import type { Participant, TrackType, VideoDimension } from './gen/video/sfu/models/models';
import type { AudioSettingsRequestDefaultDeviceEnum, CallRecordingStartedEventRecordingTypeEnum, JoinCallRequest, MemberResponse, OwnCapability, ReactionResponse, StartRecordingRequest, StartRecordingResponse } from './gen/coordinator';
import type { StreamClient } from './coordinator/connection/client';
import type { RejectReason, StreamClientOptions, TokenProvider, User } from './coordinator/connection/types';
import type { Comparator } from './sorting';
import type { StreamVideoWriteableStateStore } from './store';
import { AxiosError } from 'axios';
import type { Call } from './Call';
export type StreamReaction = Pick<ReactionResponse, 'type' | 'emoji_code' | 'custom'>;
export declare enum VisibilityState {
    UNKNOWN = "UNKNOWN",
    VISIBLE = "VISIBLE",
    INVISIBLE = "INVISIBLE"
}
export declare enum DebounceType {
    IMMEDIATE = 20,
    FAST = 100,
    MEDIUM = 600,
    SLOW = 1200
}
export interface StreamVideoParticipant extends Participant {
    /**
     * The participant's audio stream, if they are publishing audio and
     * we have subscribed to it.
     */
    audioStream?: MediaStream;
    /**
     * The participant's video stream, if they are sharing their video,
     * and we are subscribed to it.
     */
    videoStream?: MediaStream;
    /**
     * The participant's screen share stream, if they are sharing their screen,
     * and we are subscribed to it.
     */
    screenShareStream?: MediaStream;
    /**
     * The participant's screen audio stream, if they are sharing their audio,
     * and we are subscribed to it.
     */
    screenShareAudioStream?: MediaStream;
    /**
     * The preferred video dimensions for this participant.
     * Set it to `undefined` to unsubscribe from this participant's video.
     */
    videoDimension?: VideoDimension;
    /**
     * The preferred screen share dimensions for this participant.
     * Set it to `undefined` to unsubscribe from this participant's screen share.
     */
    screenShareDimension?: VideoDimension;
    /**
     * A list of tracks that are currently paused by our servers.
     * Typically, a server-side pause happens when the local participant doesn't
     * have enough bandwidth to receive all tracks. In this case, the server
     * will pause some tracks to optimize the bandwidth usage.
     * Once the bandwidth is restored, the server will resume the paused tracks.
     * This is useful to avoid any unwanted video and audio artifacts.
     */
    pausedTracks?: TrackType[];
    /**
     * The list of tracks that are currently not producing media.
     *
     * For remote participants this is currently surfaced for `TrackType.AUDIO`
     * only and reflects the receiver-side `RTCRtpReceiver` track `mute`/`unmute`
     * state, so it covers system mute on the sender (OS audio session
     * interruption, etc.), the sender pausing its track, sustained RTP stalls,
     * and SFU drops. Remote video and screen-share interruption is not tracked.
     *
     * For the local participant it reflects the local track `mute`/`unmute`
     * events surfaced by the browser (e.g. bluetooth disconnect, OS-level
     * mic/camera kill switch, iOS audio session interruption).
     *
     * Orthogonal to `publishedTracks`: a track can be in `publishedTracks`
     * AND in `interruptedTracks` (the participant intends to publish, but
     * no media is flowing right now).
     */
    interruptedTracks?: TrackType[];
    /**
     * True if the participant is the local participant.
     */
    isLocalParticipant?: boolean;
    /**
     * The pin state of the participant.
     */
    pin?: ParticipantPin;
    /**
     * The last reaction this user has sent to this call.
     * Integrators can batch/collect past reactions and show them to the UI.
     */
    reaction?: StreamReaction;
    /**
     * The visibility state of the participant's tracks within a defined viewport.
     */
    viewportVisibilityState?: Record<VideoTrackType, VisibilityState>;
    /**
     * The volume of the participant's audio stream (from 0 to 1).
     * Set it to `undefined` to use the default volume.
     *
     * Note: this value is not applicable in React Native.
     */
    audioVolume?: number;
}
export type VideoTrackType = 'videoTrack' | 'screenShareTrack';
export type AudioTrackType = 'audioTrack' | 'screenShareAudioTrack';
export type TrackMuteType = 'audio' | 'video' | 'screenshare' | 'screenshare_audio';
/**
 * Represents a participant's pin state.
 */
export type ParticipantPin = {
    /**
     * Set to true if the participant is pinned by the local user.
     * False if the participant is pinned server-side, by the call moderator.
     */
    isLocalPin: boolean;
    /**
     * Timestamp when the participant is pinned.
     */
    pinnedAt: number;
};
export type ClosedCaptionsSettings = {
    /**
     * The time in milliseconds to keep a closed caption in the state (visible).
     * Default is 2700 ms.
     */
    visibilityDurationMs?: number;
    /**
     * The maximum number of closed captions to keep in the state (visible).
     * When the maximum number is reached, the oldest closed caption is removed.
     *
     * Default is 2.
     */
    maxVisibleCaptions?: number;
};
/**
 * A partial representation of the StreamVideoParticipant.
 */
export type StreamVideoParticipantPatch = Partial<StreamVideoParticipant>;
/**
 * A collection of {@link StreamVideoParticipantPatch} organized by sessionId.
 */
export type StreamVideoParticipantPatches = {
    [sessionId: string]: StreamVideoParticipantPatch;
};
export type SubscriptionChange = {
    /**
     * The video dimension to request.
     * Set it to `undefined` in case you want to unsubscribe.
     */
    dimension: VideoDimension | undefined;
};
export type SubscriptionChanges = {
    [sessionId: string]: SubscriptionChange;
};
/**
 * A preferred codec to use when publishing a video or audio track.
 * @internal
 */
export type PreferredCodec = 'vp8' | 'h264' | 'vp9' | 'av1';
/**
 * A collection of track publication options.
 * @internal
 */
export type ClientPublishOptions = {
    /**
     * The preferred codec to use when publishing the video stream.
     */
    preferredCodec?: PreferredCodec;
    /**
     * The fmtp line for the video codec.
     */
    fmtpLine?: string;
    /**
     * The preferred bitrate to use when publishing the video stream.
     */
    preferredBitrate?: number;
    /**
     * The maximum number of simulcast layers to use when publishing the video stream.
     */
    maxSimulcastLayers?: number;
    /**
     * The preferred subscription (incoming video stream) codec.
     */
    subscriberCodec?: PreferredCodec;
    /**
     * The fmtp line for the subscriber codec.
     */
    subscriberFmtpLine?: string;
    /**
     * Screen share settings.
     */
    screenShareSettings?: ScreenShareSettings;
    /**
     * Forces a specific codec to be used when publishing and subscribing a video stream.
     * Never use it in production as it can have unforeseeable consequences.
     * @internal
     */
    dangerouslyForceCodec?: PreferredCodec;
    /**
     * Sets the start bitrate factor to use when publishing a vp9 or av1 or h264 video stream.
     * Must be between 0 and 1.
     * By default the start bitrate is set to 300kbps.
     * This value is used to calculate the start bitrate based on the max bitrate.
     * For example, if the max bitrate is 1500kbps and the start bitrate factor is 0.5, the start bitrate will be 750kbps.
     */
    dangerouslySetStartBitrateFactor?: number;
};
export type ScreenShareSettings = {
    /**
     * Limits the maximum framerate (in frames per second) of the screen share.
     * Defaults to 30.
     */
    maxFramerate?: number;
    /**
     * Limits the maximum bitrate (in bits per second) of the screen share.
     * Defaults to 3000000 (3Mbps).
     */
    maxBitrate?: number;
    /**
     * The content hint to use when publishing the screen share.
     * This can be used to optimize the video quality for different types of content.
     *
     * Defaults to '' (no hint, browser's default behavior).
     * Use 'motion' for video content, 'detail' for presentations or documents, and 'text' for text-heavy content.
     *
     * Please read the documentation for more information on content hints:
     * - https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack/contentHint
     */
    contentHint?: '' | 'motion' | 'detail' | 'text';
};
export type CallLeaveOptions = {
    /**
     * If true, the caller will get a `call.rejected` event.
     * Has an effect only if the call is in the `ringing` state.
     */
    reject?: boolean;
    /**
     * The reason for leaving the call.
     * This will be sent as the `reason` field in the `call.rejected` event.
     */
    reason?: RejectReason;
    /**
     * You can provide extra information about why the call is being left and/or rejected, used for logging purposes.
     */
    message?: string;
};
/**
 * The options to pass to {@link Call} constructor.
 */
export type CallConstructor = {
    /**
     * The streamClient instance to use.
     */
    streamClient: StreamClient;
    /**
     * The Call type.
     */
    type: string;
    /**
     * The Call ID.
     */
    id: string;
    /**
     * An optional list of {@link MemberResponse} from the backend.
     * If provided, the call will be initialized with the data from this object.
     * This is useful when initializing a new "pending call" from an event.
     */
    members?: MemberResponse[];
    /**
     * An optional list of {@link OwnCapability} coming from the backed.
     * If provided, the call will be initialized with the data from this object.
     * This is useful when initializing a new "pending call" from an event.
     */
    ownCapabilities?: OwnCapability[];
    /**
     * Flags the call as a ringing call.
     * @default false
     */
    ringing?: boolean;
    /**
     * Set to true if this call instance should receive updates from the backend.
     *
     * @default false.
     */
    watching?: boolean;
    /**
     * The default comparator to use when sorting participants.
     */
    sortParticipantsBy?: Comparator<StreamVideoParticipant>;
    /**
     * The state store of the client
     */
    clientStore: StreamVideoWriteableStateStore;
};
type StreamVideoClientBaseOptions = {
    apiKey: string;
    options?: StreamClientOptions;
};
type StreamVideoClientOptionsWithoutUser = StreamVideoClientBaseOptions & {
    user?: undefined;
    token?: never;
    tokenProvider?: never;
};
type GuestUser = Extract<User, {
    type: 'guest';
}>;
type AnonymousUser = Extract<User, {
    type: 'anonymous';
}>;
type AuthenticatedUser = Exclude<User, GuestUser | AnonymousUser>;
type StreamVideoClientOptionsWithGuestUser = StreamVideoClientBaseOptions & {
    user: GuestUser;
    token?: never;
    tokenProvider?: never;
};
type StreamVideoClientOptionsWithAnonymousUser = StreamVideoClientBaseOptions & {
    user: AnonymousUser;
    token?: string;
    tokenProvider?: TokenProvider;
};
type StreamVideoClientOptionsWithAuthenticatedUser = StreamVideoClientBaseOptions & {
    user: AuthenticatedUser;
} & ({
    token: string;
    tokenProvider?: TokenProvider;
} | {
    token?: string;
    tokenProvider: TokenProvider;
});
export type StreamVideoClientOptions = StreamVideoClientOptionsWithoutUser | StreamVideoClientOptionsWithGuestUser | StreamVideoClientOptionsWithAnonymousUser | StreamVideoClientOptionsWithAuthenticatedUser;
export type CallRecordingType = CallRecordingStartedEventRecordingTypeEnum;
export type StartCallRecordingFnType = {
    (): Promise<StartRecordingResponse>;
    (type: CallRecordingType): Promise<StartRecordingResponse>;
    (request: StartRecordingRequest): Promise<StartRecordingResponse>;
    (request: StartRecordingRequest, type: CallRecordingType): Promise<StartRecordingResponse>;
};
type StreamRNVideoSDKCallManagerRingingParams = {
    isRingingTypeCall: boolean;
};
type StreamRNVideoSDKCallManagerSetupParams = StreamRNVideoSDKCallManagerRingingParams & {
    defaultDevice: AudioSettingsRequestDefaultDeviceEnum;
};
type StreamRNVideoSDKEndCallReason = 
/** Call ended by the local user (e.g., hanging up). */
'local'
/** Call ended by the remote party, or outgoing call was not answered. */
 | 'remote'
/** Call was rejected/declined by the user. */
 | 'rejected'
/** Remote party was busy. */
 | 'busy'
/** Call was answered on another device. */
 | 'answeredElsewhere'
/** No response to an incoming call. */
 | 'missed'
/** Call failed due to an error (e.g., network issue). */
 | 'error'
/** Call was canceled before the remote party could answer. */
 | 'canceled'
/** Call restricted (e.g., airplane mode, dialing restrictions). */
 | 'restricted'
/** Unknown or unspecified disconnect reason. */
 | 'unknown';
type StreamRNVideoSDKCallingX = {
    joinCall: (call: Call, activeCalls: Call[]) => Promise<void>;
    endCall: (call: Call, reason?: StreamRNVideoSDKEndCallReason) => Promise<void>;
    registerOutgoingCall: (call: Call) => Promise<void>;
};
export type StreamRNVideoSDKGlobals = {
    callingX: StreamRNVideoSDKCallingX;
    callManager: {
        /**
         * Sets up the in call manager.
         */
        setup({ defaultDevice, isRingingTypeCall, }: StreamRNVideoSDKCallManagerSetupParams): void;
        /**
         * Starts the in call manager.
         */
        start({ isRingingTypeCall, }: StreamRNVideoSDKCallManagerRingingParams): void;
        /**
         * Stops the in call manager.
         */
        stop({ isRingingTypeCall }: StreamRNVideoSDKCallManagerRingingParams): void;
    };
    permissions: {
        /**
         * Checks whether a native device permission has been granted.
         */
        check(permission: 'microphone' | 'camera'): Promise<boolean>;
    };
    nativeEvents: {
        speechActivity: {
            /**
             * Subscribes to native speech activity events.
             * Returns an unsubscribe function.
             */
            subscribe(cb: (state: {
                isSoundDetected: boolean;
            }) => void): () => void;
        };
    };
};
declare global {
    var streamRNVideoSDK: StreamRNVideoSDKGlobals | undefined;
}
/**
 * The options to pass to {@link Call.join} method.
 */
export type JoinCallData = Omit<JoinCallRequest, 'location'>;
export { AxiosError };