@byteplus/rtc 4.46.3 → 4.47.0

package/index.d.ts

@@ -138,6 +138,10 @@ declare enum ErrorCode {
 	 * @brief The user has been removed from the room because the same user joined the room on the other client.
 	 */
 	DUPLICATE_LOGIN = "DUPLICATE_LOGIN",
+	
+	/** 
+	 * @brief The specified video stream doesn't exist when you take the snapshot.
+	 */
 	INVOKED_BEFORE_CAPTURE = "INVOKED_BEFORE_CAPTURE",
 	
 	/** 
@@ -423,10 +427,12 @@ declare enum AudioMixingError {
 	AUDIO_MIXING_ERROR_INVALID_AUDIO_TRACK = 9
 }
 /**
+ * @id AudioMixingManager
  * @list 104477
  * @detail 104478
  */
 /** 
+ * @id AudioMixingManager
  * @list 104477
  * @detail 104478
  */
@@ -676,13 +682,14 @@ export interface DeviceInfo {
 	 */
 	deviceState: "active" | "inactive";
 }
+/**
+ * {zh}
+ * 音频编码参数细节
+ */
 export interface AudioCodecProfile {
 	chn: number;
 	sr: number;
 }
-/**
- * 音频采集参数下发，定义参考文档：https://bytedance.feishu.cn/sheets/shtcnu2nj9ghFLWiANQb6A1snxh?sheet=7d720e
- */
 export interface IEngineBAC {
 	codec?: {
 		enc_fs?: number;
@@ -695,9 +702,6 @@ export interface IEngineBAC {
 	aec?: any;
 	ans?: any;
 }
-/**
- * 音频编码参数下发，定义参考文档：https://bytedance.feishu.cn/sheets/shtcnu2nj9ghFLWiANQb6A1snxh?sheet=7d720e
- */
 export interface IEngineANM {
 	enc_param?: {
 		bitrate: number;
@@ -707,9 +711,11 @@ export interface IEngineANM {
 }
 export interface IAudioConfig {
 	bitrate?: number;
+	encChannel?: number;
 	channel?: number;
 	playChannel?: number;
 	sampleRate?: number;
+	encSampleRate?: number;
 	playbackRate?: number;
 	agc?: boolean;
 	aec?: boolean;
@@ -728,10 +734,6 @@ declare class AudioProfileManager {
 	private _init;
 	updateEngineBAC(newProfile?: IEngineBAC): void;
 	updateEngineANM(newProfile?: IEngineANM): void;
-	/**
-	 * 通过setAudioProfile设置进来的预设编码参数
-	 * @param profile AudioProfile
-	 */
 	setAudioProfile(profile: AudioProfileType): void;
 	setBitrate(bitrate: number): void;
 	setAEC(enable: boolean): void;
@@ -740,6 +742,42 @@ declare class AudioProfileManager {
 	getOpusConfigStr(): string;
 	get config(): IAudioConfig;
 }
+/** 
+ * @brief Cloud proxy informatio
+ */
+export interface CloudProxyInfo {
+	
+	/** 
+	 * @brief The cloud proxy IP of log report
+	 */
+	logProxy: string;
+	
+	/** 
+	 * @brief The cloud proxy IP of the access
+	 */
+	accessProxy: string;
+	
+	/** 
+	 * @brief The cloud proxy IP of RTC admin control
+	 */
+	configProxy: string;
+}
+export interface RTCContext {
+	audioProfileManager?: AudioProfileManager;
+	serverConfig?: {
+		videoCodec?: string;
+		audioRed: boolean;
+		muteReplaceUnsub: boolean;
+		simulcastOnDemand: boolean;
+	};
+	streamRTT?: {
+		[streamId: string]: {
+			audio: number;
+			video: number;
+		};
+	};
+}
+export declare type RTCExtension = any;
 declare enum StreamMixingEventType {
 	
 	/** 
@@ -787,29 +825,6 @@ declare enum StreamMixingEventType {
 	 */
 	STOP_FAILED = 9
 }
-/** 
- * @brief Cloud proxy informatio
- */
-export interface CloudProxyInfo {
-	
-	/** 
-	 * @brief The cloud proxy IP of log report
-	 */
-	logProxy: string;
-	
-	/** 
-	 * @brief The cloud proxy IP of the access
-	 */
-	accessProxy: string;
-	
-	/** 
-	 * @brief The cloud proxy IP of RTC admin control
-	 */
-	configProxy: string;
-}
-export interface RTCContext {
-	audioProfileManager: AudioProfileManager;
-}
 
 /** 
  * A user joins the room
@@ -895,6 +910,11 @@ export interface onVideoFirstFrameRenderedEvent {
 	 * @brief Width
 	 */
 	width: number;
+	
+	/** 
+	 * @brief If this event comes from a screen video stream
+	 */
+	isScreen: boolean;
 }
 
 /** 
@@ -943,6 +963,11 @@ export interface onAudioFirstFrameDecodedEvent {
 	 * @brief userId of the user published the video stream
 	 */
 	userId: string;
+	
+	/** 
+	 * @brief is this event comes from screen sharing
+	 */
+	isScreen: boolean;
 }
 
 /** 
@@ -1461,19 +1486,32 @@ declare enum EngineEventsTypes {
 	
 	/** 
 	 * @brief The first video frame has been rendered successfully.
+	 * @deprecated
 	 */
 	onVideoFirstFrameRendered = "onVideoFirstFrameRendered",
 	
 	/** 
 	 * @brief The first video frame has been decoded successfully.
+	 * @deprecated
 	 */
 	onVideoFirstFrameDecoded = "onVideoFirstFrameDecoded",
 	
 	/** 
+	 * @brief The first remote video frame has been rendered successfully.
+	 */
+	onRemoteVideoFirstFrame = "onRemoteVideoFirstFrame",
+	
+	/** 
+	 * @deprecated
 	 * @brief The first audio frame has been decoded successfully.
 	 */
 	onAudioFirstFrameDecoded = "onAudioFirstFrameDecoded",
 	
+	/** 
+	 * @brief The first audio frame has been decoded successfully.
+	 */
+	onRemoteAudioFirstFrame = "onRemoteAudioFirstFrame",
+	
 	/** 
 	 * The first video frame has been rendered successfully.
 	 */
@@ -1873,6 +1911,7 @@ declare enum StreamMixingEventErrorCode {
 }
 
 /** 
+ * @id EngineEvents
  * @list 104477
  * @detail 104479
  */
@@ -1992,12 +2031,19 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onVideoFirstFrameRendered]: (event: onVideoFirstFrameRenderedEvent) => void;
 	
 	/** 
-	 * @brief Decoding of the first video frame
+	 * @brief Rendering of the first video frame
 	 * @param event
 	 * @region Video Management Event
 	 */
 	[EngineEventsTypes.onVideoFirstFrameDecoded]: (event: onVideoFirstFrameDecodedEvent) => void;
 	
+	/** 
+	 * @brief Decoding of the first video frame
+	 * @param event
+	 * @region Video Management Event
+	 */
+	[EngineEventsTypes.onRemoteVideoFirstFrame]: (event: onVideoFirstFrameRenderedEvent) => void;
+	
 	/** 
 	 * @brief Decoding of the first audio frame
 	 * @param event
@@ -2005,6 +2051,13 @@ export interface IEngineEvents {
 	 */
 	[EngineEventsTypes.onAudioFirstFrameDecoded]: (event: onAudioFirstFrameDecodedEvent) => void;
 	
+	/** 
+	 * @brief first remote audio frame Received
+	 * @param event
+	 * @region Audio Management Event
+	 */
+	[EngineEventsTypes.onRemoteAudioFirstFrame]: (event: onAudioFirstFrameDecodedEvent) => void;
+	
 	/** 
 	 * @brief Event of video device state changing.
 	 * @param event Information and state of a media device.
@@ -2464,6 +2517,7 @@ export interface IEngineEvents {
 }
 
 /** 
+ * @id RTCEngine
  * @list 104477
  * @detail 104478
  */
@@ -2557,7 +2611,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        You can call this API in both automatic subscription mode and manual subscription mode.
 	 * @param userId The ID of the remote user who published the target audio/video stream.
 	 * @param mediaType Media stream type, used for specifying whether to subscribe to the audio stream or the video stream.
-	 * @notes You must first get the remote stream information through onUserPublishStream{@link #IRTCRoomEventHandler#onUserPublishStream} before calling this API to subscribe to streams accordingly.
+	 * @notes You must first get the remote stream
+	 * information through onUserPublishStream{@link #IRTCRoomEventHandler#onUserPublishStream} before calling this API to subscribe to streams accordingly.
 	 * @region Room Management
 	 */
 	subscribeStream: (userId: string, mediaType: MediaType) => Promise<void>;
@@ -2814,7 +2869,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Sets the video encoder configuration of the publishing streams.
 	 * @param descriptions List of configurations for multiple streams. You should note that the resolution you set is the maximum resolution of each stream.
 	 *        You can set parameters for up to 3 streams, SDK will always take the value of the first 3 streams when parameters for more streams are set.
-	 *        The code rate of the parameters must be in descending order.
+	 *        The resolutions and the frame rates must be in descending order.
 	 *        When the resolution of the max stream is in [540P, 1080P], you can simulcast 3 streams; when in [270P, 450P], you can simulcast 2 streams; when in [0, 180P], you can not enable simulcast.
 	 * @notes + For the video stream captured by the camera, the default value is 360P, 15FPS; for the screen-sharing stream, the default value is 1080P, 15FPS.
 	 *        + Before calling this API to set parameters for multiple video streams, the largest resolution of published streams is 640px × 360px with a frame rate of 15fps by default.
@@ -3034,7 +3089,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @brief Send point-to-point (P2P) text message to a user.
 	 * @param userId ID of the recipient
-	 * @param message Text message within 62 KB.
+	 * @param message Text message within 64 KB.
 	 * @return
 	 * @notes + Call [login](104478@#login) before calling this API.
 	 * + The recipient must be logged into the server and will receive the text message via [EngineEventsTypes.onUserMessageReceivedOutsideRoom](104479#onUserMessageReceivedOutsideRoom) if the message has been sent successfully.
@@ -3044,7 +3099,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @brief Send point-to-point (P2P) binary message to a user.
 	 * @param userId ID of the recipient
-	 * @param message Binary message within 46 KB
+	 * @param message Binary message within 64 KB
 	 * @return + Call [login](#login) before calling this API.
 	 *  + The recipient must be logged into the server and will receive the text message via [EngineEventsTypes.onUserBinaryMessageReceivedOutsideRoom](104479#onuserbinarymessagereceivedoutsideroom) if the message has been sent successfully.
 	 * @notes
@@ -3062,7 +3117,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Send point-to-point (P2P) text message to the server.
-	 * @param message Text message within 62 KB.
+	 * @param message Text message within 64 KB.
 	 * @return
 	 * @notes
 	 */
@@ -3070,7 +3125,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Send point-to-point (P2P) binary message to the server.
-	 * @param message Binary message within 46 KB.
+	 * @param message Binary message within 64 KB.
 	 * @return
 	 * @notes
 	 */
@@ -3243,8 +3298,73 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes We recommend to call this API before the audio capture starts.
 	 */
 	setAudioProfile: (profile: AudioProfileType) => void;
+	
+	/** 
+	 * @brief Get the [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack) of the local audios or videos.
+	 * @param streamIndex Mainstream or screen-sharing stream.
+	 * @param type Video or audio
+	 * Audio: Track of the sounds collected from the microphone, customized audio track, audio files, and the sound card.
+	 * Video: Track of the videos collected from the camera, customized video track and screen-sharings.
+	 * @return `undefined` for the MediaStreamTrack does not exist. For example, the local media capturing has not started.
+	 * @notes + Call this API after you start the local media capture or designate a customized MediaStreamTrack.
+	 * + Callback of the MediaStreamTrack will inform you by the event of `ended` that the local media capture stops, probably caused either by an API call or the removal of devices.
+	 * + You can get the following info from the MediaStreamTrack: video resolution, device ID, and volume. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
+	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.
+	 *   1. Get the video resolution or audio volume of the local streams: [onLocalStreamStats](70081#onlocalstreamstats)
+	 *   2. Get device IDs: [`enumerateDevices`](#enumeratedevices)
+	 *   3. stop: [`stopAudioCapture`](#stopaudiocapture), [`stopVideoCapture`](#stopvideocapture)
+	 */
+	getLocalStreamTrack: (streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
+	
+	/** 
+	 * @brief Get the [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack) of the remote audios or videos.
+	 * @param userId ID of the remote user
+	 * @param streamIndex Mainstream or screen-sharing stream.
+	 * @param type Video or audio
+	 * @return `undefined` for the MediaStreamTrack does not exist. For example, the local user has not subscribed to the remote stream.
+	 * @notes + Call this API after the local user subscribe to the remote media stream.
+	 * + Callback of the MediaStreamTrack will inform you by the event of `ended` that the remote media stream stops, probably caused by the API call of [`unsubscribeStream`](#unsubscribestream) or [`unsubscribeScreen`](#unsubscribescreen).
+	 * + You can get video resolution and audio volume from the MediaStreamTrack. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
+	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.
+	 *   1. Get the video resolution or audio volume of the remote streams: [`onRemoteStreamStats`](70081#onremotestreamstats)
+	 *   2. stop: [`unsubscribeStream`](#unsubscribeStream), [`unsubscribeScreen`](#unsubscribescreen)
+	 */
+	getRemoteStreamTrack: (userId: string, streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
+	
+	/** 
+	 * @brief Get the [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack) of the audios or videos from the subscribed public stream.
+	 * @param id ID of the public stream
+	 * @param type Video or audio
+	 * @return `undefined` for the MediaStreamTrack does not exist. For example, the public stream does not exist.
+	 * @notes + Call this API after you start playing the public stream by calling [`stopPlayPublicStream`](#stopplaypublicstream).
+	 * + Callback of the MediaStreamTrack will inform you the event of `ended` that the public stream stops, probably caused by the API call of [`stopPlayPublicStream`](#stopplaypublicstream).
+	 */
+	getPublicStreamTrack: (id: string, type: "video" | "audio") => MediaStreamTrack | undefined;
+	
+	/** 
+	 * @brief Take a snapshot of the local video.
+	 * @param streamIndex The specified video stream.
+	 * @notes + The snapshot is taken with all video effects on, like rotation, and mirroring.
+	 *        + You can take the snapshot either using SDK internal video capture or customized capture.
+	 */
 	takeLocalSnapshot: (streamIndex: StreamIndex) => Promise<ImageData>;
+	
+	/** 
+	 * @brief Take a snapshot of the remote video.
+	 * @param id The index of the user's stream or the index of the public stream.
+	 * @param streamIndex The specified video stream.
+	 */
 	takeRemoteSnapshot: (id: string, streamIndex: StreamIndex) => Promise<ImageData>;
+	
+	/** 
+	 * @brief Registers an extension.
+	 * @param extension The extension instance.
+	 * @return
+	 * @notes You can use the following extensions.
+	 *        • [Beauty Effects Extension](https://docs.byteplus.com/byteplus-rtc/docs/148646).
+	 *        • [AI Noise-Cancelling Extension](https://docs.byteplus.com/byteplus-rtc/docs/148647).
+	 */
+	registerExtension: (extension: RTCExtension) => Promise<void>;
 }
 
 /** 
@@ -4484,6 +4604,12 @@ export interface EngineConfig {
 	 * @brief Auto-play policy
 	 */
 	autoPlayPolicy: RTCAutoPlayPolicy;
+	
+	/** 
+	 * @hidden
+	 * @brief A private parameter which is used to designate the edge server to access
+	 */
+	expectedIDC: string;
 }
 
 /** 
@@ -4556,20 +4682,24 @@ export interface CloudProxyConnectedEvent {
 declare const Config: {
 	VERSION: string;
 	ICE_CONFIG_REQUEST_URLS_INTERNAL: string[];
-	ICE_CONFIG_REQUEST_URLS: any[];
+	ICE_CONFIG_REQUEST_URLS: never[];
 	EXPECTED_ADDR: string;
 	LOG_SERVER_URL: string;
 	CONFIG_REQUEST_DOMAINS: string[];
 	DEVICE_ID: string;
 	OVERSEA: boolean;
+	PLATFORM: string;
+	PRODUCT: string;
 };
 declare function setParameter(key: keyof typeof Config, value: any): void;
-declare function getParameter(key: keyof typeof Config): string | boolean | any[];
+declare function getParameter(key: keyof typeof Config): string | boolean | string[];
 /**
+ * @id VERTC
  * @list 104477
  * @detail 104478
  */
 /** 
+ * @id VERTC
  * @list 104477
  * @detail 104478
  */
@@ -4678,7 +4808,7 @@ export interface IVERTC {
 }
 declare class VERTC implements IVERTC {
 	getSdkVersion: () => string;
-	createEngine: (appId: string, config?: EngineConfig) => IRTCEngine;
+	createEngine: (appId: string, config?: EngineConfig | undefined) => IRTCEngine;
 	destroyEngine: (engine: IRTCEngine) => void;
 	enumerateDevices: () => Promise<MediaDeviceInfo[]>;
 	enableDevices: (options?: {