@byteplus/rtc 4.55.3 → 4.56.1-meeting.11

package/index.d.ts

@@ -1,5 +1,6 @@
 // Generated by dts-bundle-generator v6.13.0
 
+import { CoreSetParamterInterface } from '@byted/core-lib';
 import { EventArgs, EventEmitter, EventNames, ValidEventTypes } from 'eventemitter3';
 
 /**
@@ -317,11 +318,6 @@ declare enum ErrorCode {
 	 */
 	INVALID_TOKEN = "INVALID_TOKEN",
 	
-	/** 
-	 * @brief Token expired.
-	 */
-	EXPIRED_TOKEN = "EXPIRED_TOKEN",
-	
 	/** 
 	 * @brief Failure of joining room
 	 */
@@ -718,7 +714,17 @@ declare enum ErrorCode {
 	/** 
 	 * @brief Unexpected cross-room media stream forwarding API call. For example, calling `stopForwardStreamToRooms` when media forwarding has already stopped.
 	 */
-	UNEXPECTED_INVOKE_FORWARD_STREAM = "UNEXPECTED_INVOKE_FORWARD_STREAM"
+	UNEXPECTED_INVOKE_FORWARD_STREAM = "UNEXPECTED_INVOKE_FORWARD_STREAM",
+	
+	/** 
+	 * @brief Failed to enter the room because the room is banned.
+	 */
+	ROOM_FORBIDDEN = "ROOM_FORBIDDEN",
+	
+	/** 
+	 * @brief Failed to enter the room because the user is banned.
+	 */
+	USER_FORBIDDEN = "USER_FORBIDDEN"
 }
 declare enum AudioMixingError {
 	
@@ -772,6 +778,16 @@ declare enum AudioMixingError {
 	 */
 	AUDIO_MIXING_ERROR_INVALID_AUDIO_TRACK = 9
 }
+export interface IJoinRoomConfig {
+	useTcpAfterJoinTimeout: boolean;
+	joinWithTcpOnly: boolean;
+	joinWithTcpOnlyDelay: number;
+}
+export interface ExtraJoinRoomParams {
+	serviceType?: string;
+	tenantId?: string;
+	[key: string]: any;
+}
 export declare type RTCExtension = any;
 declare enum StreamMixingEventType {
 	
@@ -1937,7 +1953,9 @@ declare enum EngineEventsTypes {
 	onVendorConnectionStateChanged = "onVendorConnectionStateChanged",
 	onForwardStreamError = "onForwardStreamError",
 	onRejoinWithTcp = "onRejoinWithTcp",
-	onIceConnectWithTcp = "onIceConnectWithTcp"
+	onIceConnectWithTcp = "onIceConnectWithTcp",
+	onPublishRetry = "onPublishRetry",
+	onSubscribeRetry = "onSubscribeRetry"
 }
 
 /** 
@@ -2129,7 +2147,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onAudioDeviceStateChanged]: (event: DeviceInfo) => void;
 	
 	/** 
-	 * @brief Event of media track ending
+	 * @brief Media track ended event. Please restart capture in the callback thread.
 	 * @param event
 	 * @group Video Management
 	 * @order 75
@@ -2432,7 +2450,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onLiveTranscodingResult]: (event: LiveTranscodingResultEvent) => void;
 	
 	/** 
-	 * @brief event of error
+	 * @brief An error occurred in the SDK.
 	 * @param event
 	 * @group Room Management
 	 * @order 40
@@ -2440,9 +2458,15 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onError]: (event: {
 		
 		/** 
-		 * @brief The reason why the current user has been removed from the room.
+		 * @brief See [ErrorCode](104480#errorcode).
+		 */
+		errorCode: ErrorCode.DUPLICATE_LOGIN | ErrorCode.RTM_DUPLICATE_LOGIN | ErrorCode.RTM_TOKEN_ERROR | ErrorCode.TOKEN_EXPIRED | ErrorCode.RECONNECT_FAILED | ErrorCode.KICKED_OUT | ErrorCode.ROOM_DISMISS;
+		
+		/** 
+		 * @brief The Unix timestamp in milliseconds when the room will be unbanned.
+		 *        When `forbiddenTime` is 0, it represents that the user can enter the room again.
 		 */
-		errorCode: ErrorCode.DUPLICATE_LOGIN | ErrorCode.RTM_DUPLICATE_LOGIN | ErrorCode.RTM_TOKEN_ERROR | ErrorCode.EXPIRED_TOKEN | ErrorCode.RECONNECT_FAILED;
+		forbiddenTime?: number;
 	}) => void;
 	
 	/** 
@@ -2708,6 +2732,21 @@ export interface IEngineEvents {
 	 * @hidden
 	 */
 	[EngineEventsTypes.onIceConnectWithTcp]: () => void;
+	
+	/** 
+	 * @hidden
+	 */
+	[EngineEventsTypes.onPublishRetry]: (e: {
+		screen: boolean;
+	}) => void;
+	
+	/** 
+	 * @hidden
+	 */
+	[EngineEventsTypes.onSubscribeRetry]: (e: {
+		userId: string;
+		screen: boolean;
+	}) => void;
 }
 
 /** 
@@ -2727,7 +2766,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param roomId  The Room ID. The string must conform to the regular expression: `[a-zA-Z0-9_@\-\.]{1,128}`.
 	 * @param userInfo  User information.
 	 * @param roomConfig  Room parameter configuration, set the room mode and whether to automatically publish or subscribe to the flow.
-	 * @return
+	 * @return The error code `ErrorCode.ROOM_FORBIDDEN` indicates that the user failed to enter the room because the room has been banned.
+	 *         The error code `ErrorCode.USER_FORBIDDEN` indicates that the user failed to enter the room because the user has been banned.
 	 * @notes + In the same room with the same App ID, the user ID of each user must be unique. The user will be kicked out of the room when a user with the same userID joins the room because of the error of `DUPLICATE_LOGIN`.
 	 *        + Remote users will receive [onUserJoined](104479#onuserjoined) callback when a visible user joins the room. Call [setUserVisibility](#setuservisibility) to switch the visibility of the user.
 	 *        + RTC SDK will try reconnect after losing connection to the server caused by a poor network connection. At this point, [onConnectionStateChanged](104479#onconnectionstatechanged) callback will be triggered and the SDK will automatically retry until it successfully reconnects to the server. The remote user will receive [onUserJoined](104479#onuserjoined) if a visible user reconnected successfully.
@@ -2923,10 +2963,56 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + Call [startAudioCapture](#startaudiocapture) to enable the internal audio capture.
 	 *        + Without calling this API the internal audio capture will sustain until you destroy the engine instance.
 	 * @group Audio Management
-	 * @order 6
+	 * @order 7
 	 */
 	stopAudioCapture: () => Promise<void>;
 	
+	/** 
+	 * @brief Enable audio and video capture using the internal capture module. When calling this method, the browser will request microphone and camera permissions simultaneously, and the user only needs to grant permission once.
+	 *        When calling this method, other users in the room will receive the callback of [onUserStartVideoCapture](104479#onuserstartvideocapture) and [onUserStartAudioCapture](104479#onuserstartaudiocapture).
+	 * @param optionsOrAudioDeviceId When using a single parameter, both audio and video device IDs can be set simultaneously. When using two parameters, the audio device's ID can be set.
+	 * @param videoDeviceId Video device's ID.
+	 * @return The actual audio and video capture parameters that take effect. Refer to [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackSettings). Pay attention to the properties: `deviceId`, `channelCount`, `sampleRate`, `frameRate`, `height` and `width`.
+	 * @notes + The audio and video capture can be published using [publishStream](#publishstream) and unpublished using [unPublishStream](#unpublishstream).
+	 *        + The method cannot be called simultaneously with [startVideoCapture](#startvideocapture) / [startAudioCapture](#startaudiocapture).
+	 *        + If either the camera or the microphone fails to complete the capture when calling this method, the audio and video capture will fail.
+	 *        + To stop audio and video capture, call [stopVideoCapture] (#stopvideocapture) / [stopAudioCapture](#stopaudiocapture).
+	 * @group Audio Management
+	 * @order 6
+	 */
+	startAudioAndVideoCapture: (optionsOrAudioDeviceId?: {
+		
+		/** 
+		 * @brief Audio device's ID.
+		 */
+		audioDeviceId?: string;
+		
+		/** 
+		 * @brief Video device's ID.
+		 */
+		videoDeviceId?: string;
+	} | string, videoDeviceId?: string) => Promise<{
+		audioTrackSettings: MediaTrackSettings;
+		videoTrackSettings: MediaTrackSettings;
+	}>;
+	
+	/** 
+	 * @hidden
+	 * @brief Enable audio and video capture using the internal capture module. When calling this method, the browser will request microphone and camera permissions simultaneously, and the user only needs to grant permission once.
+	 * When calling this method, other users in the room will receive the callback of [onUserStartVideoCapture](104479#onuserstartvideocapture) and [onUserStartAudioCapture](104479#onuserstartaudiocapture).
+	 * @param videoDeviceId Video device's ID.
+	 * @param audioDeviceId Audio device's ID.
+	 * @return The actual audio and video capture parameters that take effect. Refer to [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackSettings). Pay attention to the properties: `deviceId`, `channelCount`, `sampleRate`, `frameRate`, `height` and `width`.
+	 * @notes + The audio and video capture can be published using [publishStream](#publishstream) and unpublished using [unPublishStream](#unpublishstream).
+	 *        + The method cannot be called simultaneously with [startVideoCapture](#startvideocapture) / [startAudioCapture](#startaudiocapture).
+	 *        + If either the camera or the microphone fails to complete the capture when calling this method, the audio and video capture will fail.
+	 *        + To stop audio and video capture, call [stopVideoCapture](#stopvideocapture) / [stopAudioCapture](#stopaudiocapture).
+	 */
+	startVideoAndAudioCapture: (videoDeviceId?: string, audioDeviceId?: string) => Promise<{
+		audioTrackSettings: MediaTrackSettings;
+		videoTrackSettings: MediaTrackSettings;
+	}>;
+	
 	/** 
 	 * @brief Use the capture module provided by the RTC SDK to capture the current screen stream for sharing.
 	 * @param screenConfig Specify properties for the sharing screen.
@@ -2949,22 +3035,22 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Bind a view to the local media stream and set the rendering mode for the rendering of the local stream
-	 * Listen to `onPlayerEvent` for the state after binding a view to the local video stream.
-	 * Call this API again with a blank view to unbind the view from the local video stream.
+	 *        Listen to `onPlayerEvent` for the state after binding a view to the local video stream.
+	 *        Call this API again with a blank view to unbind the view from the local video stream.
 	 * @param streamIndex Video stream properties
 	 * @param videoPlayerOption View information and rendering mode
 	 * @notes You should bind the view after the user joins a room. This setting will sustain when the user leaves the room.
 	 * @group Video Management
 	 * @order 20
 	 */
-	setLocalVideoPlayer: (streamIndex: StreamIndex, videoPlayerOption?: Partial<VideoPlayerOption>) => HTMLVideoElement | undefined;
+	setLocalVideoPlayer: (streamIndex: StreamIndex, videoPlayerOption?: Partial<Omit<VideoPlayerOption, "rotation">>) => HTMLVideoElement | undefined;
 	
 	/** 
 	 * @brief Bind a view to the remote media stream and set the rendering mode for the rendering of the remote stream
 	 *     It's recommended to call this API and bind a view once the event of [onUserPublishStream](104479#onuserpublishstream)/[onUserPublishScreen](104479#onuserpublishscreen) is triggered.
 	 *     Call this API again with a blank view to unbind the view from the remote video stream.
 	 * @param streamIndex Video stream properties
-	 * @param videoPlayer View information and rendering mode
+	 * @param videoPlayerOption View information and rendering mode. Starting from version 4.56, you can set the rotation angle of the remote video rendering using `rotation`.
 	 * @notes You should bind the view after the user joins a room. This setting will sustain when the user leaves the room.
 	 * @group Video Management
 	 * @order 21
@@ -3722,6 +3808,16 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 */
 	setRemoteStreamRenderSync: (sync: boolean) => boolean;
 	
+	/** 
+	 * @hidden
+	 */
+	setJoinRoomParams: (params: ExtraJoinRoomParams) => void;
+	
+	/** 
+	 * @hidden
+	*/
+	connect: () => Promise<void>;
+	
 	/** 
 	 * @brief Adjust audio capture volume.
 	 * @param streamIndex The index of the target stream whose recording volume is to be adjusted.
@@ -3816,6 +3912,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @order 44
 	 */
 	resumeForwardStreamToAllRooms: () => Promise<ForwardStreamStateInfo[]>;
+	peerConnection: RTCPeerConnection | undefined;
+	iceState: RTCIceConnectionState | undefined;
 }
 
 /** 
@@ -4036,8 +4134,8 @@ export declare enum MirrorMode {
  */
 export interface VideoPlayerOption {
 	
-	/**  Dom
-	 *
+	/** 
+	 * @brief DOM
 	 */
 	renderDom?: string | HTMLElement;
 	
@@ -4054,6 +4152,12 @@ export interface VideoPlayerOption {
 	 *      + underscore "_", at character "@", minus, number "-"
 	 */
 	userId: string;
+	
+	/** 
+	 * @brief The clockwise rotation angle of the video frame, which can be set to 0, 90, 180, 270. The default value is 0, which means no rotation is applied.<br>
+	 *        This parameter only applies to remote video. Setting this parameter will not take effect when calling [setLocalVideoPlayer](104478#setlocalvideoplayer).
+	 */
+	rotation?: number;
 }
 
 /** 
@@ -4329,6 +4433,15 @@ export interface VideoEncoderConfig {
 	 * @brief The maximum bitrate in Kbps (>0). The default value is 600 Kbps.
 	 */
 	maxKbps: number;
+	
+	/** 
+	 * @brief The type of content the track contains.
+	 *        `'text'`：The track should be treated as if video details are extra important, and that significant sharp edges and areas of consistent color can occur frequently. For example, presentations or web pages with text content.
+	 *        `'motion'`：The track should be treated as if it contains video where motion is important. For example, webcam video, movies or video games.
+	 *        `'detail'`：The track should be treated as if video details are extra important. For example, presentations or web pages with text content, painting or line art.
+	 * @notes This property is not supported on Firefox. See [contentHint browser compatibility](https://caniuse.com/?search=contentHint).
+	 */
+	contentHint?: "text" | "motion" | "detail";
 }
 
 /** 
@@ -4360,11 +4473,11 @@ export interface ScreenEncoderConfig {
 	maxKbps: number;
 	
 	/** 
-	 * @brief Content types of the shared video.
-	 *        - 'detail' [Default setting]：clarity first, for video content containing a mix of images and text.
-	 *        - 'text'：clarity first, for video content containing only a large amount of text.
-	 *        - 'motion'：fluency first, applicable to scenarios when video content is camera captured source, movies, videos, games.
-	 * @notes Only Chrome browser is supported to set the configuration.
+	 * @brief The type of content the track contains.
+	 *        `'text'`：The track should be treated as if video details are extra important, and that significant sharp edges and areas of consistent color can occur frequently. For example, presentations or web pages with text content.
+	 *        `'motion'`：The track should be treated as if it contains video where motion is important. For example, webcam video, movies or video games.
+	 *        `'detail'`：The track should be treated as if video details are extra important. For example, presentations or web pages with text content, painting or line art.
+	 * @notes This property is not supported on Firefox. See [contentHint browser compatibility](https://caniuse.com/?search=contentHint).
 	 */
 	contentHint?: "text" | "motion" | "detail";
 }
@@ -4409,6 +4522,55 @@ export interface AudioPropertiesConfig {
 	 *        + false: Disable the prompt
 	 */
 	enableInBackground?: boolean;
+	
+	/** 
+	 * @brief Audio callback mode of local main stream.
+	 */
+	localMainReportMode?: LocalMainReportMode;
+	
+	/** 
+	 * @brief The data range of the local main stream audio callback. It controls whether the callback includes mixed audio.
+	 */
+	audioReportMode?: AudioReportMode;
+}
+
+/** 
+ * @brief Audio callback mode of local main stream.
+ *        This feature affects the callback [onActiveSpeaker](104479#onactivespeaker). For example, if [LocalMainReportMode](#localmainreportmode) is set to `RESET` when the local main stream is unpublished, the callback will never include the local user's main stream audio.
+ */
+export declare enum LocalMainReportMode {
+	
+	/** 
+	 * @brief (Default) When the local main stream exists, the callback is always enabled.
+	 */
+	NORMAL = 0,
+	
+	/** 
+	 * @brief When the local main stream is unpublished, the callback is disabled. However, once it is published, the callback function will return the actual volume value.
+	 */
+	DISCONNECT = 1,
+	
+	/** 
+	 * @brief When the local main stream is unpublished, the callback function will return a fixed volume value of 0. However, once it is published, the callback function will return the actual volume value.
+	 */
+	RESET = 2
+}
+
+/** 
+ * @brief The data range of the local main stream audio callback. It controls whether the callback includes mixed audio.
+ *        This feature affects the callback [onActiveSpeaker](104479#onactivespeaker). For example, if [AudioReportMode](#audioreportmode) is set to `AUDIOMIXING`, the callback may still include the local user's main stream audio even with zero microphone volume.
+ */
+export declare enum AudioReportMode {
+	
+	/** 
+	 * @brief (Default) The microphone audio or the audio before mixing.
+	 */
+	MICROPHONE = 0,
+	
+	/** 
+	 * @brief The audio after mixing.
+	 */
+	AUDIOMIXING = 1
 }
 
 /** 
@@ -5428,15 +5590,51 @@ export interface CloudProxyConnectedEvent {
 	 */
 	interval: number;
 }
+export interface SetParamInterface {
+	VERSION: string;
+	ICE_CONFIG_REQUEST_URLS_INTERNAL: string[];
+	ICE_CONFIG_REQUEST_URLS: string[] | string;
+	EXPECTED_ADDR: string;
+	LOG_SERVER_URL: string;
+	CONFIG_REQUEST_DOMAINS: string[];
+	DEVICE_ID: string;
+	OVERSEA: boolean;
+	PLATFORM: string;
+	PRODUCT: string;
+	FORCE_ENABLED_REPORT_CALLBACKS: string[];
+	SKIP_WEB_AUDIO_IN_TRACK: boolean;
+	AUDIO_STALL: boolean;
+	VIDEO_STALL: boolean;
+	STATS_SCALLBACK_SUPPORT: boolean;
+	JOIN_ROOM_CONFIG: IJoinRoomConfig;
+	ENABLE_REPORT_IDB_BUFFER: boolean;
+	SIGNAL_COMPRESSION: boolean;
+	SIGNAL_CROP_JOINROOM: boolean;
+}
+export declare type SetParamKey = keyof (SetParamInterface & CoreSetParamterInterface);
 declare const Config: {
-	[key: string]: any;
+	VERSION: string;
+	ICE_CONFIG_REQUEST_URLS_INTERNAL: string[];
+	ICE_CONFIG_REQUEST_URLS: string[];
+	EXPECTED_ADDR: string;
+	LOG_SERVER_URL: string;
+	CONFIG_REQUEST_DOMAINS: string[];
+	DEVICE_ID: string;
+	OVERSEA: boolean;
+	PLATFORM: string;
+	PRODUCT: string;
+	FORCE_ENABLED_REPORT_CALLBACKS: string[];
+	SKIP_WEB_AUDIO_IN_TRACK: boolean;
+	AUDIO_STALL: boolean;
+	VIDEO_STALL: boolean;
+	STATS_SCALLBACK_SUPPORT: boolean;
+	JOIN_ROOM_CONFIG: IJoinRoomConfig;
+	SIGNAL_COMPRESSION: boolean;
+	SIGNAL_CROP_JOINROOM: boolean;
+	IOS_SAFARI_ORIENTATION: boolean;
 };
-declare function setParameter(key: keyof typeof Config, value: any): {
-	useTcpAfterJoinTimeout: boolean;
-	joinWithTcpOnly: boolean;
-	joinWithTcpOnlyDelay: number;
-} | undefined;
-declare function getParameter(key: keyof typeof Config): any;
+declare function setParameter<T extends SetParamKey>(key: T, value: (SetParamInterface & CoreSetParamterInterface)[T]): IJoinRoomConfig | undefined;
+declare function getParameter(key: keyof typeof Config): string | boolean | string[] | IJoinRoomConfig;
 
 /** 
  * @id BytePlusRTC
@@ -5460,6 +5658,11 @@ export interface IVERTC {
 	 */
 	createEngine: (appId: string, config?: EngineConfig) => IRTCEngine;
 	
+	/** 
+	 * @hidden
+	 */
+	createBLWEngine: (appId: string, config?: EngineConfig) => IRTCEngine;
+	
 	/** 
 	 * @brief Call this API to destroy the IRTCEngine instance created by [createEngine](104478#createengine).
 	 * @param engine By calling this API, this `IRTCEngine` instance and resource related will be wiped from memory. And any interaction with the media server will end.
@@ -5592,9 +5795,17 @@ export interface IVERTC {
 	 */
 	downloadLog: (key?: string) => void;
 }
+/**
+ * @hidden
+ */
+export interface IBLWRTCEngine extends IRTCEngine {
+	
+	setRemoteScreenVideoStreamIndex: (streamIndex: StreamIndex) => boolean;
+}
 declare class VERTC implements IVERTC {
-	getSdkVersion: () => any;
+	getSdkVersion: () => string;
 	createEngine: (appId: string, config?: EngineConfig | undefined) => IRTCEngine;
+	createBLWEngine: (appId: string, config?: EngineConfig | undefined) => IBLWRTCEngine;
 	destroyEngine: (engine: IRTCEngine) => void;
 	enumerateDevices: () => Promise<MediaDeviceInfo[]>;
 	enableDevices: (options?: {