5C,0DAAwC;AACxC,mDAAiC;AACjC,wCAAsB;AACtB,mDAAiC;AACjC,kDAAgC;AAChC,mDAAiC;AACjC,iDAA+B;AAC/B,0CAAwB","sourcesContent":["export * from \"../../ice/src\";\nexport * from \"../../rtp/src\";\nexport * from \"./dataChannel\";\nexport * from \"./helper\";\nexport * from \"./media/extension/rtcpFeedback\";\nexport * from \"./media/extension/rtpExtension\";\nexport * from \"./media/parameters\";\nexport * from \"./media/rtpTransceiver\";\nexport * from \"./media/track\";\nexport * from \"./nonstandard/lipsync\";\nexport * from \"./nonstandard/recorder\";\nexport * from \"./nonstandard/sampleBuilder\";\nexport * from \"./nonstandard/userMedia\";\nexport * from \"./peerConnection\";\nexport * from \"./sdp\";\nexport * from \"./transport/dtls\";\nexport * from \"./transport/ice\";\nexport * from \"./transport/sctp\";\nexport * from \"./types/domain\";\nexport * from \"./utils\";\n"]}

package/lib/webrtc/src/{extension → media/extension}/rtcpFeedback.d.ts

@@ -1,4 +1,4 @@
-import { RTCPFB } from "../media/parameters";
+import { RTCPFB } from "../parameters";
 export declare const useFIR: () => RTCPFB;
 export declare const useNACK: () => RTCPFB;
 export declare const usePLI: () => RTCPFB;

package/lib/webrtc/src/media/extension/rtcpFeedback.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"rtcpFeedback.js","sourceRoot":"","sources":["../../../../../src/media/extension/rtcpFeedback.ts"],"names":[],"mappings":";;;AAEO,MAAM,MAAM,GAAG,GAAW,EAAE,CAAC,CAAC,EAAE,IAAI,EAAE,KAAK,EAAE,SAAS,EAAE,KAAK,EAAE,CAAC,CAAC;AAA3D,QAAA,MAAM,UAAqD;AAEjE,MAAM,OAAO,GAAG,GAAW,EAAE,CAAC,CAAC,EAAE,IAAI,EAAE,MAAM,EAAE,CAAC,CAAC;AAA3C,QAAA,OAAO,WAAoC;AAEjD,MAAM,MAAM,GAAG,GAAW,EAAE,CAAC,CAAC,EAAE,IAAI,EAAE,MAAM,EAAE,SAAS,EAAE,KAAK,EAAE,CAAC,CAAC;AAA5D,QAAA,MAAM,UAAsD;AAElE,MAAM,OAAO,GAAG,GAAW,EAAE,CAAC,CAAC,EAAE,IAAI,EAAE,WAAW,EAAE,CAAC,CAAC;AAAhD,QAAA,OAAO,WAAyC;AAEtD,MAAM,OAAO,GAAG,GAAW,EAAE,CAAC,CAAC,EAAE,IAAI,EAAE,cAAc,EAAE,CAAC,CAAC;AAAnD,QAAA,OAAO,WAA4C","sourcesContent":["import { RTCPFB } from \"../parameters\";\n\nexport const useFIR = (): RTCPFB => ({ type: \"ccm\", parameter: \"fir\" });\n\nexport const useNACK = (): RTCPFB => ({ type: \"nack\" });\n\nexport const usePLI = (): RTCPFB => ({ type: \"nack\", parameter: \"pli\" });\n\nexport const useREMB = (): RTCPFB => ({ type: \"goog-remb\" });\n\nexport const useTWCC = (): RTCPFB => ({ type: \"transport-cc\" });\n"]}

package/lib/webrtc/src/{extension → media/extension}/rtpExtension.d.ts

@@ -1,4 +1,4 @@
-import { RTCRtpHeaderExtensionParameters } from "../media/parameters";
+import { RTCRtpHeaderExtensionParameters } from "../parameters";
 export declare const RTP_EXTENSION_URI: {
     readonly sdesMid: "urn:ietf:params:rtp-hdrext:sdes:mid";
     readonly sdesRTPStreamID: "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";

package/lib/webrtc/src/{extension → media/extension}/rtpExtension.js

@@ -1,7 +1,7 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.useAbsSendTime = exports.useTransportWideCC = exports.useRepairedRtpStreamId = exports.useSdesRTPStreamId = exports.useSdesMid = exports.RTP_EXTENSION_URI = void 0;
-const parameters_1 = require("../media/parameters");
+const parameters_1 = require("../parameters");
 exports.RTP_EXTENSION_URI = {
     sdesMid: "urn:ietf:params:rtp-hdrext:sdes:mid",
     sdesRTPStreamID: "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id",

package/lib/webrtc/src/media/extension/rtpExtension.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"rtpExtension.js","sourceRoot":"","sources":["../../../../../src/media/extension/rtpExtension.ts"],"names":[],"mappings":";;;AAAA,8CAAgE;AAEnD,QAAA,iBAAiB,GAAG;IAC/B,OAAO,EAAE,qCAAqC;IAC9C,eAAe,EAAE,+CAA+C;IAChE,mBAAmB,EAAE,wDAAwD;IAC7E,eAAe,EACb,2EAA2E;IAC7E,WAAW,EAAE,4DAA4D;CACjE,CAAC;AAEX,SAAgB,UAAU;IACxB,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,yBAAiB,CAAC,OAAO;KAC/B,CAAC,CAAC;AACL,CAAC;AAJD,gCAIC;AAED,SAAgB,kBAAkB;IAChC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,yBAAiB,CAAC,eAAe;KACvC,CAAC,CAAC;AACL,CAAC;AAJD,gDAIC;AAED,SAAgB,sBAAsB;IACpC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,yBAAiB,CAAC,mBAAmB;KAC3C,CAAC,CAAC;AACL,CAAC;AAJD,wDAIC;AAED,SAAgB,kBAAkB;IAChC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,yBAAiB,CAAC,eAAe;KACvC,CAAC,CAAC;AACL,CAAC;AAJD,gDAIC;AAED,SAAgB,cAAc;IAC5B,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,yBAAiB,CAAC,WAAW;KACnC,CAAC,CAAC;AACL,CAAC;AAJD,wCAIC","sourcesContent":["import { RTCRtpHeaderExtensionParameters } from \"../parameters\";\n\nexport const RTP_EXTENSION_URI = {\n  sdesMid: \"urn:ietf:params:rtp-hdrext:sdes:mid\",\n  sdesRTPStreamID: \"urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\",\n  repairedRtpStreamId: \"urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\",\n  transportWideCC:\n    \"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\",\n  absSendTime: \"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\",\n} as const;\n\nexport function useSdesMid() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.sdesMid,\n  });\n}\n\nexport function useSdesRTPStreamId() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.sdesRTPStreamID,\n  });\n}\n\nexport function useRepairedRtpStreamId() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.repairedRtpStreamId,\n  });\n}\n\nexport function useTransportWideCC() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.transportWideCC,\n  });\n}\n\nexport function useAbsSendTime() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.absSendTime,\n  });\n}\n"]}

package/lib/webrtc/src/media/{nack.d.ts → receiver/nack.d.ts}

@@ -1,6 +1,6 @@
 import Event from "rx.mini";
-import { GenericNack, RtpPacket } from "../../../rtp/src";
-import { RTCRtpReceiver } from "./rtpReceiver";
+import { GenericNack, RtpPacket } from "../../../../rtp/src";
+import { RTCRtpReceiver } from "../rtpReceiver";
 export declare class Nack {
     private receiver;
     private newEstSeqNum;

package/lib/webrtc/src/media/{nack.js → receiver/nack.js}

@@ -6,8 +6,8 @@ Object.defineProperty(exports, "__esModule", { value: true });
 exports.Nack = void 0;
 const lodash_1 = require("lodash");
 const rx_mini_1 = __importDefault(require("rx.mini"));
-const src_1 = require("../../../common/src");
-const src_2 = require("../../../rtp/src");
+const src_1 = require("../../../../common/src");
+const src_2 = require("../../../../rtp/src");
 const LOST_SIZE = 30 * 5;
 class Nack {
     constructor(receiver) {

package/lib/webrtc/src/media/receiver/nack.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"nack.js","sourceRoot":"","sources":["../../../../../src/media/receiver/nack.ts"],"names":[],"mappings":";;;;;;AAAA,mCAA+B;AAC/B,sDAA4B;AAE5B,gDAAmD;AACnD,6CAI6B;AAG7B,MAAM,SAAS,GAAG,EAAE,GAAG,CAAC,CAAC;AAEzB,MAAa,IAAI;IAQf,YAAoB,QAAwB;QAAxB,aAAQ,GAAR,QAAQ,CAAgB;QAPpC,iBAAY,GAAG,CAAC,CAAC;QACjB,UAAK,GAAiC,EAAE,CAAC;QACzC,aAAQ,GAAG,WAAW,CAAC,GAAG,EAAE,CAAC,IAAI,CAAC,UAAU,EAAE,EAAE,EAAE,CAAC,CAAC;QAEnD,iBAAY,GAAG,IAAI,iBAAK,EAAiB,CAAC;IAGJ,CAAC;IAEhD,IAAI,IAAI;QACN,OAAO,MAAM,CAAC,IAAI,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC,GAAG,CAAC,MAAM,CAAC,CAAC;IAC7C,CAAC;IAED,SAAS,CAAC,MAAiB;QACzB,MAAM,EAAE,cAAc,EAAE,IAAI,EAAE,GAAG,MAAM,CAAC,MAAM,CAAC;QAC/C,IAAI,CAAC,eAAe,GAAG,IAAI,CAAC;QAE5B,IAAI,IAAI,CAAC,YAAY,KAAK,CAAC,EAAE;YAC3B,IAAI,CAAC,YAAY,GAAG,cAAc,CAAC;YACnC,OAAO;SACR;QAED,IAAI,IAAI,CAAC,KAAK,CAAC,cAAc,CAAC,EAAE;YAC9B,OAAO,IAAI,CAAC,KAAK,CAAC,cAAc,CAAC,CAAC;YAClC,OAAO;SACR;QAED,IAAI,cAAc,KAAK,eAAS,CAAC,IAAI,CAAC,YAAY,EAAE,CAAC,CAAC,EAAE;YACtD,IAAI,CAAC,YAAY,GAAG,cAAc,CAAC;SACpC;aAAM,IAAI,cAAc,GAAG,eAAS,CAAC,IAAI,CAAC,YAAY,EAAE,CAAC,CAAC,EAAE;YAC3D,uBAAuB;YACvB,cAAK,CAAC,eAAS,CAAC,IAAI,CAAC,YAAY,EAAE,CAAC,CAAC,EAAE,cAAc,CAAC,CAAC,OAAO,CAAC,CAAC,GAAG,EAAE,EAAE;gBACrE,IAAI,CAAC,KAAK,CAAC,GAAG,CAAC,GAAG,CAAC,CAAC;YACtB,CAAC,CAAC,CAAC;YACH,IAAI,CAAC,QAAQ,CAAC,WAAW,CAAC,IAAI,CAAC,eAAe,CAAC,CAAC;YAEhD,IAAI,CAAC,YAAY,GAAG,cAAc,CAAC;YAEnC,IAAI,MAAM,CAAC,IAAI,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC,MAAM,GAAG,SAAS,EAAE;gBAC9C,IAAI,CAAC,KAAK,GAAG,MAAM,CAAC,OAAO,CAAC,IAAI,CAAC,KAAK,CAAC;qBACpC,KAAK,CAAC,CAAC,SAAS,CAAC;qBACjB,MAAM,CAAC,CAAC,GAAG,EAAE,CAAC,GAAG,EAAE,CAAC,CAAC,EAAE,EAAE;oBACxB,GAAG,CAAC,GAAG,CAAC,GAAG,CAAC,CAAC;oBACb,OAAO,GAAG,CAAC;gBACb,CAAC,EAAE,EAAkC,CAAC,CAAC;aAC1C;SACF;IACH,CAAC;IAED,KAAK;QACH,aAAa,CAAC,IAAI,CAAC,QAAQ,CAAC,CAAC;IAC/B,CAAC;IAEO,SAAS;QACf,MAAM,CAAC,IAAI,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC,OAAO,CAAC,CAAC,GAAG,EAAE,EAAE;YACtC,IAAI,EAAE,IAAI,CAAC,KAAK,CAAC,GAAG,CAAC,GAAG,EAAE,EAAE;gBAC1B,OAAO,IAAI,CAAC,KAAK,CAAC,GAAG,CAAC,CAAC;aACxB;QACH,CAAC,CAAC,CAAC;IACL,CAAC;IAEO,UAAU;QAChB,IAAI,IAAI,CAAC,IAAI,CAAC,MAAM,GAAG,CAAC,IAAI,IAAI,CAAC,eAAe,EAAE;YAChD,MAAM,IAAI,GAAG,IAAI,iBAAW,CAAC;gBAC3B,UAAU,EAAE,IAAI,CAAC,QAAQ,CAAC,QAAQ;gBAClC,eAAe,EAAE,IAAI,CAAC,eAAe;gBACrC,IAAI,EAAE,IAAI,CAAC,IAAI;aAChB,CAAC,CAAC;YACH,MAAM,IAAI,GAAG,IAAI,gCAA0B,CAAC;gBAC1C,QAAQ,EAAE,IAAI;aACf,CAAC,CAAC;YACH,IAAI,CAAC,QAAQ,CAAC,aAAa,CAAC,QAAQ,CAAC,CAAC,IAAI,CAAC,CAAC,CAAC;YAE7C,IAAI,CAAC,SAAS,EAAE,CAAC;YACjB,IAAI,CAAC,YAAY,CAAC,OAAO,CAAC,IAAI,CAAC,CAAC;SACjC;IACH,CAAC;CACF;AA9ED,oBA8EC","sourcesContent":["import { range } from \"lodash\";\nimport Event from \"rx.mini\";\n\nimport { uint16Add } from \"../../../../common/src\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../../rtp/src\";\nimport { RTCRtpReceiver } from \"../rtpReceiver\";\n\nconst LOST_SIZE = 30 * 5;\n\nexport class Nack {\n  private newEstSeqNum = 0;\n  private _lost: { [seqNum: number]: number } = {};\n  private nackLoop = setInterval(() => this.packetLost(), 20);\n\n  readonly onPacketLost = new Event<[GenericNack]>();\n  mediaSourceSsrc?: number;\n\n  constructor(private receiver: RTCRtpReceiver) {}\n\n  get lost() {\n    return Object.keys(this._lost).map(Number);\n  }\n\n  addPacket(packet: RtpPacket) {\n    const { sequenceNumber, ssrc } = packet.header;\n    this.mediaSourceSsrc = ssrc;\n\n    if (this.newEstSeqNum === 0) {\n      this.newEstSeqNum = sequenceNumber;\n      return;\n    }\n\n    if (this._lost[sequenceNumber]) {\n      delete this._lost[sequenceNumber];\n      return;\n    }\n\n    if (sequenceNumber === uint16Add(this.newEstSeqNum, 1)) {\n      this.newEstSeqNum = sequenceNumber;\n    } else if (sequenceNumber > uint16Add(this.newEstSeqNum, 1)) {\n      // packet lost detected\n      range(uint16Add(this.newEstSeqNum, 1), sequenceNumber).forEach((seq) => {\n        this._lost[seq] = 1;\n      });\n      this.receiver.sendRtcpPLI(this.mediaSourceSsrc);\n\n      this.newEstSeqNum = sequenceNumber;\n\n      if (Object.keys(this._lost).length > LOST_SIZE) {\n        this._lost = Object.entries(this._lost)\n          .slice(-LOST_SIZE)\n          .reduce((acc, [key, v]) => {\n            acc[key] = v;\n            return acc;\n          }, {} as { [seqNum: number]: number });\n      }\n    }\n  }\n\n  close() {\n    clearInterval(this.nackLoop);\n  }\n\n  private increment() {\n    Object.keys(this._lost).forEach((seq) => {\n      if (++this._lost[seq] > 10) {\n        delete this._lost[seq];\n      }\n    });\n  }\n\n  private packetLost() {\n    if (this.lost.length > 0 && this.mediaSourceSsrc) {\n      const nack = new GenericNack({\n        senderSsrc: this.receiver.rtcpSsrc,\n        mediaSourceSsrc: this.mediaSourceSsrc,\n        lost: this.lost,\n      });\n      const rtcp = new RtcpTransportLayerFeedback({\n        feedback: nack,\n      });\n      this.receiver.dtlsTransport.sendRtcp([rtcp]);\n\n      this.increment();\n      this.onPacketLost.execute(nack);\n    }\n  }\n}\n"]}

package/lib/webrtc/src/media/{statistics.d.ts → receiver/statistics.d.ts}

@@ -1,4 +1,4 @@
-import { RtpPacket } from "../../../rtp/src";
+import { RtpPacket } from "../../../../rtp/src";
 export declare class StreamStatistics {
     base_seq?: number;
     max_seq?: number;

package/lib/webrtc/src/media/{statistics.js → receiver/statistics.js}

@@ -1,7 +1,7 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.StreamStatistics = void 0;
-const src_1 = require("../../../common/src");
+const src_1 = require("../../../../common/src");
 // from aiortc
 class StreamStatistics {
     constructor(clockRate) {

package/lib/webrtc/src/media/receiver/statistics.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"statistics.js","sourceRoot":"","sources":["../../../../../src/media/receiver/statistics.ts"],"names":[],"mappings":";;;AAAA,gDAAuD;AAGvD,cAAc;AAEd,MAAa,gBAAgB;IAgB3B,YAAY,SAAiB;QAb7B,WAAM,GAAG,CAAC,CAAC;QACX,qBAAgB,GAAG,CAAC,CAAC;QAIrB,cAAS,GAAG,CAAC,CAAC;QAId,kBAAkB;QAClB,mBAAc,GAAG,CAAC,CAAC;QACnB,mBAAc,GAAG,CAAC,CAAC;QAGjB,IAAI,CAAC,SAAS,GAAG,SAAS,CAAC;IAC7B,CAAC;IAED,GAAG,CAAC,MAAiB,EAAE,MAAc,IAAI,CAAC,GAAG,EAAE,GAAG,IAAI;QACpD,MAAM,OAAO,GACX,IAAI,CAAC,OAAO,IAAI,SAAS;YACzB,cAAQ,CAAC,MAAM,CAAC,MAAM,CAAC,cAAc,EAAE,IAAI,CAAC,OAAO,CAAC,CAAC;QACvD,IAAI,CAAC,gBAAgB,EAAE,CAAC;QAExB,IAAI,IAAI,CAAC,QAAQ,IAAI,SAAS,EAAE;YAC9B,IAAI,CAAC,QAAQ,GAAG,MAAM,CAAC,MAAM,CAAC,cAAc,CAAC;SAC9C;QAED,IAAI,OAAO,EAAE;YACX,MAAM,OAAO,GAAG,SAAG,CAAC,GAAG,GAAG,IAAI,CAAC,SAAS,CAAC,CAAC;YAE1C,IACE,IAAI,CAAC,OAAO,IAAI,SAAS;gBACzB,MAAM,CAAC,MAAM,CAAC,cAAc,GAAG,IAAI,CAAC,OAAO,EAC3C;gBACA,IAAI,CAAC,MAAM,IAAI,CAAC,IAAI,EAAE,CAAC;aACxB;YACD,IAAI,CAAC,OAAO,GAAG,MAAM,CAAC,MAAM,CAAC,cAAc,CAAC;YAE5C,IACE,MAAM,CAAC,MAAM,CAAC,SAAS,KAAK,IAAI,CAAC,cAAc;gBAC/C,IAAI,CAAC,gBAAgB,GAAG,CAAC,EACzB;gBACA,MAAM,IAAI,GAAG,IAAI,CAAC,GAAG,CACnB,OAAO;oBACL,CAAC,IAAI,CAAC,YAAY,IAAI,CAAC,CAAC;oBACxB,CAAC,MAAM,CAAC,MAAM,CAAC,SAAS,GAAG,CAAC,IAAI,CAAC,cAAc,IAAI,CAAC,CAAC,CAAC,CACzD,CAAC;gBACF,IAAI,CAAC,SAAS,IAAI,IAAI,GAAG,CAAC,CAAC,IAAI,CAAC,SAAS,GAAG,CAAC,CAAC,IAAI,CAAC,CAAC,CAAC;aACtD;YAED,IAAI,CAAC,YAAY,GAAG,OAAO,CAAC;YAC5B,IAAI,CAAC,cAAc,GAAG,MAAM,CAAC,MAAM,CAAC,SAAS,CAAC;SAC/C;IACH,CAAC;IAED,IAAI,aAAa;QACf,MAAM,iBAAiB,GAAG,IAAI,CAAC,gBAAgB,GAAG,IAAI,CAAC,cAAc,CAAC;QACtE,IAAI,CAAC,cAAc,GAAG,IAAI,CAAC,gBAAgB,CAAC;QAC5C,MAAM,iBAAiB,GAAG,IAAI,CAAC,gBAAgB,GAAG,IAAI,CAAC,cAAc,CAAC;QACtE,IAAI,CAAC,cAAc,GAAG,IAAI,CAAC,gBAAgB,CAAC;QAC5C,MAAM,aAAa,GAAG,iBAAiB,GAAG,iBAAiB,CAAC;QAC5D,IAAI,iBAAiB,IAAI,CAAC,IAAI,aAAa,IAAI,CAAC,EAAE;YAChD,OAAO,CAAC,CAAC;SACV;aAAM;YACL,OAAO,IAAI,CAAC,KAAK,CAAC,CAAC,aAAa,IAAI,CAAC,CAAC,GAAG,iBAAiB,CAAC,CAAC;SAC7D;IACH,CAAC;IAED,IAAI,MAAM;QACR,OAAO,IAAI,CAAC,SAAS,IAAI,CAAC,CAAC;IAC7B,CAAC;IAED,IAAI,gBAAgB;QAClB,OAAO,IAAI,CAAC,MAAM,GAAG,CAAC,IAAI,CAAC,OAAO,IAAI,CAAC,CAAC,GAAG,CAAC,IAAI,CAAC,QAAQ,IAAI,CAAC,CAAC,GAAG,CAAC,CAAC;IACtE,CAAC;IAED,IAAI,YAAY;QACd,MAAM,IAAI,GAAG,IAAI,CAAC,gBAAgB,GAAG,IAAI,CAAC,gBAAgB,CAAC;QAC3D,OAAO,IAAI,GAAG,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,IAAI,CAAC;IAC7B,CAAC;CACF;AAnFD,4CAmFC","sourcesContent":["import { int, uint16Gt } from \"../../../../common/src\";\nimport { RtpPacket } from \"../../../../rtp/src\";\n\n// from aiortc\n\nexport class StreamStatistics {\n  base_seq?: number;\n  max_seq?: number;\n  cycles = 0;\n  packets_received = 0;\n\n  // # jitter\n  private clockRate: number;\n  jitter_q4 = 0;\n  private last_arrival?: number;\n  private last_timestamp?: number;\n\n  // # fraction lost\n  expected_prior = 0;\n  received_prior = 0;\n\n  constructor(clockRate: number) {\n    this.clockRate = clockRate;\n  }\n\n  add(packet: RtpPacket, now: number = Date.now() / 1000) {\n    const inOrder =\n      this.max_seq == undefined ||\n      uint16Gt(packet.header.sequenceNumber, this.max_seq);\n    this.packets_received++;\n\n    if (this.base_seq == undefined) {\n      this.base_seq = packet.header.sequenceNumber;\n    }\n\n    if (inOrder) {\n      const arrival = int(now * this.clockRate);\n\n      if (\n        this.max_seq != undefined &&\n        packet.header.sequenceNumber < this.max_seq\n      ) {\n        this.cycles += 1 << 16;\n      }\n      this.max_seq = packet.header.sequenceNumber;\n\n      if (\n        packet.header.timestamp !== this.last_timestamp &&\n        this.packets_received > 1\n      ) {\n        const diff = Math.abs(\n          arrival -\n            (this.last_arrival ?? 0) -\n            (packet.header.timestamp - (this.last_timestamp ?? 0))\n        );\n        this.jitter_q4 += diff - ((this.jitter_q4 + 8) >> 4);\n      }\n\n      this.last_arrival = arrival;\n      this.last_timestamp = packet.header.timestamp;\n    }\n  }\n\n  get fraction_lost() {\n    const expected_interval = this.packets_expected - this.expected_prior;\n    this.expected_prior = this.packets_expected;\n    const received_interval = this.packets_received - this.received_prior;\n    this.received_prior = this.packets_received;\n    const lost_interval = expected_interval - received_interval;\n    if (expected_interval == 0 || lost_interval <= 0) {\n      return 0;\n    } else {\n      return Math.floor((lost_interval << 8) / expected_interval);\n    }\n  }\n\n  get jitter() {\n    return this.jitter_q4 >> 4;\n  }\n\n  get packets_expected() {\n    return this.cycles + (this.max_seq ?? 0) - (this.base_seq ?? 0) + 1;\n  }\n\n  get packets_lost() {\n    const lost = this.packets_expected - this.packets_received;\n    return lost < 0 ? 0 : lost;\n  }\n}\n"]}

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

@@ -7,7 +7,7 @@ exports.RtpRouter = void 0;
 const debug_1 = __importDefault(require("debug"));
 const src_1 = require("../../../common/src");
 const src_2 = require("../../../rtp/src");
-const rtpExtension_1 = require("../extension/rtpExtension");
+const rtpExtension_1 = require("./extension/rtpExtension");
 const rtpReceiver_1 = require("./rtpReceiver");
 const track_1 = require("./track");
 const log = debug_1.default("werift:packages/webrtc/src/media/router.ts");

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

@@ -1 +1 @@
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AC,CAAC,CAAC,CAAC,CAAC,CAAC;qBAC/C,CAAC;aACL;QACH,CAAC,CAAC;aACD,MAAM,CAAC,CAAC,GAA2B,EAAE,GAAG,EAAE,EAAE;YAC3C,IAAI,GAAG;gBAAE,GAAG,CAAC,GAAG,CAAC,GAAG,CAAC,GAAG,GAAG,CAAC,KAAK,CAAC;YAClC,OAAO,GAAG,CAAC;QACb,CAAC,EAAE,EAAE,CAAC,CAAC;IACX,CAAC;CAmGF;AAvLD,8BAuLC","sourcesContent":["import debug from \"debug\";\n\nimport { bufferReader } from \"../../../common/src\";\nimport {\n  Extension,\n  ReceiverEstimatedMaxBitrate,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { RTP_EXTENSION_URI } from \"../extension/rtpExtension\";\nimport {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport { RTCRtpSender } from \"./rtpSender\";\nimport { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport type Extensions = { [uri: string]: number | string };\n\nexport class RtpRouter {\n  private ssrcTable: { [ssrc: number]: RTCRtpReceiver | RTCRtpSender } = {};\n  private ridTable: { [rid: string]: RTCRtpReceiver | RTCRtpSender } = {};\n  private extIdUriMap: { [id: number]: string } = {};\n\n  constructor() {}\n\n  registerRtpSender(sender: RTCRtpSender) {\n    this.ssrcTable[sender.ssrc] = sender;\n  }\n\n  private registerRtpReceiver(receiver: RTCRtpReceiver, ssrc: number) {\n    this.ssrcTable[ssrc] = receiver;\n  }\n\n  registerRtpReceiverBySsrc(\n    transceiver: RTCRtpTransceiver,\n    params: RTCRtpReceiveParameters\n  ) {\n    log(\"registerRtpReceiverBySsrc\", params);\n\n    params.encodings\n      .filter((e) => e.ssrc != undefined) // todo fix\n      .forEach((encode) => {\n        this.registerRtpReceiver(transceiver.receiver, encode.ssrc);\n        transceiver.addTrack(\n          new MediaStreamTrack({\n            ssrc: encode.ssrc,\n            kind: transceiver.kind,\n            id: transceiver.sender.trackId,\n            remote: true,\n          })\n        );\n        if (encode.rtx) {\n          this.registerRtpReceiver(transceiver.receiver, encode.rtx.ssrc);\n        }\n      });\n\n    params.headerExtensions.forEach((extension) => {\n      this.extIdUriMap[extension.id] = extension.uri;\n    });\n  }\n\n  registerRtpReceiverByRid(\n    transceiver: RTCRtpTransceiver,\n    param: RTCRtpSimulcastParameters\n  ) {\n    log(\"registerRtpReceiverByRid\", param);\n    transceiver.addTrack(\n      new MediaStreamTrack({\n        rid: param.rid,\n        kind: transceiver.kind,\n        id: transceiver.sender.trackId,\n        remote: true,\n      })\n    );\n    this.ridTable[param.rid] = transceiver.receiver;\n  }\n\n  static rtpHeaderExtensionsParser(\n    extensions: Extension[],\n    extIdUriMap: { [id: number]: string }\n  ): Extensions {\n    return extensions\n      .map((extension) => {\n        const uri = extIdUriMap[extension.id];\n        switch (uri) {\n          case RTP_EXTENSION_URI.sdesMid:\n          case RTP_EXTENSION_URI.sdesRTPStreamID:\n          case RTP_EXTENSION_URI.repairedRtpStreamId:\n            return { uri, value: extension.payload.toString() };\n          case RTP_EXTENSION_URI.transportWideCC:\n            return { uri, value: extension.payload.readUInt16BE() };\n          case RTP_EXTENSION_URI.absSendTime:\n            return {\n              uri,\n              value: bufferReader(extension.payload, [3])[0],\n            };\n        }\n      })\n      .reduce((acc: { [uri: string]: any }, cur) => {\n        if (cur) acc[cur.uri] = cur.value;\n        return acc;\n      }, {});\n  }\n\n  routeRtp = (packet: RtpPacket) => {\n    const extensions = RtpRouter.rtpHeaderExtensionsParser(\n      packet.header.extensions,\n      this.extIdUriMap\n    );\n\n    let ssrcReceiver: RTCRtpReceiver | undefined = this.ssrcTable[\n      packet.header.ssrc\n    ] as RTCRtpReceiver;\n\n    const rid = extensions[RTP_EXTENSION_URI.sdesRTPStreamID];\n    if (typeof rid === \"string\") {\n      ssrcReceiver = this.ridTable[rid] as RTCRtpReceiver;\n      ssrcReceiver.latestRid = rid;\n      ssrcReceiver.handleRtpByRid(packet, rid, extensions);\n    } else if (ssrcReceiver) {\n      ssrcReceiver.handleRtpBySsrc(packet, extensions);\n    } else {\n      // simulcast after send receiver report\n      ssrcReceiver = Object.values(this.ridTable)\n        .filter((r): r is RTCRtpReceiver => r instanceof RTCRtpReceiver)\n        .find((r) => r.trackBySSRC[packet.header.ssrc]);\n      if (ssrcReceiver) {\n        log(\"simulcast register receiver by ssrc\", packet.header.ssrc);\n        this.registerRtpReceiver(ssrcReceiver, packet.header.ssrc);\n        ssrcReceiver.handleRtpBySsrc(packet, extensions);\n      }\n    }\n\n    if (!ssrcReceiver) {\n      log(\"ssrcReceiver not found\");\n      return;\n    }\n\n    const sdesMid = extensions[RTP_EXTENSION_URI.sdesMid];\n    if (typeof sdesMid === \"string\") {\n      ssrcReceiver.sdesMid = sdesMid;\n    }\n\n    const repairedRid = extensions[\n      RTP_EXTENSION_URI.repairedRtpStreamId\n    ] as string;\n    if (typeof repairedRid === \"string\") {\n      ssrcReceiver.latestRepairedRid = repairedRid;\n    }\n  };\n\n  routeRtcp = (packet: RtcpPacket) => {\n    const recipients: (RTCRtpReceiver | RTCRtpSender)[] = [];\n\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          packet = packet as RtcpSrPacket;\n          recipients.push(this.ssrcTable[packet.ssrc]);\n        }\n        break;\n      case RtcpRrPacket.type:\n        {\n          packet = packet as RtcpRrPacket;\n          packet.reports.forEach((report) => {\n            recipients.push(this.ssrcTable[report.ssrc]);\n          });\n        }\n        break;\n      case RtcpSourceDescriptionPacket.type:\n        {\n          const sdes = packet as RtcpSourceDescriptionPacket;\n          // log(\"sdes\", JSON.stringify(sdes.chunks));\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const rtpfb = packet as RtcpTransportLayerFeedback;\n          if (rtpfb.feedback) {\n            recipients.push(this.ssrcTable[rtpfb.feedback.mediaSourceSsrc]);\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const psfb = packet as RtcpPayloadSpecificFeedback;\n          switch (psfb.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;\n              recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);\n              break;\n            default:\n              recipients.push(this.ssrcTable[psfb.feedback.senderSsrc]);\n          }\n        }\n        break;\n    }\n    recipients\n      .filter((v) => v) // todo simulcast\n      .forEach((recipient) => recipient.handleRtcpPacket(packet));\n  };\n}\n"]}
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debug from \"debug\";\n\nimport { bufferReader } from \"../../../common/src\";\nimport {\n  Extension,\n  ReceiverEstimatedMaxBitrate,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport { RTCRtpSender } from \"./rtpSender\";\nimport { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport type Extensions = { [uri: string]: number | string };\n\nexport class RtpRouter {\n  private ssrcTable: { [ssrc: number]: RTCRtpReceiver | RTCRtpSender } = {};\n  private ridTable: { [rid: string]: RTCRtpReceiver | RTCRtpSender } = {};\n  private extIdUriMap: { [id: number]: string } = {};\n\n  constructor() {}\n\n  registerRtpSender(sender: RTCRtpSender) {\n    this.ssrcTable[sender.ssrc] = sender;\n  }\n\n  private registerRtpReceiver(receiver: RTCRtpReceiver, ssrc: number) {\n    this.ssrcTable[ssrc] = receiver;\n  }\n\n  registerRtpReceiverBySsrc(\n    transceiver: RTCRtpTransceiver,\n    params: RTCRtpReceiveParameters\n  ) {\n    log(\"registerRtpReceiverBySsrc\", params);\n\n    params.encodings\n      .filter((e) => e.ssrc != undefined) // todo fix\n      .forEach((encode) => {\n        this.registerRtpReceiver(transceiver.receiver, encode.ssrc);\n        transceiver.addTrack(\n          new MediaStreamTrack({\n            ssrc: encode.ssrc,\n            kind: transceiver.kind,\n            id: transceiver.sender.trackId,\n            remote: true,\n          })\n        );\n        if (encode.rtx) {\n          this.registerRtpReceiver(transceiver.receiver, encode.rtx.ssrc);\n        }\n      });\n\n    params.headerExtensions.forEach((extension) => {\n      this.extIdUriMap[extension.id] = extension.uri;\n    });\n  }\n\n  registerRtpReceiverByRid(\n    transceiver: RTCRtpTransceiver,\n    param: RTCRtpSimulcastParameters\n  ) {\n    log(\"registerRtpReceiverByRid\", param);\n    transceiver.addTrack(\n      new MediaStreamTrack({\n        rid: param.rid,\n        kind: transceiver.kind,\n        id: transceiver.sender.trackId,\n        remote: true,\n      })\n    );\n    this.ridTable[param.rid] = transceiver.receiver;\n  }\n\n  static rtpHeaderExtensionsParser(\n    extensions: Extension[],\n    extIdUriMap: { [id: number]: string }\n  ): Extensions {\n    return extensions\n      .map((extension) => {\n        const uri = extIdUriMap[extension.id];\n        switch (uri) {\n          case RTP_EXTENSION_URI.sdesMid:\n          case RTP_EXTENSION_URI.sdesRTPStreamID:\n          case RTP_EXTENSION_URI.repairedRtpStreamId:\n            return { uri, value: extension.payload.toString() };\n          case RTP_EXTENSION_URI.transportWideCC:\n            return { uri, value: extension.payload.readUInt16BE() };\n          case RTP_EXTENSION_URI.absSendTime:\n            return {\n              uri,\n              value: bufferReader(extension.payload, [3])[0],\n            };\n        }\n      })\n      .reduce((acc: { [uri: string]: any }, cur) => {\n        if (cur) acc[cur.uri] = cur.value;\n        return acc;\n      }, {});\n  }\n\n  routeRtp = (packet: RtpPacket) => {\n    const extensions = RtpRouter.rtpHeaderExtensionsParser(\n      packet.header.extensions,\n      this.extIdUriMap\n    );\n\n    let ssrcReceiver: RTCRtpReceiver | undefined = this.ssrcTable[\n      packet.header.ssrc\n    ] as RTCRtpReceiver;\n\n    const rid = extensions[RTP_EXTENSION_URI.sdesRTPStreamID];\n    if (typeof rid === \"string\") {\n      ssrcReceiver = this.ridTable[rid] as RTCRtpReceiver;\n      ssrcReceiver.latestRid = rid;\n      ssrcReceiver.handleRtpByRid(packet, rid, extensions);\n    } else if (ssrcReceiver) {\n      ssrcReceiver.handleRtpBySsrc(packet, extensions);\n    } else {\n      // simulcast after send receiver report\n      ssrcReceiver = Object.values(this.ridTable)\n        .filter((r): r is RTCRtpReceiver => r instanceof RTCRtpReceiver)\n        .find((r) => r.trackBySSRC[packet.header.ssrc]);\n      if (ssrcReceiver) {\n        log(\"simulcast register receiver by ssrc\", packet.header.ssrc);\n        this.registerRtpReceiver(ssrcReceiver, packet.header.ssrc);\n        ssrcReceiver.handleRtpBySsrc(packet, extensions);\n      }\n    }\n\n    if (!ssrcReceiver) {\n      log(\"ssrcReceiver not found\");\n      return;\n    }\n\n    const sdesMid = extensions[RTP_EXTENSION_URI.sdesMid];\n    if (typeof sdesMid === \"string\") {\n      ssrcReceiver.sdesMid = sdesMid;\n    }\n\n    const repairedRid = extensions[\n      RTP_EXTENSION_URI.repairedRtpStreamId\n    ] as string;\n    if (typeof repairedRid === \"string\") {\n      ssrcReceiver.latestRepairedRid = repairedRid;\n    }\n  };\n\n  routeRtcp = (packet: RtcpPacket) => {\n    const recipients: (RTCRtpReceiver | RTCRtpSender)[] = [];\n\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          packet = packet as RtcpSrPacket;\n          recipients.push(this.ssrcTable[packet.ssrc]);\n        }\n        break;\n      case RtcpRrPacket.type:\n        {\n          packet = packet as RtcpRrPacket;\n          packet.reports.forEach((report) => {\n            recipients.push(this.ssrcTable[report.ssrc]);\n          });\n        }\n        break;\n      case RtcpSourceDescriptionPacket.type:\n        {\n          const sdes = packet as RtcpSourceDescriptionPacket;\n          // log(\"sdes\", JSON.stringify(sdes.chunks));\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const rtpfb = packet as RtcpTransportLayerFeedback;\n          if (rtpfb.feedback) {\n            recipients.push(this.ssrcTable[rtpfb.feedback.mediaSourceSsrc]);\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const psfb = packet as RtcpPayloadSpecificFeedback;\n          switch (psfb.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;\n              recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);\n              break;\n            default:\n              recipients.push(this.ssrcTable[psfb.feedback.senderSsrc]);\n          }\n        }\n        break;\n    }\n    recipients\n      .filter((v) => v) // todo simulcast\n      .forEach((recipient) => recipient.handleRtcpPacket(packet));\n  };\n}\n"]}

package/lib/webrtc/src/media/rtpReceiver.d.ts

@@ -1,3 +1,4 @@
+import Event from "rx.mini";
 import { RtcpPacket, RtpPacket } from "../../../rtp/src";
 import { RTCDtlsTransport } from "../transport/dtls";
 import { Kind } from "../types/domain";
@@ -22,12 +23,13 @@ export declare class RTCRtpReceiver {
         [rid: string]: MediaStreamTrack;
     };
     readonly lsr: {
-        [ssrc: number]: BigInt;
+        [ssrc: number]: number;
     };
     readonly lsrTime: {
         [ssrc: number]: number;
     };
-    readonly onPacketLost: import("rx.mini").default<[import("../../../rtp/src").GenericNack]>;
+    readonly onPacketLost: Event<[import("../../../rtp/src").GenericNack]>;
+    readonly onRtcp: Event<[RtcpPacket]>;
     sdesMid?: string;
     latestRid?: string;
     latestRepairedRid?: string;

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

@@ -1,16 +1,21 @@
 "use strict";
+var __importDefault = (this && this.__importDefault) || function (mod) {
+    return (mod && mod.__esModule) ? mod : { "default": mod };
+};
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.unwrapRtx = exports.RTCRtpReceiver = void 0;
 const debug_1 = require("debug");
 const jspack_1 = require("jspack");
+const rx_mini_1 = __importDefault(require("rx.mini"));
 const promises_1 = require("timers/promises");
 const uuid_1 = require("uuid");
 const src_1 = require("../../../common/src");
 const src_2 = require("../../../rtp/src");
-const rtpExtension_1 = require("../extension/rtpExtension");
-const nack_1 = require("./nack");
+const utils_1 = require("../utils");
+const rtpExtension_1 = require("./extension/rtpExtension");
+const nack_1 = require("./receiver/nack");
 const receiverTwcc_1 = require("./receiver/receiverTwcc");
-const statistics_1 = require("./statistics");
+const statistics_1 = require("./receiver/statistics");
 const log = debug_1.debug("werift:packages/webrtc/src/media/rtpReceiver.ts");
 class RTCRtpReceiver {
     constructor(kind, dtlsTransport, rtcpSsrc) {
@@ -29,6 +34,7 @@ class RTCRtpReceiver {
         this.lsr = {};
         this.lsrTime = {};
         this.onPacketLost = this.nack.onPacketLost;
+        this.onRtcp = new rx_mini_1.default();
         this.supportTWCC = false;
         this.stopped = false;
         this.rtcpRunning = false;
@@ -104,7 +110,7 @@ class RTCRtpReceiver {
                     signal: this.rtcpCancel.signal,
                 });
                 const reports = Object.entries(this.remoteStreams).map(([ssrc, stream]) => {
-                    let lsr = 0n, dlsr = 0;
+                    let lsr = 0, dlsr = 0;
                     if (this.lsr[ssrc]) {
                         lsr = this.lsr[ssrc];
                         const delay = Date.now() / 1000 - this.lsrTime[ssrc];
@@ -118,7 +124,7 @@ class RTCRtpReceiver {
                         packetsLost: stream.packets_lost,
                         highestSequence: stream.max_seq,
                         jitter: stream.jitter,
-                        lsr: Number(lsr),
+                        lsr,
                         dlsr,
                     });
                 });
@@ -155,11 +161,12 @@ class RTCRtpReceiver {
             case src_2.RtcpSrPacket.type:
                 {
                     const sr = packet;
-                    this.lsr[sr.ssrc] = (sr.senderInfo.ntpTimestamp >> 16n) & 0xffffffffn;
+                    this.lsr[sr.ssrc] = utils_1.compactNtp(sr.senderInfo.ntpTimestamp);
                     this.lsrTime[sr.ssrc] = Date.now() / 1000;
                 }
                 break;
         }
+        this.onRtcp.execute(packet);
     }
     handleRTP(packet, extensions, track) {
         if (this.stopped)

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

@@ -1 +1 @@
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{ debug } from \"debug\";\nimport { jspack } from \"jspack\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { RTP_EXTENSION_URI } from \"../extension/rtpExtension\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { Nack } from \"./nack\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { Extensions } from \"./router\";\nimport { StreamStatistics } from \"./statistics\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new Nack(this);\n\n  readonly type = \"receiver\";\n  readonly uuid = uuid();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  // last senderReport\n  readonly lsr: { [ssrc: number]: BigInt } = {};\n  readonly lsrTime: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  supportTWCC = false;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    public kind: Kind,\n    public dtlsTransport: RTCDtlsTransport,\n    public rtcpSsrc: number\n  ) {}\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc?: number) {\n    this.supportTWCC = !!Object.values(this.codecs).find((codec) =>\n      codec.rtcpFeedback.find((v) => v.type === \"transport-cc\")\n    );\n    log(\"twcc support\", this.supportTWCC);\n\n    if (this.supportTWCC && mediaSourceSsrc) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\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 reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lsr = 0n,\n              dlsr = 0;\n            if (this.lsr[ssrc]) {\n              lsr = this.lsr[ssrc];\n              const delay = Date.now() / 1000 - this.lsrTime[ssrc];\n              if (delay > 0 && delay < 65536) {\n                dlsr = int(delay * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: Number(lsr),\n              dlsr,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lsr[sr.ssrc] = (sr.senderInfo.ntpTimestamp >> 16n) & 0xffffffffn;\n          this.lsrTime[sr.ssrc] = Date.now() / 1000;\n        }\n        break;\n    }\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack\n  ) {\n    if (this.stopped) return;\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as number;\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.supportTWCC) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const rtxCodec = this.codecs[codec.parameters[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    // todo fix select use or not use nack\n    if (track?.kind === \"video\") this.nack.addPacket(packet);\n\n    if (track) track.onReceiveRtp.execute(packet.clone());\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.slice(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.slice(2)\n  );\n  return packet;\n}\n"]}
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{ debug } from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { Nack } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\nimport { Extensions } from \"./router\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new Nack(this);\n\n  readonly type = \"receiver\";\n  readonly uuid = uuid();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  // last senderReport\n  readonly lsr: { [ssrc: number]: number } = {};\n  readonly lsrTime: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  supportTWCC = false;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    public kind: Kind,\n    public dtlsTransport: RTCDtlsTransport,\n    public rtcpSsrc: number\n  ) {}\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc?: number) {\n    this.supportTWCC = !!Object.values(this.codecs).find((codec) =>\n      codec.rtcpFeedback.find((v) => v.type === \"transport-cc\")\n    );\n    log(\"twcc support\", this.supportTWCC);\n\n    if (this.supportTWCC && mediaSourceSsrc) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\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 reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lsr = 0,\n              dlsr = 0;\n            if (this.lsr[ssrc]) {\n              lsr = this.lsr[ssrc];\n              const delay = Date.now() / 1000 - this.lsrTime[ssrc];\n              if (delay > 0 && delay < 65536) {\n                dlsr = int(delay * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr,\n              dlsr,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lsr[sr.ssrc] = compactNtp(sr.senderInfo.ntpTimestamp);\n          this.lsrTime[sr.ssrc] = Date.now() / 1000;\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack\n  ) {\n    if (this.stopped) return;\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as number;\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.supportTWCC) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const rtxCodec = this.codecs[codec.parameters[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    // todo fix select use or not use nack\n    if (track?.kind === \"video\") this.nack.addPacket(packet);\n\n    if (track) track.onReceiveRtp.execute(packet.clone());\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.slice(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.slice(2)\n  );\n  return packet;\n}\n"]}

package/lib/webrtc/src/media/rtpSender.d.ts

@@ -1,10 +1,10 @@
 /// <reference types="node" />
 import Event from "rx.mini";
-import { GenericNack, RtcpPacket, RtpPacket } from "../../../rtp/src";
+import { GenericNack, RtcpPacket, RtpHeader, RtpPacket } from "../../../rtp/src";
 import { RTCDtlsTransport } from "../transport/dtls";
 import { Kind } from "../types/domain";
 import { RTCRtpSendParameters } from "./parameters";
-import { SenderBandwidthEstimator } from "./senderBWE/senderBWE";
+import { SenderBandwidthEstimator } from "./sender/senderBWE";
 import { MediaStreamTrack } from "./track";
 export declare class RTCRtpSender {
     trackOrKind: Kind | MediaStreamTrack;
@@ -13,7 +13,7 @@ export declare class RTCRtpSender {
     readonly kind: Kind;
     readonly ssrc: number;
     readonly rtxSsrc: number;
-    readonly streamId: string;
+    streamId: string;
     readonly trackId: string;
     readonly onReady: Event<any[]>;
     readonly onRtcp: Event<[RtcpPacket]>;
@@ -52,7 +52,7 @@ export declare class RTCRtpSender {
     replaceTrack(track: MediaStreamTrack | null): Promise<void>;
     stop(): void;
     runRtcp(): Promise<void>;
-    private replaceRTP;
+    replaceRTP({ sequenceNumber, timestamp, }: Pick<RtpHeader, "sequenceNumber" | "timestamp">): void;
     sendRtp(rtp: Buffer | RtpPacket): void;
     handleRtcpPacket(rtcpPacket: RtcpPacket): void;
 }

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

@@ -31,9 +31,9 @@ const promises_1 = require("timers/promises");
 const uuid = __importStar(require("uuid"));
 const src_1 = require("../../../common/src");
 const src_2 = require("../../../rtp/src");
-const rtpExtension_1 = require("../extension/rtpExtension");
 const utils_1 = require("../utils");
-const senderBWE_1 = require("./senderBWE/senderBWE");
+const rtpExtension_1 = require("./extension/rtpExtension");
+const senderBWE_1 = require("./sender/senderBWE");
 const track_1 = require("./track");
 const log = debug_1.default("werift:packages/webrtc/src/media/rtpSender.ts");
 const RTP_HISTORY_SIZE = 128;
@@ -75,6 +75,9 @@ class RTCRtpSender {
             }
         });
         if (trackOrKind instanceof track_1.MediaStreamTrack) {
+            if (trackOrKind.streamId) {
+                this.streamId = trackOrKind.streamId;
+            }
             this.registerTrack(trackOrKind);
         }
     }
@@ -111,6 +114,9 @@ class RTCRtpSender {
         if (this.codec) {
             track.codec = this.codec;
         }
+        track.onSourceChanged.subscribe((header) => {
+            this.replaceRTP(header);
+        });
     }
     async replaceTrack(track) {
         if (track === null) {
@@ -124,7 +130,7 @@ class RTCRtpSender {
             this.replaceRTP(header);
         }
         this.registerTrack(track);
-        log("replaceTrack", track.ssrc, track.rid);
+        log("replaceTrack", "ssrc", track.ssrc, "rid", track.rid);
     }
     stop() {
         this.stopped = true;
@@ -152,7 +158,7 @@ class RTCRtpSender {
                         }),
                     }),
                 ];
-                this.lsr = (this.ntpTimestamp >> 16n) & 0xffffffffn;
+                this.lsr = utils_1.compactNtp(this.ntpTimestamp);
                 this.lsrTime = Date.now() / 1000;
                 if (this.cname) {
                     packets.push(new src_2.RtcpSourceDescriptionPacket({
@@ -177,7 +183,7 @@ class RTCRtpSender {
         }
         catch (error) { }
     }
-    replaceRTP({ sequenceNumber, timestamp }) {
+    replaceRTP({ sequenceNumber, timestamp, }) {
         if (this.sequenceNumber != undefined) {
             this.seqOffset = src_1.uint16Add(this.sequenceNumber, -sequenceNumber);
         }
@@ -260,7 +266,7 @@ class RTCRtpSender {
                     packet.reports
                         .filter((report) => report.ssrc === this.ssrc)
                         .forEach((report) => {
-                        if (this.lsr === BigInt(report.lsr) && report.dlsr) {
+                        if (this.lsr === report.lsr && report.dlsr) {
                             const rtt = Date.now() / 1000 - this.lsrTime - report.dlsr / 65536;
                             if (this.rtt === undefined) {
                                 this.rtt = rtt;

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

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{ 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  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 { RTP_EXTENSION_URI } from \"../extension/rtpExtension\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { milliTime, ntpTime } from \"../utils\";\nimport {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n  RTCRtpSendParameters,\n} from \"./parameters\";\nimport { SenderBandwidthEstimator, SentInfo } from \"./senderBWE/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  readonly 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  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lsr?: bigint;\n  private lsrTime: number = Date.now() / 1000;\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  private codec?: RTCRtpCodecParameters;\n\n  track?: MediaStreamTrack;\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(\n    public trackOrKind: Kind | MediaStreamTrack,\n    public dtlsTransport: RTCDtlsTransport\n  ) {\n    dtlsTransport.onStateChange.subscribe((state) => {\n      if (state === \"connected\") {\n        this.onReady.execute();\n      }\n    });\n    if (trackOrKind instanceof MediaStreamTrack) {\n      this.registerTrack(trackOrKind);\n    }\n  }\n\n  prepareSend(params: RTCRtpSendParameters) {\n    this.cname = params.rtcp?.cname;\n    this.mid = params.muxId;\n    this.headerExtensions = params.headerExtensions;\n    this.rtpStreamId = params.rtpStreamId;\n    this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n    for (const codec of params.codecs) {\n      if (\n        codec.name.toLowerCase() === \"rtx\" &&\n        codec.parameters[\"apt\"] === params.codecs[0].payloadType\n      ) {\n        this.rtxPayloadType = codec.payloadType;\n        break;\n      }\n    }\n\n    this.codec = params.codecs[0];\n    if (this.track) {\n      this.track.codec = this.codec;\n    }\n  }\n\n  registerTrack(track: MediaStreamTrack) {\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n\n    track.id = this.trackId;\n\n    const { unSubscribe } = track.onReceiveRtp.subscribe((rtp) => {\n      this.sendRtp(rtp);\n    });\n    this.track = track;\n    this.disposeTrack = unSubscribe;\n\n    if (this.codec) {\n      track.codec = this.codec;\n    }\n  }\n\n  async replaceTrack(track: MediaStreamTrack | null) {\n    if (track === null) {\n      // todo impl\n      return;\n    }\n\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.sequenceNumber != undefined) {\n      const header =\n        track.header || (await track.onReceiveRtp.asPromise())[0].header;\n\n      this.replaceRTP(header);\n    }\n\n    this.registerTrack(track);\n    log(\"replaceTrack\", track.ssrc, track.rid);\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    this.track = undefined;\n  }\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.lsr = (this.ntpTimestamp >> 16n) & 0xffffffffn;\n        this.lsrTime = Date.now() / 1000;\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  private replaceRTP({ sequenceNumber, timestamp }: RtpHeader) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = Number(\n        uint32Add(BigInt(this.timestamp), BigInt(-timestamp))\n      );\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  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 = Number(\n      uint32Add(BigInt(header.timestamp), BigInt(this.timestampOffset))\n    );\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 = Number(uint32Add(BigInt(this.packetCount), 1n));\n\n    rtp.header = header;\n\n    this.rtpCache.push(rtp);\n    this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);\n\n    const size = this.dtlsTransport.sendRtp(rtp.payload, 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.lsr === BigInt(report.lsr) && report.dlsr) {\n                const rtt =\n                  Date.now() / 1000 - this.lsrTime - 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        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((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                    this.dtlsTransport.sendRtp(packet.payload, packet.header);\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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{ 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  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 { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime } 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  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lsr?: number;\n  private lsrTime: number = Date.now() / 1000;\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  private codec?: RTCRtpCodecParameters;\n\n  track?: MediaStreamTrack;\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(\n    public trackOrKind: Kind | MediaStreamTrack,\n    public dtlsTransport: RTCDtlsTransport\n  ) {\n    dtlsTransport.onStateChange.subscribe((state) => {\n      if (state === \"connected\") {\n        this.onReady.execute();\n      }\n    });\n    if (trackOrKind instanceof MediaStreamTrack) {\n      if (trackOrKind.streamId) {\n        this.streamId = trackOrKind.streamId;\n      }\n      this.registerTrack(trackOrKind);\n    }\n  }\n\n  prepareSend(params: RTCRtpSendParameters) {\n    this.cname = params.rtcp?.cname;\n    this.mid = params.muxId;\n    this.headerExtensions = params.headerExtensions;\n    this.rtpStreamId = params.rtpStreamId;\n    this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n    for (const codec of params.codecs) {\n      if (\n        codec.name.toLowerCase() === \"rtx\" &&\n        codec.parameters[\"apt\"] === params.codecs[0].payloadType\n      ) {\n        this.rtxPayloadType = codec.payloadType;\n        break;\n      }\n    }\n\n    this.codec = params.codecs[0];\n    if (this.track) {\n      this.track.codec = this.codec;\n    }\n  }\n\n  registerTrack(track: MediaStreamTrack) {\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n\n    track.id = this.trackId;\n\n    const { unSubscribe } = track.onReceiveRtp.subscribe((rtp) => {\n      this.sendRtp(rtp);\n    });\n    this.track = track;\n    this.disposeTrack = unSubscribe;\n\n    if (this.codec) {\n      track.codec = this.codec;\n    }\n\n    track.onSourceChanged.subscribe((header) => {\n      this.replaceRTP(header);\n    });\n  }\n\n  async replaceTrack(track: MediaStreamTrack | null) {\n    if (track === null) {\n      // todo impl\n      return;\n    }\n\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.sequenceNumber != undefined) {\n      const header =\n        track.header || (await track.onReceiveRtp.asPromise())[0].header;\n\n      this.replaceRTP(header);\n    }\n\n    this.registerTrack(track);\n    log(\"replaceTrack\", \"ssrc\", track.ssrc, \"rid\", track.rid);\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    this.track = undefined;\n  }\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.lsr = compactNtp(this.ntpTimestamp);\n        this.lsrTime = Date.now() / 1000;\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    sequenceNumber,\n    timestamp,\n  }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = Number(\n        uint32Add(BigInt(this.timestamp), BigInt(-timestamp))\n      );\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  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 = Number(\n      uint32Add(BigInt(header.timestamp), BigInt(this.timestampOffset))\n    );\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 = Number(uint32Add(BigInt(this.packetCount), 1n));\n\n    rtp.header = header;\n\n    this.rtpCache.push(rtp);\n    this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);\n\n    const size = this.dtlsTransport.sendRtp(rtp.payload, 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.lsr === report.lsr && report.dlsr) {\n                const rtt =\n                  Date.now() / 1000 - this.lsrTime - 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        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((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                    this.dtlsTransport.sendRtp(packet.payload, packet.header);\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/media/sender/cumulativeResult.js.map

@@ -0,0 +1 @@
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