@byteplus/rtc 4.42.0 → 4.43.1

package/index.d.ts

@@ -123,7 +123,6 @@ declare enum ErrorCode {
 	 * @brief Users have been removed from the room because the administrator dismissed the room by calling OpenAPI.
 	 */
 	ROOM_DISMISS = "ROOM_DISMISS",
-	/** Token过期被踢出房间 */
 	
 	/** 
 	 * @brief Token expired. Call `joinRoom` to rejoin with a valid Token.
@@ -367,7 +366,13 @@ declare enum ErrorCode {
 	/** 
 	 * @brief The user publishes a public stream before joining room.
 	 */
-	START_PUBLIC_STREAM_BEFORE_JOIN = "START_PUBLIC_STREAM_BEFORE_JOIN"
+	START_PUBLIC_STREAM_BEFORE_JOIN = "START_PUBLIC_STREAM_BEFORE_JOIN",
+	
+	/** 
+	 * @brief Reconnection with the server fails and SDK will no longer automatically retry.  <br>
+	 *        Please leave room first and join room again, or contact technical support.
+	 */
+	RECONNECT_FAILED = "RECONNECT_FAILED"
 }
 declare enum AudioMixingError {
 	
@@ -698,6 +703,76 @@ export interface RTCDevice {
 	 */
 	deviceState: "active" | "inactive";
 }
+export interface AudioCodecProfile {
+	chn: number;
+	sr: number;
+}
+export interface IEngineBAC {
+	codec?: any;
+	play?: AudioCodecProfile;
+	record?: AudioCodecProfile;
+	agc?: any;
+	aec?: any;
+	ans?: any;
+}
+export interface IEngineANM {
+	enc_param: {
+		bitrate: number;
+		chn: number;
+	};
+}
+export interface IAudioConfig {
+	bitrate: number;
+	channel: number;
+	sampleRate: number;
+	agc: boolean;
+	aec: boolean;
+	ans: boolean;
+}
+declare class AudioProfileManager {
+	engineBAC: any;
+	engineANM: any;
+	bitrate?: number;
+	sampleRate?: number;
+	channel: number;
+	aec: boolean;
+	agc: boolean;
+	ans: boolean;
+	constructor();
+	private _init;
+	updateEngineBAC(newProfile?: IEngineBAC): void;
+	updateEngineANM(newProfile?: IEngineANM): void;
+	setAudioProfile(profile: AudioProfileType): void;
+	setBitrate(bitrate: number): void;
+	setAEC(enable: boolean): void;
+	setAGC(enable: boolean): void;
+	setANS(enable: boolean): void;
+	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;
+}
 
 /** 
  * A user joins the room
@@ -809,7 +884,12 @@ export interface onVideoFirstFrameDecodedEvent {
 /** 
  * Event related to decoding the first video frame of the public stream.
  */
-export declare type PublicStreamVideoFirstFrameDecodedEvent = Omit<onVideoFirstFrameDecodedEvent, "userId"> & {
+export declare type PublicStreamVideoFirstFrameDecodedEvent = Omit<
+
+/** 
+ * @brief Event related to the decoding of the first video frame.
+ */
+onVideoFirstFrameDecodedEvent, "userId"> & {
 	
 	/** 
 	 * @brief  The public stream's ID.
@@ -1686,10 +1766,14 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief A new stream added into the room.
-	 * @param event `stream` object.
+	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onStreamAdd]: (event: {
+		
+		/** 
+		 * @brief `stream` object.
+		 */
 		stream: RTCStream;
 	}) => void;
 	
@@ -1709,11 +1793,19 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Users will receive this event when a user in the room pauses/ resumes publishing audio stream.
-	 * @param event <li>stream: media stream</li><li>reason</li>.
+	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onStreamRemove]: (event: {
+		
+		/** 
+		* @brief Media stream
+		*/
 		stream: RTCStream;
+		
+		/** 
+		* @brief Reason
+		*/
 		reason: StreamRemoveReason;
 	}) => void;
 	
@@ -1754,31 +1846,55 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Event of media track ending
-	 * @param event <li>kind: string Type of the track</li><li>isScreen: boolean Whether it is a screen-sharing stream</li>
+	 * @param event
 	 * @region Video Management Event
 	 */
 	[EngineEventsTypes.onTrackEnded]: (event: {
+		
+		/** 
+		* @brief Type of the trac
+		*/
 		kind: string;
+		
+		/** 
+		* @brief Whether it is a screen-sharing stream.
+		*/
 		isScreen: boolean;
 	}) => void;
 	
 	/** 
 	 * @brief Event of receiving a broadcast text message in the room
-	 * @param event <li>userId: string userID of the message sender</li><li>message: Contents of the message</li>
+	 * @param event
 	 * @region Message Event
 	 */
 	[EngineEventsTypes.onRoomMessageReceived]: (event: {
+		
+		/** 
+		* @brief userID of the message sender.
+		*/
 		userId: string;
+		
+		/** 
+		* @brief Contents of the message.
+		*/
 		message: any;
 	}) => void;
 	
 	/** 
 	 * @brief Event of receiving a broadcast binary message in the room
-	 * @param event <li>userId: string userID of the message sender</li><li>message: any Contents of the message</li>
+	 * @param event
 	 * @region Message Event
 	 */
 	[EngineEventsTypes.onRoomBinaryMessageReceived]: (event: {
+		
+		/** 
+		* @brief userID of the message sender
+		*/
 		userId: string;
+		
+		/** 
+		* @brief Contents of the message
+		*/
 		message: ArrayBuffer;
 	}) => void;
 	
@@ -1811,7 +1927,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onLocalStreamStats]: (stats: LocalStreamStats) => void;
 	
 	/** 
-	 * @brief Subscribed public stream's statistics and network condition, reference period 2s.
+	 * @brief This 2 s periodical callback notifies information of the public stream after calling [startPlayPublicStream](104478#startplaypublicstream).
 	 * @param stats
 	 */
 	[EngineEventsTypes.onPublicStreamStats]: (stats: PublicStreamStats) => void;
@@ -1821,10 +1937,14 @@ export interface IEngineEvents {
 	 * @deprecated since 4.42, use onLocalAudioPropertiesReport and onRemoteAudioPropertiesReport instead.
 	 * @brief Volume of each of the users in the room
 	 * If the user has not joined a room, the list will only include volume of `local_user`.
-	 * @param event Active user in the room and his/her volume.
+	 * @param event
 	 * @region Room Event
 	 */
 	[EngineEventsTypes.onAudioVolumeIndication]: (event: {
+		
+		/** 
+		 * @brief Active user in the room and his/her volume.
+		 */
 		speakers: AudioVolumeInfo[];
 	}) => void;
 	
@@ -1853,37 +1973,53 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief The remote clients in the room will be informed of the state change via this event after the publisher starts audio capture by calling [startAudioCapture](104478#startAudioCapture).
-	 * @param event `userId: string` The publisher who starts the internal audio capture
+	 * @param event
 	 * @region Audio Management Event
 	 */
 	[EngineEventsTypes.onUserStartAudioCapture]: (event: {
+		
+		/** 
+		 * @brief The publisher who starts the internal audio capture
+		 */
 		userId: string;
 	}) => void;
 	
 	/** 
 	 * @brief The remote clients in the room will be informed of the state change via this event after the publisher stops audio capture by calling [stopAudioCapture](104478#stopAudioCapture).
-	 * @param event `userId: string` userId of the publisher who stops the internal audio capture
+	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onUserStopAudioCapture]: (event: {
+		
+		/** 
+		 * @brief The userId of the publisher who stops the internal audio capture.
+		 */
 		userId: string;
 	}) => void;
 	
 	/** 
 	 * @brief The remote clients in the room will be informed of the state change via this event after the publisher starts video capture by calling [startVideoCapture](104478#startVideoCapture).
-	 * @param event `userId: string` userId of the publisher who starts the internal video capture
+	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onUserStartVideoCapture]: (event: {
+		
+		/** 
+		 * @brief The userId of the publisher who starts the internal video capture.
+		 */
 		userId: string;
 	}) => void;
 	
 	/** 
 	 The remote clients in the room will be informed of the state change via  this event after the publisher stops video capture by calling [stopVideoCapture](104478#stopVideoCapture).
-	 * @param event `userId: string` userId of the publisher who stops the internal video capture
+	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onUserStopVideoCapture]: (event: {
+		
+		/** 
+		 * @brief The userId of the publisher who stops the internal video capture.
+		 */
 		userId: string;
 	}) => void;
 	
@@ -1896,22 +2032,42 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Result of the Auto-Subscription
-	 * @param event <li>`userId: string` userId of the publisher to be subscribed</li><li>`isScreen: boolean` Wether it is a screen-sharing stream</li><li>`state: SubscribeState` </li>
+	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onAutoSubscribeResult]: (event: {
+		
+		/** 
+		 * @brief The userId of the publisher to be subscribed
+		 */
 		userId: string;
+		
+		/** 
+		 * @brief Wether it is a screen-sharing stream
+		 */
 		isScreen: boolean;
+		
+		/** 
+		 * @brief Subscription state.
+		 */
 		state: SubscribeState;
 	}) => void;
 	
 	/** 
 	 * @brief Result of Auto-publishing. You will receive this event if you enable Auto-publish when calling `joinRoom`
-	 * @param event <li>`isScreen: boolean` Wether it is a screen-sharing stream </li><li>`state: PublishState` </li>
+	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onAutoPublishResult]: (event: {
+		
+		/** 
+		 * @brief Wether it is a screen-sharing stream
+		 */
 		isScreen: boolean;
+		
+		/** 
+		 * @brief Publish state.
+		 */
 		state: PublishState;
 	}) => void;
 	
@@ -1931,11 +2087,15 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief event of error
-	 * @param event The reason why the current user has been removed from the room.
+	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onError]: (event: {
-		errorCode: ErrorCode.DUPLICATE_LOGIN | ErrorCode.RTM_DUPLICATE_LOGIN | ErrorCode.RTM_TOKEN_ERROR | ErrorCode.EXPIRED_TOKEN;
+		
+		/** 
+		 * @brief The reason why the current user has been removed from the room.
+		 */
+		errorCode: ErrorCode.DUPLICATE_LOGIN | ErrorCode.RTM_DUPLICATE_LOGIN | ErrorCode.RTM_TOKEN_ERROR | ErrorCode.EXPIRED_TOKEN | ErrorCode.RECONNECT_FAILED;
 	}) => void;
 	
 	/** 
@@ -2005,6 +2165,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Callback of the successful rendering of the first video frame of the public stream.
+	 *         Refer to [startPlayPublicStream](104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
 	 */
 	[EngineEventsTypes.onFirstPublicStreamVideoFrameRendered]: (event: PublicStreamAudioFirstFrameDecodedEvent) => void;
@@ -2017,27 +2178,6 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onAudioPlaybackDeviceTestVolume]: (volume: number) => void;
 }
 
-/** 
- * @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;
-}
-
 /** 
  * @list 104477
  * @detail 104478
@@ -2767,6 +2907,13 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * Before the test ends by itself, you can call this API to stop the recording or playing.
 	 */
 	stopAudioDevicePlayTest(): void;
+	
+	/** 
+	 * @brief Set the sound quality.
+	 * @param profile  A sound quality option consists of a set of audio profiles and applying each brings a different listening experience. Choose the appropriate audio quality to meet the requirement of the scenario.
+	 * @notes We recommend to call this API before the audio capture starts.
+	 */
+	setAudioProfile(profile: AudioProfileType): void;
 }
 
 /** 
@@ -2878,6 +3025,13 @@ export interface RoomConfig {
 	 * Call subscribeUserStream, unsubscribe to revise the subscription.
 	 */
 	isAutoSubscribeVideo: boolean;
+	
+	/** 
+	 * @brief  Room profile<br>
+	 *        Choose an appropriate room-profile option to meet the requirement of the scenario for media encoding algorithm, video profiles, and network configurations.
+	 *        Call [setAudioProfile](104478#setaudioprofile) to choose an audio quality option.
+	 */
+	roomProfileType: RoomProfileType;
 }
 
 /** 
@@ -2923,12 +3077,12 @@ export interface SubscribeConfig {
 export declare enum MuteState {
 	
 	/** 
-	 * @brief Enable
+	 * @brief Disable
 	 */
 	MUTE_STATE_OFF = 0,
 	
 	/** 
-	 * @brief Disable
+	 * @brief Enable
 	 */
 	MUTE_STATE_ON = 1
 }
@@ -3446,7 +3600,7 @@ export interface LiveTranscodeAudioConfig {
 	channels: number;
 	
 	/** 
-	 * @brief The AAC profile to be set. The value must be `LC`, `HEv1`, ` or 'HEv2`. The default value is `LC`.
+	 * @brief The AAC profile to be set. The value must be `LC`, `HEv1`, or `HEv2`. The default value is `LC`.
 	 */
 	AACProfile: "LC" | "HEv1" | "HEv2";
 }
@@ -3720,6 +3874,97 @@ export interface PublicStreamConfig {
 	};
 }
 
+/** 
+ * @brief  Room profile<br>
+ *        While calling [joinRoom](104478#joinroom) choose an appropriate room-profile option to meet the requirement of the scenario for media encoding algorithm, video profiles, and network configurations.
+ *        Call [setAudioProfile](104478#setaudioprofile) to choose an audio quality option if it is necessary.
+ */
+export declare enum RoomProfileType {
+	
+	/** 
+	 * @brief Default mode for video/audio calls, such as voice chat room
+   */
+	communication = 0,
+	
+	/** 
+	 * @brief For 1 v.s. 1 audio and video calls
+   */
+	chat = 5,
+	
+	/** 
+	 * @brief Audio and video chat for 3 or more people
+	 */
+	chatRoom = 6,
+	
+	/** 
+	 * @brief For the event of co-hosting during live streaming broadcast
+	 */
+	coHost = 9,
+	
+	/** 
+	 * @brief For meeting Apps installed on terminals of meeting rooms
+	 */
+	meeting = 16,
+	
+	/** 
+	 * @brief For the online classrooms and lectures that over 9 participants can join the audio and video chat.
+	 */
+	classRoom = 18
+}
+
+/** 
+ * @brief  Audio quality
+ *         Call [setAudioProfile](104478#setaudioprofile) to choose an audio quality option.
+ */
+export declare enum AudioProfileType {
+	
+	/** 
+	 * @brief Default sound quality
+	 *        The sound quality configuration of RoomProfileType{@link #RoomProfileType} set by the server or client side
+	 */
+	default = 0,
+	
+	/** 
+	 * @brief Smooth  <br>
+	 *        Sample rate: 16 KHz<br>
+	 *        Mono-channel<br>
+	 *        Encoding birrate: 24 Kpbs
+	 *        Fluent priority, low power consumption, and low traffic consumption, suitable for most game scenarios, such as team-wide voice chat, group-wide voice chat, nation-wide voice chat.
+	 */
+	fluent = 1,
+	
+	/** 
+	 * @brief Mono-channel standard  <br>
+	 *        Sample rate: 48 KHz<br>
+	 *        Encoding bitrate: 48 Kpbs<br>
+	 *        For scenarios requiring distinct voice, you can choose this mode, which achieves balanced latency, consumption, and network packets. It is suitable for educational Apps and the online Mafia Games.
+	 */
+	standard = 2,
+	
+	/** 
+	 * @brief Dual-channel music  <br>
+	 *        Sample rate: 48 KHz<br>
+	 *        Encoding birrate: 128 Kpbs<br>
+	 *        This mode provides high-resolution audio at a cost of high latency, consumption, and large network packets. It is suitable for entertainment Apps such as co-hosting and online talent contests. <br>
+	 *        Not recommended for game Apps.
+	 */
+	hd = 3,
+	
+	/** 
+	 * @brief Dual-channel standard  <br>
+	 *        Sample rate: 48 KHz<br>
+	 *        Encoding birrate: 80 Kpbs
+	 */
+	standardStereo = 4,
+	
+	/** 
+	 * @brief Mono-channel music  <br>
+	 *        Sample rate: 48 KHz<br>
+	 *        Encoding birrate: 128 Kpbs<br>
+	 */
+	hdMono = 5
+}
+
 /** 
  * @brief Point-to-point text message
  */
@@ -3793,9 +4038,10 @@ declare const Config: {
 	EXPECTED_ADDR: string;
 	LOG_SERVER_URL: string;
 	CONFIG_REQUEST_DOMAINS: string[];
+	DEVICE_ID: string;
 };
 declare function setParameter(key: keyof typeof Config, value: any): void;
-declare function getParameter(key: keyof typeof Config): string | string[] | never[];
+declare function getParameter(key: keyof typeof Config): string | string[];
 /**
  * @list 104477
  * @detail 104478
@@ -3807,10 +4053,10 @@ declare function getParameter(key: keyof typeof Config): string | string[] | nev
 export interface IVERTC {
 	
 	/** 
-	 * @brief Call this API to create an IRTCEngine instance that provides RTC service.   <br>
-	 *        Call this API before calling any other APIs within the RTC SDK.<br>
+	 * @brief Call this API to create an IRTCEngine instance that provides RTC service.
+	 *        Call this API before calling any other APIs within the RTC SDK.
 	 *        Calling this API repeatedly will create multiple IRTCEngine instances.
-	 * @param appId A unique identifier for each application. You can apply appIDs on the RTC console.   <br>
+	 * @param appId A unique identifier for each application. You can apply appIDs on the RTC console.
 	 *              Instances generated by different AppIDs are independent in RTC service. Users served by one AppID cannot communicate with users served by another AppID.
 	 * @return Return an `IRTCEngine` instance if you call this API successfully.
 	 * @notes + Call [destroyEngine](#destroyengine) respectively to destroy each IRTCEngine instance.
@@ -3822,7 +4068,7 @@ export interface IVERTC {
 	/** 
 	 * @brief Call this API to destroy the IRTCEngine instance created by [createEngine](#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.
-	 * @notes + Before calling this API， ensure all business scenarios related to this `IRTCEngine` instance have ended. <br>
+	 * @notes + Before calling this API， ensure all business scenarios related to this `IRTCEngine` instance have ended.
 	 *       + Calling this API will block the main thread until it completes the exiting process. Therefore, do not include this API in a callback. Nor wait in a callback for the execution of this API in the main thread, thus causing a deadlock.
 	 * @region Engine Management
 	 */
@@ -3846,12 +4092,20 @@ export interface IVERTC {
 	/** 
 	 * @brief Call this API to request for access to video and audio devices.
 	 * @param options The device type you need
-	 *                 <li>`video: boolean` Video devices<li>`audio:boolean` Audio devices</li>
-	 * @return <li>`video: boolean` Video devices<li>`audio:boolean` Audio devices</li>
+	 * @return + `video: boolean` Video devices
+	 *         + `audio:boolean` Audio devices
 	 * @region Device Management
 	 */
 	enableDevices: (options?: {
+		
+		/** 
+		 * @brief Video devices
+		 */
 		video: boolean;
+		
+		/** 
+		 * @brief Audio devices
+		 */
 		audio: boolean;
 	}) => Promise<{
 		video: boolean;
@@ -3884,9 +4138,8 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to check whether the browser is supported.
-	 * @return
-	 *         + `True`: Supported
-	 *         - `False`: Not Supported
+	 * @return + `True`: Supported
+	 *         + `False`: Not Supported
 	 * @region Engine Management
 	 */
 	isSupported: () => Promise<boolean>;
@@ -3894,7 +4147,7 @@ export interface IVERTC {
 	/** 
 	 * @brief Gets the codec type supported by current browser.
 	 * If you have enabled vp8 codec, you can call the API to check whether the current browser supports vp8 encoding type before entering the room.
-	 * @return 当前浏览器支持的编解码类型
+	 * @return Codec types supported by the current browser
 	 * @region Engine Management
 	 */
 	getSupportedCodecs: () => Promise<string[]>;