@byteplus/rtc 4.65.3 → 4.66.0

package/index.d.ts

@@ -1,4 +1,4 @@
-// Generated by dts-bundle-generator v8.0.1
+// Generated by dts-bundle-generator v8.1.2
 
 import { EventArgs, EventEmitter, EventNames, ValidEventTypes } from 'eventemitter3';
 
@@ -35,45 +35,11 @@ declare class VERTC implements IVERTC {
 	commitInfo: string;
 	downloadLog: (key?: string | undefined) => void;
 	setLogConfig: ({ logLevel, LogfileSize, }: {
-		logLevel?: "none" | "warning" | "error" | "info" | "trace" | undefined;
+		logLevel?: "none" | "error" | "warning" | "info" | "trace" | undefined;
 		LogfileSize?: number | undefined;
 	}) => void;
 }
-declare const Config: {
-	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;
-	VIDEO_STALL_100MS: boolean;
-	STATS_SCALLBACK_SUPPORT: boolean;
-	JOIN_ROOM_CONFIG: IJoinRoomConfig;
-	SIGNAL_COMPRESSION: boolean;
-	SIGNAL_CROP_JOINROOM: boolean;
-	VIDEO_STALL_DATA: number;
-	AUDIO_STALL_DATA: number;
-	IOS_SAFARI_ORIENTATION: boolean;
-	BLACK_FRAME_LIFETIME: number;
-	FALLBACK_ENCODE_CODEC: string;
-	SEI_TIME_OUT: number;
-	SEI_COUNT_FPS: number;
-	PRE_ICE: boolean;
-	STATS_LOOP_INTERVAL: number;
-	HIDDEN_STATS: boolean;
-	UPLOAD_REMOTE_STATS: ExtendMediaType;
-	SDK_CODEC_NEGOTIATION: boolean;
-	AUDIO_CODEC: string;
-	SKIP_SEI_FILTER: boolean;
-};
+declare const Config: SetParamInterface;
 declare const _default: VERTC;
 declare enum AudioMixingError {
 	
@@ -864,7 +830,9 @@ declare enum ErrorCode {
 	/** 
 	 * @brief Failed to subscribe to the WTN stream via [`startPlayWTN`](docs-104478#wtnstream-startplaywtn).
 	 */
-	WTN_PLAY_FAILED = "WTN_PLAY_FAILED"
+	WTN_PLAY_FAILED = "WTN_PLAY_FAILED",
+	NOT_ALLOWED_IN_RTS_ROOM = "NOT_ALLOWED_IN_RTS_ROOM",
+	NOT_ALLOWED_IN_RESTRICTED_MODE = "NOT_ALLOWED_IN_RESTRICTED_MODE"
 }
 declare enum ExtendMediaType {
 	NONE = 0
@@ -963,7 +931,7 @@ declare enum StreamMixingEventType {
 	STOP_FAILED = 9
 }
 declare function getElectronScreenSources(type?: ScreenSourceType): Promise<ElectronDesktopCapturerSource[]>;
-declare function getParameter(key: keyof typeof Config): string | number | boolean | string[] | IJoinRoomConfig;
+declare function getParameter(key: keyof typeof Config): string | number | boolean | string[] | IJoinRoomConfig | undefined;
 declare function setParameter<T extends SetParamKey>(key: T, value: (SetParamInterface & CoreSetParamterInterface)[T]): IJoinRoomConfig | undefined;
 
 /** 
@@ -1510,6 +1478,24 @@ export declare enum RemoteUserPriority {
 	HIGH = 200
 }
 
+/** 
+ * @brief Room mode.
+ */
+export declare enum RoomMode {
+	
+	/** 
+	 * @brief Create both the RTC room and the RTS room.
+	 * Under this mode, [audio/video call](docs-69871) charges will still be applied even if a user enters the room and only utilizes RTS functionality, without using any audio or video functions. This is because billing starts as soon as a user enters the RTC room (at a minimum, pure audio charges will apply). For detailed billing rules, please refer to [Billing of Audio and Video Calls](docs-69871).
+	 */
+	ROOM_MODE_RTC = 0,
+	
+	/** 
+	 * @brief Create only the RTS room.
+	 * In this mode, the following interfaces cannot be called: `setUserVisibility`、`setAudioSelectionConfig`、`publishStream`、`unpublishStream`、`subscribeStream`、`unsubscribeStream`、`pauseAllSubscribedStream`、`resumeAllSubscribedStream`、`startForwardStreamToRooms`、`updateForwardStreamToRooms`、`stopForwardStreamToRooms`、`pauseForwardStreamToAllRooms`、`resumeForwardStreamToAllRooms`、`publishScreen`、`unpublishScreen`、`subscribeScreen`、`unsubscribeScreen`、`startLiveTranscoding`、`updateLiveTranscoding`、`stopLiveTranscoding`、`sendSEIMessage`、`startSubtitle`、`stopSubtitle`、`updateSubtitleConfig`.
+	 */
+	ROOM_MODE_RTS_ONLY = 1
+}
+
 /** 
  * @brief  Room profile
  *        While calling [joinRoom](docs-104478#joinroom) choose an appropriate room-profile option to meet the requirement of the scenario for media encoding algorithm, video profiles, and network configurations.
@@ -3063,10 +3049,9 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onLocalStreamStats]: (stats: LocalStreamStats) => void;
 	
 	/** 
+	 * @deprecated since 4.65, use [onWTNRemoteVideoStats](docs-104479#wtnstreamevents-onwtnremotevideostats) and [onWTNRemoteAudioStats](docs-104479#wtnstreamevents-onwtnremoteaudiostats) instead.
 	 * @brief This 2 s periodical callback notifies information of the public stream after calling [startPlayPublicStream](docs-104478#startplaypublicstream).
 	 * @param stats
-	 * @group WTN stream
-	 * @order 20
 	 */
 	[EngineEventsTypes.onPublicStreamStats]: (stats: PublicStreamStats) => void;
 	
@@ -3375,7 +3360,7 @@ export interface IEngineEvents {
 	 *        After calling [startPushPublicStream](docs-104478#startpushpublicstream), you will be informed of the result through this callback.
 	 * @param event
 	 * @group WTN stream
-	 * @order 25
+	 * @order 8
 	 */
 	[EngineEventsTypes.onPushPublicStreamResult]: (event: PublicStreamResultEvent) => void;
 	
@@ -3638,13 +3623,13 @@ export interface IJoinRoomConfig {
 export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
-	 * @brief Join the room.
-	 * Call this method to join the room and make audio & video calls with other users in the room.
+	 * @brief Join an existing room or create a new room and join.
+	 * When calling this method, you need to specify the `roomId`. If a room corresponding to the specified roomId already exists, the SDK will directly join that room. If no room corresponds to the specified roomId, the SDK will automatically create a new room using that roomId and join it.
 	 * @param token Dynamic key. It is used for authentication and verification of users entering the room.
 	 *         You need to bring Token to enter the room. When testing, you can use the console to generate temporary tokens. The official launch requires the use of the key SDK to generate and issue tokens at your server level. See [Use Token to complete authentication](docs-70121) for token validity and generation method.
 	 *         Apps with different AppIDs are not interoperable.
 	 *         Make sure that the AppID used to generate the Token is the same as the AppID used to create the engine, otherwise it will cause the join room to fail.
-	 * @param roomId  The Room ID. The string must conform to the regular expression: `[a-zA-Z0-9_@\-\.]{1,128}`.
+	 * @param roomId  The ID of room you want to join or create. 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 + `INVALID_TOKEN`: Failed to enter the room due to using an expired or invalid token.
@@ -4159,7 +4144,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * The SEI PayLoadType must be `5` or `100`.
 	 * @param repeatCount Number of times a message is sent repeatedly. The SEI will be added to a consecutive `repeatCount+1` number of video frames starting from the current frame. The value range is `[0, min{30, %{video frame rate}×2}]`, and the recommended range is `[2,4]`.
 	 * @notes + **Compatibility**<br/>
-	 *         &ensp; - SEI messages only support H.264 encoding, and use on browsers with Chromium 86+.
+	 *         &ensp; - SEI messages are only supported with H.264 encoding and are available only in browsers with Chromium 86+, Firefox 117+, or Safari 15.4+.
 	 *         &ensp; - Android devices do not support sending SEI data in voice call scenarios.
 	 *         &ensp; - SEI messages cannot be used on iOS devices.
 	 *        + In voice call scenarios, you still need to publish and subscribe to video streams, instead of audio streams. The subsciber can enable automatic subscribing to video streams when entering the room, or call [subscribeStream](#subscribestream) with the media stream type `AUDIO_AND_VIDEO`.
@@ -4387,7 +4372,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	setAudioSourceType: (index: StreamIndex, audioSourceType: AudioSourceType) => Promise<void>;
 	
 	/** 
-	 * @brief Create an audio mixing manager object
+	 * @brief Creates an audio mixing manager object.
 	 * @return audio mixing manager object
 	 * @group Audio Mixing
 	 * @order 0
@@ -4395,13 +4380,23 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	getAudioMixingManager: () => IAudioMixingManager;
 	
 	/** 
-	 * @brief Creates a WTN stream manager object
+	 * @brief Creates a WTN stream manager object.
 	 * @return WTN stream manager object
 	 * @group WTN stream
-	 * @order 0
+	 * @order 10
 	 */
 	getWTNStreamManager: () => IWTNStream;
 	
+	/** 
+	 * @brief > Available since v4.66.
+	 * Get the call ID. Call this method after joining a room.
+	 * After creating a room, a `callId` is generated to identify the call for this room.
+	 * @return The call ID (`callId`) of the room.
+	 * @group Room Management
+	 * @order 0
+	 */
+	getCallId: () => string | undefined;
+	
 	/** 
 	 * @brief Configure the internal audio capture. Different browser has its own default settings.
 	 * @param config Configuration of the internal audio capture such as, disabling audio 3A processing. Refer to [MediaTrackConstraints](https://developer.mozilla.org/zh-CN/docs/Web/API/MediaTrackConstraints) for more details.
@@ -4587,7 +4582,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	startPushPublicStream: (publicStreamId: string, publicStreamParam: PublicStreamConfig) => Promise<void>;
 	
 	/** 
-	 * @brief Update the properties of the WTN stream published by the current user via [startPushPublicStream](104478#rtcengine-startpushpublicstream).
+	 * @brief Update the properties of the WTN stream published by the current user via [startPushPublicStream](docs-104478#rtcengine-startpushpublicstream).
 	 * @param publicStreamId The WTN stream's ID.
 	 *                       The WTN stream must be published by the current user.
 	 * @param publicStreamParam The WTN stream's parameters.
@@ -4598,7 +4593,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	updatePublicStreamParam: (publicStreamId: string, publicStreamParam: Partial<PublicStreamConfig>) => Promise<void>;
 	
 	/** 
-	 * @brief Stops publishing the WTN stream published by the current user via [startPushPublicStream](104478#rtcengine-startpushpublicstream).
+	 * @brief Stops publishing the WTN stream published by the current user via [startPushPublicStream](docs-104478#rtcengine-startpushpublicstream).
 	 * @param publicStreamId The WTN stream's ID.
 	 *                       The WTN stream must be published by the current user.
 	 * @return
@@ -4933,7 +4928,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @deprecated since 4.65, use [setWTNRemoteAudioPlaybackVolume](docs-104478#wtnstream-setwtnremoteaudioplaybackvolume) instead.
-	 * @brief Adjust audio playback volume of a WTN stream.
+	 * @brief Adjusts audio playback volume of a WTN stream.
 	 * @param publicStreamId The Id of a specified WTN stream.
 	 * @param volume The percentage of playback volume to the original volume in range of [0, 400]. The default value is 100%.
 	 *               For better call quality, it is recommended to set the value in range of [0, 100].
@@ -5293,7 +5288,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 *        + Call this API with different stream ID to publish multiple WTN streams.
 	 *        + Call [stopPushWTN](#stopPushWTN) to stop publishing the WTN stream.
 	 * @group WTN stream
-	 * @order 1
+	 * @order 11
 	 */
 	startPushWTN: (token: string | null, streamId: string, videoMute: boolean, audioMute: boolean) => Promise<void>;
 	
@@ -5305,7 +5300,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 * @return You will also receive the [onWTNPushStateChanged](docs-104479#wtnstreamevents-onwtnpushstatechanged) callback.
 	 * @notes To stop publishing a WTN stream initiated by [startPushPublicStream](docs-104478#rtcengine-startpushpublicstream), call [stopPushPublicStream](104478#rtcengine-stoppushpublicstream).
 	 * @group WTN stream
-	 * @order 2
+	 * @order 12
 	 */
 	stopPushWTN: (streamId: string) => Promise<void>;
 	
@@ -5318,7 +5313,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 *       - `false`: Include
 	 * @return You will also receive the [onWTNPushStateChanged](docs-104479#wtnstreamevents-onwtnpushstatechanged) callback.
 	 * @group WTN stream
-	 * @order 4
+	 * @order 14
 	*/
 	muteWTNLocalAudio: (streamId: string, mute: boolean) => Promise<void>;
 	
@@ -5331,7 +5326,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 *       - `false`: Include
 	 * @return You will also receive the [onWTNPushStateChanged](docs-104479#wtnstreamevents-onwtnpushstatechanged) callback.
 	 * @group WTN stream
-	 * @order 3
+	 * @order 13
 	*/
 	muteWTNLocalVideo: (streamId: string, mute: boolean) => Promise<void>;
 	
@@ -5348,7 +5343,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	*        - After the message is successfully sent, the remote will receive the [onWTNSEIMessageReceived](docs-104479#wtnstreamevents-onwtnseimessagereceived) callback.
 	*        - If the call fails, neither the local nor the remote will receive the callback.
 	* @group WTN stream
-	* @order 7
+	* @order 37
 	*/
 	sendWTNSEIMessage: (streamId: string, sei: Uint8Array, repeatCount: number) => void;
 	
@@ -5356,7 +5351,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 * @valid since 4.65
 	 * @brief Subscribes to a designated WTN stream.
 	 *        You can call this API to subscribe to a WTN stream whether you joined the room or not.
-	 * @param token Dynamic key. It is used for authentication and verification of users publishing a WTN stream.
+	 * @param token Dynamic key. It is used for authentication and verification of users subscribing to a WTN stream.
 	 *         When testing, you can use the console to generate temporary tokens. The official launch requires the use of the key SDK to generate and issue tokens at your server level. See [Use Token to complete authentication](docs-70121) for token validity and generation method.
 	 *         Apps with different AppIDs are not interoperable.
 	 *         Make sure that the AppID used to generate the Token is the same as the AppID used to create the engine, otherwise it will fail.
@@ -5370,7 +5365,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 *        + If the WTN stream contains SEI message, you will receive the message by be informed through [onPublicStreamSEIMessageReceived](docs-104479#onpublicstreamseimessagereceived).
 	 *        + Call [stopPlayPublicStream](#stopplaypublicstream) to unsubscribing from the WTN stream.
 	 * @group WTN stream
-	 * @order 10
+	 * @order 20
 	 */
 	startPlayWTN: (token: string | null, streamId: string, videoMute: boolean, audioMute: boolean) => Promise<void>;
 	
@@ -5381,7 +5376,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 * @return
 	 * @notes Call [startPlayPublicStream](#startplaypublicstream) to subscribe to the WTN stream.
 	 * @group WTN stream
-	 * @order 12
+	 * @order 22
 	 */
 	stopPlayWTN: (streamId: string) => Promise<void>;
 	
@@ -5394,7 +5389,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 *       - `false`: Include
 	 * @return
 	 * @group WTN stream
-	 * @order 13
+	 * @order 23
 	 */
 	muteWTNRemoteAudio: (streamId: string, mute: boolean) => Promise<void>;
 	
@@ -5408,7 +5403,7 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 * @return
 	 * @notes You can use this method to pause/resume the audio of a WTN stream after you have subscribed to the WTN stream via [startPlayWTN](#startplaywtn).
 	 * @group WTN stream
-	 * @order 13
+	 * @order 24
 	 */
 	muteWTNRemoteVideo: (streamId: string, mute: boolean) => Promise<void>;
 	
@@ -5417,21 +5412,22 @@ export interface IWTNStream extends EnhancedEventEmitter<IWTNStreamEvents> {
 	 * @brief Sets the player information and rendering mode for the WTN stream.
 	 * @param streamId The WTN stream's ID.
 	 * @param videoPlayerOption Player information and rendering mode.
+	 * @return Refer to [HTMLVideoElement](https://developer.mozilla.org/en-US/docs/Web/API/HTMLVideoElement).
 	 * @group WTN stream
-	 * @order 9
+	 * @order 25
 	 */
 	setWTNRemoteVideoPlayer: (streamId: string, videoPlayerOption: Omit<VideoPlayerOption, "userId" | "playerId">) => HTMLVideoElement | undefined;
 	
 	/** 
 	 * @valid since 4.65.
-	 * @brief Adjust audio playback volume of a WTN stream.
+	 * @brief Adjusts audio playback volume of a WTN stream.
 	 * @param streamId The Id of a specified WTN stream.
 	 * @param volume The percentage of playback volume to the original volume in range of [0, 400]. The default value is 100%.
 	 *               For better call quality, it is recommended to set the value in range of [0, 100].
 	 * @notes + It is recommended to set the playback volume after the playing has started.
 	 *        + If the playing has not yet started when you call this API, the setting will take effect after it starts.
 	 * @group WTN stream
-	 * @order 20
+	 * @order 26
 	 */
 	setWTNRemoteAudioPlaybackVolume: (streamId: string, volume: number) => void;
 }
@@ -5448,7 +5444,7 @@ export interface IWTNStreamEvents {
 	 *        After calling [startPushWTN](docs-104478#wtnstream-startpushwtn), you will be informed of the result through this callback.
 	 * @param event
 	 * @group WTN stream
-	 * @order 26
+	 * @order 16
 	 */
 	[WTNStreamEventsTypes.onWTNPushStateChanged]: (event: onWTNPushStateChangedEvent) => void;
 	
@@ -5457,7 +5453,7 @@ export interface IWTNStreamEvents {
 	 *        After calling [startPlayWTN](docs-104478#wtnstream-startplaywtn), you will be informed of the result through this callback.
 	 * @param event
 	 * @group WTN stream
-	 * @order 26
+	 * @order 29
 	 */
 	[WTNStreamEventsTypes.onWTNPlayStateChanged]: (event: onWTNPlayStateChangedEvent) => void;
 	
@@ -5484,7 +5480,7 @@ export interface IWTNStreamEvents {
 	 *        You can receive this callback every 2 seconds after subscribing to the video of a WTN stream via [startPlayWTN](docs-104478#wtnstream-startplaywtn).
 	 * @param event
 	 * @group WTN stream
-	 * @order 30
+	 * @order 33
 	 */
 	[WTNStreamEventsTypes.onWTNRemoteVideoStats]: (event: onWTNRemoteVideoStatsEvent) => void;
 	
@@ -6610,6 +6606,11 @@ export interface RoomConfig {
 	 * @brief Expected configuration of remote video stream.
 	 */
 	remoteVideoConfig?: RemoteVideoConfig;
+	
+	/** 
+	 * @brief Room creation mode. Supports creating both RTC and RTS rooms (default) or only RTS rooms.
+	 */
+	roomMode?: RoomMode;
 }
 
 /** 
@@ -6722,7 +6723,7 @@ export interface SetParamInterface {
 	VIDEO_STALL_100MS: boolean;
 	STATS_SCALLBACK_SUPPORT: boolean;
 	JOIN_ROOM_CONFIG: IJoinRoomConfig;
-	ENABLE_REPORT_IDB_BUFFER: boolean;
+	ENABLE_REPORT_IDB_BUFFER?: boolean;
 	SIGNAL_COMPRESSION: boolean;
 	SIGNAL_CROP_JOINROOM: boolean;
 	VIDEO_STALL_DATA: number;
@@ -6736,14 +6737,16 @@ export interface SetParamInterface {
 	PRE_ICE: boolean;
 	STATS_LOOP_INTERVAL: number;
 	HIDDEN_STATS: boolean;
-	UPLOAD_REMOTE_STATS: string | ExtendMediaType;
+	UPLOAD_REMOTE_STATS: string | SubMediaType;
 	SDK_CODEC_NEGOTIATION: boolean;
-	AINR_CACHE_TIME: number;
-	AINR_DUMP_TIME: number;
-	AINR_ENABLE_DUMP: boolean;
-	AINR_OVERLOAD_THRESHOLD: number;
-	AINR_URLS: string;
+	AINR_CACHE_TIME?: number;
+	AINR_DUMP_TIME?: number;
+	AINR_ENABLE_DUMP?: boolean;
+	AINR_OVERLOAD_THRESHOLD?: number;
+	AINR_URLS?: string;
+	MEDIA_PROCESSING_TYPE?: number;
 	AUDIO_CODEC: string;
+	DISABLE_ENCODED_TRANSFORM: boolean;
 	SKIP_SEI_FILTER: boolean;
 }
 
@@ -7559,6 +7562,7 @@ onVideoFirstFrameDecodedEvent, "userId"> & {
  */
 export type ScreenSourceType = "screen" | "window";
 export type SetParamKey = keyof (SetParamInterface & CoreSetParamterInterface);
+export type SubMediaType = ExtendMediaType | MediaType;
 
 /** 
  * @brief Internal capture configuration for video & audio. Refer to [browser documentation](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints) for more details.