@ale-rainbow/web-sdk 5.1.0

package/lib/services/webrtcGateway/webrtcGateway.service.d.ts

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+import { Subject } from "rxjs";
+import { CallRB } from "../../models/call.model";
+import { Service } from "../../services/service";
+import type { TelephonyService } from "../../services/telephony/telephony.service";
+import { Contact } from "../../models/contact.model";
+export declare const WEBRTC_GATEWAY_SVC = "WebrtcGatewayService";
+export declare enum MPCallState {
+    Unknow = "Unknown",
+    Free = "Free",
+    WaitWebrtc = "WaitWebrtc",
+    WaitTelephony = "WaitTelephony",
+    CallOffer = "CallOffer",
+    CallOnGoing = "CallOnGoing",
+    CallActive = "CallActive",
+    CallActiveNoWebMedia = "CallActiveNoWebMedia",
+    TelCallReleasing = "TelCallReleasing",
+    WebCallReleasing = "WebCallReleasing",
+    RemoteControled = "RemoteControled",
+    AnomalyCCS = "AnomalyCCS"
+}
+interface MediaPillarStates {
+    mediaPillarConfigured: boolean;
+    mediaPillarAlive: boolean;
+}
+interface MediaPillarConnection {
+    keepAlive_TO: any;
+    mediaPillarJidArray: string[];
+    rainbowPhoneNumber: string;
+    remoteExtension: string;
+    mediaPillarStatesPool: Partial<MediaPillarStates>;
+}
+interface MediaPillarCallContext {
+    id: number;
+    callState: MPCallState;
+    callState_TO: any;
+    waitTelephony_TO: any;
+    isOutgoingCall3PCC: boolean;
+    telephonyCallRefs: CallRB[];
+    webrtcCallRef: any;
+    previousCallConnectionId: string;
+    previousCallStatus: string;
+    updateContactFlag: boolean;
+    getMediaPillarJid: any;
+    onGoingCallMediaPillarJid: string;
+    callOwnerForOxoMultiClientFlag: boolean;
+}
+export declare class WebrtcGatewayService extends Service {
+    private portalURL;
+    started: boolean;
+    private listeners;
+    private myContact;
+    private mediaPillarContact;
+    private mediaPillarConfigured;
+    private mediaPillarAlive;
+    private evtQueue;
+    private SEM;
+    private mediaPillarPollingSelected;
+    private TO_transition;
+    private TO_transition_long;
+    private TO_NoWebMedia;
+    private TO_NoWebMediaNotel;
+    private TO_PILLAR_POLLING;
+    private TO_waitTelephony;
+    private mediaPillarKeepAliveSuspend;
+    private keepAliveIterationCpt;
+    mediaPillarConnection: MediaPillarConnection;
+    mediaPillarCallContext: MediaPillarCallContext;
+    mediaPillarCallContexts: MediaPillarCallContext[];
+    mediaPillarCallContextId: number;
+    private isMixPstnWebRtcAllowed;
+    private systemId;
+    private displayName;
+    private oxeBusyCapability;
+    private systDeviceName;
+    private startRetryTimer;
+    private xmppConnectionSubscription;
+    private subscriptions;
+    rxSubject: Subject<any>;
+    private mainService;
+    private xmppService;
+    private logger;
+    private authService;
+    private profileService;
+    private errorHelperService;
+    private contactService;
+    private eventService;
+    private webrtcConfigAndUtils;
+    private settingsService;
+    private conversationService;
+    private userService;
+    private webrtcP2PService;
+    telephonyService: TelephonyService;
+    private supervisionService;
+    static getInstance(): WebrtcGatewayService;
+    static build(): WebrtcGatewayService;
+    protected constructor();
+    /**
+     * Start service method
+     * @public
+     * @memberof WebrtcGatewayService
+     */
+    start(): Promise<void>;
+    private telConfigChanged;
+    stop(): void;
+    private setMediaPillarConfigured;
+    private onCallEventPreProcessing;
+    /**
+     * Method onCallEvent
+     * @private
+     * @memberof WebrtcGatewayService
+     * Manage call events (but only if media pillar situation)
+     * @param __event
+     * @param call
+     * @param infoEvt
+     */
+    private onCallEvent;
+    /**
+     * Method processEvt
+     * @private
+     * @param callEvent
+     * @memberof WebrtcGatewayService
+     * Event processing
+     */
+    private processEvt;
+    getOrCreateMediaPIllarCallContext(call: CallRB): MediaPillarCallContext | null;
+    getOrCreateContextForTelCall(call: CallRB): MediaPillarCallContext | null;
+    getOrCreateContextForWebrtcCallIncoming(call: CallRB): MediaPillarCallContext;
+    getContextForWebrtcCallExisting(call: CallRB): MediaPillarCallContext | undefined;
+    createMediaPIllarCallContext(): MediaPillarCallContext;
+    /**
+     * Method releaseMediaPillarCallContext
+     * @public
+     * @memberof WebrtcGatewayService
+     * Allow to free the media pillar call context (all field except mediaPillarJid)
+     * but do not check the associated calls (tel and webrtc)
+     * @param mediaPillarCallContext
+     */
+    releaseMediaPillarCallContext(mediaPillarCallContext?: MediaPillarCallContext): void;
+    /**
+     * Method resetEvtAutomaton
+     * @public
+     * @memberof WebrtcGatewayService
+     * Allow to reset the eventing automaton
+     * linked to onCallEvent
+     */
+    resetEvtAutomaton(): void;
+    /**
+     * Method stopCallStateTimeOut
+     * @private
+     * @memberof WebrtcGatewayService
+     * @param mediaPillarCallContext
+     */
+    private stopCallStateTimeOut;
+    /**
+     * Method stopWaitTelephonyTimeOut
+     * @private
+     * @memberof WebrtcGatewayService
+     * @param mediaPillarCallContext
+     */
+    private stopWaitTelephonyTimeOut;
+    /**
+     * Method onRemoteCtrlCall
+     * @private
+     * @memberof WebrtcGatewayService
+     * Traitment of multi ressources case when other ressource manage the call
+     * @param __event
+     * @param call
+     */
+    private onRemoteCtrlCall;
+    /**
+     * Method onRemoteReservedCall
+     * @private
+     * @memberof WebrtcGatewayService
+     * Traitment of multi ressources case when other ressource reserve the call
+     * @param __event
+     * @param call
+     */
+    private onRemoteReservedCall;
+    /**
+     * Method addTelCallRefs
+     * @private
+     * @memberof WebrtcGatewayService
+     * add/push telcall ref in telephonyCallRefs if not already in
+     * @param callRef
+     */
+    private addTelCallRefs;
+    /**
+     * Method removeTelCallRefs
+     * @private
+     * @memberof WebrtcGatewayService
+     * remove telcall ref in telephonyCallRefs
+     * @param callRef
+     */
+    private removeTelCallRefs;
+    /**
+     * Method putAsMasterTelCallRefs
+     * @private
+     * @memberof WebrtcGatewayService
+     * The telephonyCallRefs[0] is intended to be the main telcall ref
+     * usually the one which was handled as 1st to mediaPillaryse the call
+     * @param callRef
+     */
+    private putAsMasterTelCallRefs;
+    /**
+     * Method isMasterTelCallRefs
+     * @public
+     * @memberof WebrtcGatewayService
+     * return true if callRef is the master telcallref
+     * @param callRef
+     */
+    isMasterTelCallRefs(callRef: CallRB | undefined): boolean;
+    /**
+     * Method isInTelCallRefs
+     * @private
+     * @memberof WebrtcGatewayService
+     * @returns true if callRef is in telephonyCallRefs
+     * @param callRef
+     */
+    private isInTelCallRefs;
+    /**
+     * Method replaceCallRefs
+     * @private
+     * @memberof WebrtcGatewayService
+     * replace one callref with another
+     * @param newCallRef
+     * @param oldCallRef
+     */
+    replaceCallRefs(newCallRef: CallRB | undefined, oldCallRef: CallRB | undefined): void;
+    /**
+     * Method automatonDefenseTimout
+     * @private
+     * @memberof WebrtcGatewayService
+     * @param call
+     * @param mediaPillarCallContext
+     */
+    private automatonDefenseTimout;
+    /** ======================================================================================== */
+    /** ==================================== TELEPHONY EVENT =================================== */
+    /** ======================================================================================== */
+    /**
+     * Method onTelCallIncomming
+     * @private
+     * @param call - the call
+     * @memberof WebrtcGatewayService
+     */
+    private onTelCallIncomming;
+    /**
+     * Method onTelCallDialling
+     * @private
+     * @param call - the tel call
+     * @memberof WebrtcGatewayService
+     */
+    private onTelCallDialling;
+    /**
+     * Method onTelCallOutgoing
+     * @private
+     * @param call - the tel call
+     * @memberof WebrtcGatewayService
+     */
+    private onTelCallOutgoing;
+    /**
+     * Method onTelCallActive
+     * @private
+     * @param call - the tel call
+     * @memberof WebrtcGatewayService
+     */
+    private onTelCallActive;
+    /**
+     * Method onTelCallReleasing
+     * @private
+     * @param call - the tel call
+     * @memberof WebrtcGatewayService
+     */
+    private onTelCallReleasing;
+    /** ======================================================================================== */
+    /** ====================================  WEBRTC EVENT  ==================================== */
+    /** ======================================================================================== */
+    /**
+     * Method onWebrtcCallIncomming
+     * @private
+     * @param call - the call
+     * @memberof WebrtcGatewayService
+     */
+    private onWebrtcCallIncomming;
+    /**
+     * Method onWebrtcCallConnecting
+     * @private
+     * @param call - the webrtc call
+     * @memberof WebrtcGatewayService
+     */
+    private onWebrtcCallConnecting;
+    /**
+     * Method onWebrtcCallActive
+     * @private
+     * @param call - the webrtc call
+     * @memberof WebrtcGatewayService
+     */
+    private onWebrtcCallActive;
+    /**
+     * Method onWebrtcCallAnswering
+     * @private
+     * @param call - the webrtc call
+     * @memberof WebrtcGatewayService
+     */
+    private onWebrtcCallAnswering;
+    /**
+     * Method onWebrtcCallReleasing
+     * @private
+     * @param call - the webrtc call
+     * @memberof WebrtcGatewayService
+     */
+    private onWebrtcCallReleasing;
+    /**
+     * Method waitTelephonyTimeout
+     * @private
+     * @memberof WebrtcGatewayService
+     * No incoming telephony call received before timeout
+     * We display an incoming call popup with the mediapillar webrtc call if no call exists on secondary device
+     * @param call
+     * @param mediaPillarCallContext
+     */
+    private waitTelephonyTimeout;
+    /**
+     * Method releasePbxCallIfPbxConnectionDown
+     * @private
+     * @memberof WebrtcGatewayService
+     * release the active pbx call if the pbx connection is down: this is called when we receive the session termination on the mp webrtc call
+     */
+    private releasePbxCallIfPbxConnectionDown;
+    /**
+     * Method mediaPillarMakeCall1PCC
+     * @private
+     * @param phoneNumber - the number to join by mediaPillar
+     * @param contact -  the called contact if available
+     * @returns the 1PCC call object
+     * @memberof WebrtcGatewayService
+     * make a media pillar call 1PCC to the phoneNumber
+     */
+    mediaPillarMakeCall1PCC(phoneNumber: string, contact?: Contact): CallRB;
+    /**
+     * Method mediaPillarReleaseCall1PCC
+     * @private
+     * @param call - the 1PCC call to release
+     * @memberof WebrtcGatewayService
+     * release media pillar call 1PCC previously initiated
+     */
+    mediaPillarReleaseCall1PCC(call: CallRB): void;
+    /**
+     * Method mediaPillarMakeCall
+     * @private
+     * @param phoneNumber - the number to join by mediaPillar
+     * @param contact - the called contact if available
+     * @memberof WebrtcGatewayService
+     * make a media pillar call to the phoneNumber
+     */
+    mediaPillarMakeCall(phoneNumber: string, contact?: Contact): void;
+    /**
+     * Method mediaPillarStartCall
+     * @private
+     * @param call - the call to abort
+     * @param phoneNumber - the number to join by mediaPillar
+     * @param mediaPillarJid
+     * @memberof WebrtcGatewayService
+     */
+    private mediaPillarStartCall;
+    /**
+     * Method abortCall
+     * @private
+     * @param call - the call to abort
+     * @memberof WebrtcGatewayService
+     */
+    private abortCall;
+    /**
+     * Method initiateMediaPillarCall
+     * @private
+     * @memberof WebrtcGatewayService
+     * inspired by webrtcP2PServiceEventHandler.prototype.sendProposition
+     * @param call
+     */
+    private initiateMediaPillarCall;
+    /**
+     * Method registerOneMediaPillar
+     * @private
+     * @memberof WebrtcGatewayService
+     * @param MPJid
+     */
+    private registerOneMediaPillar;
+    /**
+     * Method registerMediaPillarPool
+     * @private
+     * @memberof WebrtcGatewayService
+     */
+    private registerMediaPillarPool;
+    /**
+     * Method getMediaPillarData
+     * @public
+     * @returns the mediaPillarJid data associated to the system & user
+     * @memberof WebrtcGatewayService
+     */
+    getMediaPillarData(): Promise<string[]>;
+    /**
+     * Method mediaPillarKeepAlivePolling
+     * @private
+     * @param cmd - the keep alive command
+     * @param timer - the keep alive timer
+     * @memberof WebrtcGatewayService
+     */
+    private mediaPillarKeepAlivePolling;
+    /**
+     * Method mediaPillarUserSelectAndPolling
+     * @public
+     * @param selected - selected, true if MP is selected in nomadic menu
+     * @memberof WebrtcGatewayService
+     */
+    mediaPillarUserSelectAndPolling(selected: boolean): void;
+    /**
+     * Method mediaPillarKeepAlive
+     * @private
+     * @memberof WebrtcGatewayService
+     */
+    mediaPillarKeepAlive(): Promise<void>;
+    /**
+     * Method forceMPpooling
+     * @public
+     * @memberof WebrtcGatewayService
+     */
+    forceMPpooling(): void;
+    /**
+     * Method isMediaPillarPoolConfigured
+     * @private
+     * @returns true if at least one MP is configured (or the specified one when jid is given as parameter)
+     * @memberof WebrtcGatewayService
+     * @param MPJid - when null check all MPs for at least one configured else check only the given MP jid
+     */
+    isMediaPillarPoolConfigured(MPJid: string): boolean;
+    /**
+     * Method isMediaPillarPoolAlive
+     * @private
+     * @returns true if at least one MP is alive (or the specified one when jid is given as parameter)
+     * @memberof WebrtcGatewayService
+     * @param MPJid - when null check all MPs for at least one alive else check only the given MP jid
+     */
+    isMediaPillarPoolAlive(MPJid: string): boolean;
+    /**
+     * Method getAllMediaPillarJid
+     * @public
+     * @returns all mediaPillar Jid
+     * @memberof WebrtcGatewayService
+     */
+    getAllMediaPillarJid(): string[];
+    /**
+     * Method getDefaultMediaPillarJid
+     * @public
+     * @returns the default mediaPillar Jid
+     * @memberof WebrtcGatewayService
+     */
+    getDefaultMediaPillarJid(): string;
+    /**
+     * Method getOnGoingMediaPillarJid
+     * @public
+     * @returns the current mediaPillar Jid for active call (default otherwise)
+     * @memberof WebrtcGatewayService
+     */
+    getOnGoingMediaPillarJid(): string;
+    /**
+     * Method getMyMediaPillarRemoteExtension
+     * @public
+     * @returns my mediaPillar remote extension ("" if not already obtained from server )
+     * @memberof WebrtcGatewayService
+     */
+    getMyMediaPillarRemoteExtension(): string;
+    /**
+     * Method shouldUseMediaPillar
+     * @public
+     * @returns true if media pillar should be used for PBX calls
+     * @memberof WebrtcGatewayService
+     */
+    shouldUseMediaPillar(): boolean;
+    /**
+     * Method isMediaPillarConfigured
+     * @public
+     * @returns true if media pillar is configured for this user (configuration + rights)
+     * @memberof WebrtcGatewayService
+     */
+    isMediaPillarConfigured(): boolean;
+    /**
+     * Method isMediaPillarAvailable
+     * @public
+     * @returns true if media pillar is available for this user (configuration + infra Up)
+     * @memberof WebrtcGatewayService
+     */
+    isMediaPillarAvailable(): boolean;
+    /**
+     * Method isMediaPillarUserSelected
+     * @public
+     * @returns true if media pillar is selected (through Nomadic) for this user
+     * @memberof WebrtcGatewayService
+     */
+    isMediaPillarUserSelected(): boolean;
+    /**
+     * Method isMediaPillarJid
+     * @public
+     * @param fromJid - the from jid
+     * @returns true if it is the media pillar Jid
+     * @memberof WebrtcGatewayService
+     */
+    isMediaPillarJid(fromJid: string): boolean;
+    /**
+     * Method isMediaPillarCallCase
+     * @public
+     * @returns true if Media pillar call
+     * @memberof WebrtcGatewayService
+     * check only the media pillar profile
+     */
+    isMediaPillarCallCase(): boolean;
+    /**
+     * Method isMediaPillarSituation
+     * @public
+     * @param call - a call
+     * @returns true if Media pillar call situation
+     * @memberof WebrtcGatewayService
+     * In all cases test profile and Media pilar availability
+     * If call provided check also that the call is linked to the centralized media pillar context
+     * or if from jid is media pillar
+     */
+    isMediaPillarCallSituation(call?: CallRB): boolean;
+    /**
+     * Method mediaPillaryseTheCall
+     * @public
+     * @param telephonyCall the telephonic call ref
+     * @param webrtcCall  the web rtc call ref
+     * @returns operation status
+     * @memberof WebrtcGatewayService
+     */
+    mediaPillaryseTheCall(telephonyCall: CallRB, webrtcCall: CallRB): string;
+    /**
+     * Method getMediaPillarCallContext
+     * @public
+     * @returns the media pillar call context
+     * @memberof WebrtcGatewayService
+     * Allow to link a call (tel or webrtc) to the media pillar call context
+     * @param call
+     */
+    getMediaPillarCallContext(call?: CallRB): MediaPillarCallContext;
+    /**
+     * Method isMediaPillarOutgoingCall
+     * @public
+     * @returns true if outgoing 3PCC call
+     * @memberof WebrtcGatewayService
+     * return true if the mediaPillarCallContext is used for an outgoing 3PCC call
+     */
+    isMediaPillarOutgoingCall(): boolean;
+    /**
+     * Method isMediaPillarActivableCall
+     * @private
+     * @returns true if
+     * @memberof WebrtcGatewayService
+     * return true if
+     */
+    /**
+     * Method isMediaPillarReleasableCall
+     * @private
+     * @returns true if
+     * @memberof WebrtcGatewayService
+     */
+    private isMediaPillarReleasableCall;
+    /**
+     * Method mediaPillarReleaseWebrtcCall
+     * @public
+     * @memberof WebrtcGatewayService
+     * release only webrtc call (not tel part)
+     * @param mediaPillarCallContext
+     */
+    mediaPillarReleaseWebrtcCall(mediaPillarCallContext: MediaPillarCallContext): void;
+    /**
+     * Method timeOutCallActiveNoWebMedia
+     * @public
+     * @memberof WebrtcGatewayService
+     * @param mediaPillarCallContext
+     */
+    timeOutCallActiveNoWebMedia(mediaPillarCallContext: MediaPillarCallContext): void;
+    /**
+     * Method mediaPillarCallTerminator
+     * @public
+     * @memberof WebrtcGatewayService
+     * Deep release of media pillar context including release of associated calls
+     * @param mediaPillarCallContext
+     */
+    mediaPillarCallTerminator(mediaPillarCallContext?: MediaPillarCallContext): void;
+    /**
+     * @param jid
+     */
+    private createMediaPillarContact;
+    muteAudio(pbxCall: any, muted: any, conversation: any): void;
+    private getActivePbxCall;
+    /**
+     * @param call
+     */
+    answerCall(call: CallRB): Promise<void>;
+    actionOnActiveCallIfNecessary(incomingcall: CallRB): Promise<any>;
+    releaseCall(call: CallRB): void;
+    private specialCaseNotComputerModeButMPCall;
+}
+export {};
+//# sourceMappingURL=webrtcGateway.service.d.ts.map

package/lib/services/webrtcGateway/webrtcGateway.service.d.ts.map

@@ -0,0 +1 @@
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package/lib/strophejs.d.ts

@@ -0,0 +1,3 @@
+import "./libs/strophe/strophe.rsm";
+export declare function initStrophe(): void;
+//# sourceMappingURL=strophejs.d.ts.map

package/lib/strophejs.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"strophejs.d.ts","sourceRoot":"","sources":["../src/strophejs.ts"],"names":[],"mappings":"AACA,OAAO,4BAA4B,CAAC;AAQpC,wBAAgB,WAAW,SAK1B"}

package/lib/typings/helpers.d.ts

@@ -0,0 +1,3 @@
+export type MaybePromise<T> = Promise<T> | T;
+export type MaybeArray<T> = T[] | T;
+//# sourceMappingURL=helpers.d.ts.map

package/lib/typings/helpers.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"helpers.d.ts","sourceRoot":"","sources":["../../src/typings/helpers.ts"],"names":[],"mappings":"AAAA,MAAM,MAAM,YAAY,CAAC,CAAC,IAAI,OAAO,CAAC,CAAC,CAAC,GAAG,CAAC,CAAC;AAC7C,MAAM,MAAM,UAAU,CAAC,CAAC,IAAI,CAAC,EAAE,GAAG,CAAC,CAAC"}

package/lib/utils/allSettled.d.ts

@@ -0,0 +1,2 @@
+export declare function allSettled(promises: any): Promise<any[]>;
+//# sourceMappingURL=allSettled.d.ts.map

package/lib/utils/allSettled.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"allSettled.d.ts","sourceRoot":"","sources":["../../src/utils/allSettled.ts"],"names":[],"mappings":"AAAA,wBAAgB,UAAU,CAAC,QAAQ,KAAA,kBAQlC"}

package/lib/utils/anonymizer.d.ts

@@ -0,0 +1,28 @@
+/**
+ * Anonymisation modes available depending on the data categories and classification rules
+ * defined in document 'Rainbow — PII Masking Policy for ClientApplications Log Files'
+ */
+export declare enum AnonymizeMode {
+    /** Replace every character (except spaces) with '*' */
+    FULL = "FULL",
+    /** Anonymize a person's name: First name / Last name / Nickname*/
+    NAME = "NAME",
+    /** Anonymize a phone number */
+    PHONENUMBER = "PHONENUMBER",
+    /** Anonymize an email address */
+    EMAIL = "EMAIL",
+    /** Anonymize an IP address (v4 or v6) */
+    IPADDR = "IPADDR",
+    /** Anonymize a MAC address */
+    MACADDR = "MACADDR"
+}
+declare const _default: (text: string) => string | null;
+export default _default;
+/**
+ * Anonymizes a string according to the given mode, following the PII Masking Policy.
+ * @param mode - The anonymization strategy to apply (see {@link AnonymizeMode})
+ * @param data - The string to anonymize
+ * @returns The anonymized string, or an empty string if data is falsy
+ */
+export declare function anonymize(mode: AnonymizeMode, data: string): string;
+//# sourceMappingURL=anonymizer.d.ts.map

package/lib/utils/anonymizer.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"anonymizer.d.ts","sourceRoot":"","sources":["../../src/utils/anonymizer.ts"],"names":[],"mappings":"AAEA;;;GAGG;AACH,oBAAY,aAAa;IACrB,uDAAuD;IACvD,IAAI,SAAS;IAEb,kEAAkE;IAClE,IAAI,SAAS;IAEb,+BAA+B;IAC/B,WAAW,gBAAgB;IAE3B,iCAAiC;IACjC,KAAK,UAAU;IAEf,yCAAyC;IACzC,MAAM,WAAW;IAEjB,8BAA8B;IAC9B,OAAO,YAAY;CACtB;yBAOe,MAAM,MAAM,KAAG,MAAM,GAAG,IAAI;AAA5C,wBAUE;AAOF;;;;;GAKG;AACH,wBAAgB,SAAS,CAAC,IAAI,EAAE,aAAa,EAAE,IAAI,EAAE,MAAM,GAAG,MAAM,CAwGnE"}

package/lib/utils/branded.d.ts

@@ -0,0 +1,4 @@
+export type Branded<T, DataType> = T & {
+    __brand: DataType;
+};
+//# sourceMappingURL=branded.d.ts.map

package/lib/utils/branded.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"branded.d.ts","sourceRoot":"","sources":["../../src/utils/branded.ts"],"names":[],"mappings":"AAAA,MAAM,MAAM,OAAO,CAAC,CAAC,EAAE,QAAQ,IAAI,CAAC,GAAG;IAAE,OAAO,EAAE,QAAQ,CAAC;CAAE,CAAC"}