@byteplus/rtc 4.49.8 → 4.51.1

package/index.d.ts

@@ -2,6 +2,285 @@
 
 import { EventArgs, EventEmitter, EventNames, ValidEventTypes } from 'eventemitter3';
 
+/**
+ * @id AudioMixingManager
+ * @list overview
+ * @detail api
+ */
+/** 
+ * @id AudioMixingManager
+ * @list overview
+ * @detail api
+ */
+export interface IAudioMixingManager {
+	
+	/** 
+	 * @brief Start playing audio files and mixes.
+	 *         This method can be called many times by passing in different IDs and filepath to realize simultaneous playback of multiple mixing files and mixing overlay.
+	 * @param id Mix ID. For identifying the mix, please ensure that the mix ID is unique.
+	 *         If this method is repeatedly called with the same ID, the previous mix stops and the next mix starts.
+	 * @param filePath  Specify the file path either the absolute path of a local file or the url of online audio.  Files with the extension of mp3 and aac are supported.
+	 * @param config  Mix Sound configuration.
+	 *         You can set the number of times the mix is played, whether the mix is played locally, and whether the mix is sent to the remote end. For details.
+	 * @return
+	 * @notes + If you have already loaded the sound effect into memory through [preloadAudioMixing](#preloadaudiomixing), make sure that the ID here is the same as the ID set by [preloadAudioMixing](#preloadaudiomixing).
+	 *        + After you start playing the audio file and mixing, you can call the [stopAudioMixing](#stopaudiomixing) method to stop playing the audio file.
+	 *        + The mixing data of this method comes from an external file, while the mixing data of [enableAudioMixingBuffer](#enableaudiomixingbuffer) comes from an external cache and the audio format is PCM. These two mixing methods can coexist.
+	 * @region Advanced Feature
+	 */
+	startAudioMixing: (id: number, filePath: string, options: MixOption) => Promise<void>;
+	
+	/** 
+	 * @brief Stop playing audio files and mixes
+	 * @param id ID of the mixing
+	 * @notes + After calling [startAudioMixing](#startaudiomixing) method to start playing audio files and mixes, you can call this method to stop playing audio files and mixes.
+	 *        + After calling this method to stop playing the audio file, the audio file will be automatically uninstalled.
+	 * @region Advanced Feature
+	 */
+	stopAudioMixing: (id: number) => void;
+	
+	/** 
+	 * @brief Pause playing audio files and mixes
+	 * @param id  ID of the mixing
+	 * @notes + After calling [startAudioMixing](#startaudiomixing) method to start playing audio files and mixes, you can call this method to pause playing audio files and mixes.
+	 *        + After calling this method to pause playing audio files and mixing, you can call the [resumeAudioMixing](#resumeaudiomixing) method to resume playing and mixing.
+	 * @region Advanced Feature
+	 */
+	pauseAudioMixing: (id: number) => void;
+	
+	/** 
+	 * @brief Resume playing audio files and mixing
+	 * @param id  Mixing ID
+	 * @notes After calling the [pauseAudioMixing](#pauseaudiomixing) method to pause playing audio files and mixing, you can resume playing and mixing by calling this method.
+	 * @region Advanced Feature
+	 */
+	resumeAudioMixing: (id: number) => void;
+	
+	/** 
+	 * @brief Preload the specified audio file into memory
+	 * @param id Mix ID. For identifying the mix, please ensure that the mix ID is unique.
+	 *         If this method is called repeatedly with the same ID, the latter overrides the previous.
+	 *         If you call [startAudioMixing](#startaudiomixing) and then call this method with the same ID, the previous mix stopped, and then the next mix was loaded.
+	 *         After calling this method with an ID to preload A.mp3, if you need to call [startAudioMixing](#startaudiomixing) to play B.mp3 with the same ID, please call [unloadAudioMixing](#unloadaudiomixing) to uninstall A.mp3 first, otherwise you will report an error AUDIO_MIXING_ERROR_LOAD_CONFLICT.
+	 * @param filePath  Used for mixing file paths. It can be the absolute path of the local file or the url of the online audio.
+	 * @return
+	 * @notes + When you need to play an audio file frequently, call this method to preload the file. When playing, you can only load the file once to reduce CPU usage.
+	 *        + This method pre-loads the specified audio file, and only calling [startAudioMixing](#startaudiomixing) API can you start playing the specified audio file.
+	 *        + The specified audio file preloaded by calling this method can be unloaded through [unloadAudioMixing](#unloadaudiomixing).
+	 * @region Advanced Feature
+	 */
+	preloadAudioMixing: (id: number, filePath: string) => Promise<void>;
+	
+	/** 
+	 * @brief Uninstall the specified audio file
+	 * @param id Mix ID
+	 * @region Advanced Feature
+	 */
+	unloadAudioMixing: (id: number) => void;
+	
+	/** 
+	 * @brief Adjust the volume of audio mixing, including audio files and PCM data.
+	 * @param id Mixing ID
+	 * @param volume The ratio of the mixing volume to the original volume. The range is `[0, 400]`. The recommended range is `[0, 100]`.
+	 *         + 0: Mute
+	 *         + 100: Original volume
+	 *         + 400: Maximum volume (with overflow protection)
+	 * @region Advanced Feature
+	 */
+	setAudioMixingVolume: (id: number, volume: number) => void;
+	
+	/** 
+	 * @brief Get volume of the audio files.
+	 * @param id Audio mix id
+	 * @return + `0`: Failure
+	 *         + `>0`: Successful. Volume
+	 * @region Advanced Feature
+	 */
+	getAudioMixingVolume: (id: number) => number;
+	
+	/** 
+	 * @brief Get audio file duration
+	 * @param id   Mix ID
+	 * @return + `> 0`: Success, audio file duration, in milliseconds.
+	 *         + `< 0`: Failure
+	 * @notes  Before calling this method to obtain the length of the audio file, you need to call [preloadAudioMixing](#preloadaudiomixing) or [startAudioMixing](#startaudiomixing).
+	 * @region Advanced Feature
+	 */
+	getAudioMixingDuration: (id: number) => number;
+	
+	/** 
+	 * @brief Get audio file playback progress
+	 * @param id   Mix ID
+	 * @return + `> 0`: Success, audio file playback progress, in milliseconds.
+	 *         + `< 0`: Failed
+	 * @notes Before calling this method to get the audio file playback progress, you need to call [startAudioMixing](#startaudiomixing) to start playing the audio file.
+	 * @region Advanced Feature
+	 */
+	getAudioMixingCurrentPosition: (id: number) => number;
+	
+	/** 
+	 * @brief Set the starting playback position of the audio file
+	 * @param id Mixing ID
+	 * @param position The starting playback position of the audio file in milliseconds.
+	 *                 Call [getAudioMixingDuration](#getaudiomixingduration) to get the duration of the audio file. The value of  `position` must not be greater than the duration of the audio file.
+	 * @notes + This method allows you to play an audio file from a specified position as appropriate, without having to play an audio file in its entirety from the beginning to finish.
+	 *       + Before calling this method to set the playback position of the audio file, you need to call [startAudioMixing](#startaudiomixing) to start playing the audio file.
+	 * @region Advanced Feature
+	 */
+	setAudioMixingPosition: (id: number, position: number) => void;
+	
+	/** 
+	 * @brief Start PCM mixing.
+	 *        To mix multiple PCM audio data, call this API multiple times with different ids.
+	 * @param type Mixing type.
+	 *        Whether the PCM data is mixed locally and sent to the remotes.
+	 * @notes + After this API, you must call [pushAudioMixingBuffer](#pushaudiomixingbuffer) to start mixing.
+	 *        + To end PCM mixing, use [disableAudioMixingBuffer](#disableaudiomixingbuffer).
+	 * @region Advanced Feature
+	 */
+	enableAudioMixingBuffer: (type: AudioMixingType) => void;
+	
+	/** 
+	 * @brief Disable PCM mixing.
+	 * @region Advanced Feature
+	 */
+	disableAudioMixingBuffer: () => void;
+	
+	/** 
+	 * @brief Push PCM audio frame data for mixing
+	 * @param buffer Refer to [Browser documentation](https://developer.mozilla.org/en-US/docs/Web/API/AudioBuffer)
+	 * @notes Before calling this method, you must start the external audio stream mixing through [enableAudioMixingBuffer](#enableaudiomixingbuffer).
+	 * @region Advanced Feature
+	 */
+	pushAudioMixingBuffer: (buffer: AudioBuffer) => void;
+}
+
+/** 
+ * @brief Mix playback channel type
+ */
+export declare enum AudioMixingDualMonoMode {
+	
+	/** 
+	 * @brief Consistent with audio files
+	 */
+	AUTO = 0,
+	
+	/** 
+	 * @brief Only the left channel audio in the audio file can be heard
+	 */
+	MODE_L = 1,
+	
+	/** 
+	 * @brief Only the right channel audio in the audio file can be heard
+	 */
+	MODE_R = 2,
+	
+	/** 
+	 * @brief Can hear the left and right audio channels in the audio file at the same time
+	 */
+	MODE_MIX = 3
+}
+
+/** 
+ * @brief Mixing type
+ */
+export declare enum AudioMixingType {
+	
+	/** 
+	 * @brief Played at the local device only
+	 */
+	PLAYOUT = 0,
+	
+	/** 
+	 * @brief Sent to the remote devices only
+	 */
+	PUBLISH = 1,
+	
+	/** 
+	 * @brief Played at the local device and sent to the remote devices.
+	 */
+	PLAYOUT_AND_PUBLISH = 2
+}
+declare enum AudioMixingState {
+	
+	/** 
+	 * @brief Mix loaded
+	 */
+	AUDIO_MIXING_STATE_PRELOADED = 0,
+	
+	/** 
+	 * @brief Mix is playing
+	 */
+	AUDIO_MIXING_STATE_PLAYING = 1,
+	
+	/** 
+	 * @brief Mix Pause
+	 */
+	AUDIO_MIXING_STATE_PAUSED = 2,
+	
+	/** 
+	 * @brief Mixing stopped
+	 */
+	AUDIO_MIXING_STATE_STOPPED = 3,
+	
+	/** 
+	 * @brief Mix playback failed
+	 */
+	AUDIO_MIXING_STATE_FAILED = 4,
+	
+	/** 
+	 * @brief End of mixing
+	 */
+	AUDIO_MIXING_STATE_FINISHED = 5,
+	
+	/** 
+	 * @brief Prepare PCM Mix
+	 */
+	AUDIO_MIXING_STATE_PCM_ENABLED = 6,
+	
+	/** 
+	 * @brief End of PCM mix playback
+	 */
+	AUDIO_MIXING_STATE_PCM_DISABLED = 7
+}
+
+/** 
+ * @brief Mixing configuration
+ */
+export interface MixOption {
+	
+	/** 
+	 * @brief Mix playback times
+	 * @notes + Play_count < = 0: Infinite loop
+	 *        + Play_count == 1: Play once (default)
+	 *        + Play_count > 1: Play play_count times
+	 */
+	playCount: number;
+	
+	/** 
+	 * @brief For mixing playback types.
+	 */
+	type?: AudioMixingType;
+}
+declare class EnhancedEventEmitter<EventTypes extends ValidEventTypes = string | symbol> extends EventEmitter<EventTypes> {
+	safeEmit<T extends EventNames<EventTypes>>(event: T, ...args: EventArgs<EventTypes, T>): boolean;
+}
+
+/** 
+ * @brief Information and state of a media device.
+ */
+export interface DeviceInfo {
+	
+	/** 
+	 * @brief Information of a media device. See [MediaDeviceInfo](https://developer.mozilla.org/en-US/docs/Web/API/MediaDeviceInfo).
+	 */
+	mediaDeviceInfo: MediaDeviceInfo;
+	
+	/** 
+	 * @brief Device state.
+	 */
+	deviceState: "active" | "inactive";
+}
 declare enum ErrorCode {
 	
 	/** 
@@ -320,10 +599,17 @@ declare enum ErrorCode {
 	USER_MESSAGE_NO_RECEIVER = "USER_MESSAGE_NO_RECEIVER",
 	
 	/** 
+	 * @hidden
+	 * @deprecated since 4.49, use USER_MESSAGE_NO_RECEIVER instead.
 	 * @brief No relay path
 	 */
 	USER_MESSAGE_NO_RELAYPATH = "USER_MESSAGE_NO_RELAYPATH",
 	
+	/** 
+	 * @brief Exceeds QPS limit
+	 */
+	USER_MESSAGE_EXCEED_QPS = "USER_MESSAGE_EXCEED_QPS",
+	
 	/** 
 	 * @brief No response after sending messages to a server probably due to Http request failure
 	 */
@@ -338,409 +624,142 @@ declare enum ErrorCode {
 	 * @brief The user has not joined any room.
 	 */
 	USER_MESSAGE_NOT_JOIN = "USER_MESSAGE_NOT_JOIN",
-	
-	/** 
-	 * @brief Connection has not initiated.
-	 */
-	USER_MESSAGE_INIT = "USER_MESSAGE_INIT",
-	
-	/** 
-	 * @brief No available data connection
-	 */
-	USER_MESSAGE_NO_CONNECTION = "USER_MESSAGE_NO_CONNECTION",
-	
-	/** 
-	 * @brief Sending a message before login to the IM server
-	 */
-	USER_MESSAGE_NOT_LOGIN = "USER_MESSAGE_NOT_LOGIN",
-	
-	/** 
-	 * @brief Lack of params of the IM server
-	 */
-	USER_MESSAGE_SERVER_PARAMS_NOTSET = "USER_MESSAGE_SERVER_PARAMS_NOTSET",
-	
-	/** 
-	 * @brief Unknown error of messaging
-	 */
-	USER_MESSAGE_UNKNOWN = "USER_MESSAGE_UNKNOWN",
-	
-	/** 
-	 * @brief The user publishes a public stream before joining room.
-	 */
-	START_PUBLIC_STREAM_BEFORE_JOIN = "START_PUBLIC_STREAM_BEFORE_JOIN",
-	
-	/** 
-	 * @brief Reconnection with the server fails and SDK will no longer automatically retry.
-	 *        Please leave room first and join room again, or contact technical support.
-	 */
-	RECONNECT_FAILED = "RECONNECT_FAILED"
-}
-declare enum AudioMixingError {
-	
-	/** 
-	 * @brief Normal
-	 */
-	AUDIO_MIXING_ERROR_OK = 0,
-	
-	/** 
-	 * @brief Pre-load fails because of missing audio file or the length of the audio file exceeding 20s.
-	 */
-	AUDIO_MIXING_ERROR_PRELOAD_FAILED = 1,
-	
-	/** 
-	 * @brief Audio mixing starts fails because of missing audio file or failure of reading the audio file.
-	 */
-	AUDIO_MIXING_ERROR_START_FAILED = 2,
-	
-	/** 
-	 * @brief Invalid audio mixing ID
-	 */
-	AUDIO_MIXING_ERROR_ID_NOT_FOUND = 3,
-	
-	/** 
-	 * @brief Invalid start position
-	 */
-	AUDIO_MIXING_ERROR_SET_POSITION_FAILED = 4,
-	
-	/** 
-	 * @brief Invalid volume argument. The volume value ranges [0, 400].
-	 */
-	AUDIO_MIXING_ERROR_INVALID_VOLUME = 5,
-	
-	/** 
-	 * @brief The audio file to be played does not match to the preload one. Call `unloadAudioMixing` to unload file and proceed.
-	 */
-	AUDIO_MIXING_ERROR_LOAD_CONFLICT = 6,
-	
-	/** 
-	 * @brief Type of the audio file to be played is not supported.
-	 */
-	AUDIO_MIXING_ERROR_ID_TYPE_NOT_MATCH = 7,
-	
-	/** 
-	 * @brief Invalid audio pitch
-	 */
-	AUDIO_MIXING_ERROR_ID_TYPE_INVALID_PITCH = 8,
-	
-	/** 
-	 * @brief Invalid audio track
-	 */
-	AUDIO_MIXING_ERROR_INVALID_AUDIO_TRACK = 9
-}
-/**
- * @id AudioMixingManager
- * @list 104477
- * @detail 104478
- */
-/** 
- * @id AudioMixingManager
- * @list 104477
- * @detail 104478
- */
-export interface IAudioMixingManager {
-	
-	/** 
-	 * @brief Start playing audio files and mixes.
-	 *         This method can be called many times by passing in different IDs and filepath to realize simultaneous playback of multiple mixing files and mixing overlay.
-	 * @param id Mix ID. For identifying the mix, please ensure that the mix ID is unique.
-	 *         If this method is repeatedly called with the same ID, the previous mix stops and the next mix starts.
-	 * @param filePath  Specify the file path either the absolute path of a local file or the url of online audio.  Files with the extension of mp3 and aac are supported.
-	 * @param config  Mix Sound configuration.
-	 *         You can set the number of times the mix is played, whether the mix is played locally, and whether the mix is sent to the remote end. For details.
-	 * @return
-	 * @notes + If you have already loaded the sound effect into memory through [preloadAudioMixing](#preloadaudiomixing), make sure that the ID here is the same as the ID set by [preloadAudioMixing](#preloadaudiomixing).
-	 *        + After you start playing the audio file and mixing, you can call the [stopAudioMixing](#stopaudiomixing) method to stop playing the audio file.
-	 *        + The mixing data of this method comes from an external file, while the mixing data of [enableAudioMixingFrame](#enableAudioMixingFrame) comes from an external cache and the audio format is PCM. These two mixing methods can coexist.
-	 * @region Advanced Feature
-	 */
-	startAudioMixing: (id: number, filePath: string, options: MixOption) => Promise<void>;
-	
-	/** 
-	 * @brief Stop playing audio files and mixes
-	 * @param id ID of the mixing
-	 * @notes + After calling [startAudioMixing](#startaudiomixing) method to start playing audio files and mixes, you can call this method to stop playing audio files and mixes.
-	 *        + After calling this method to stop playing the audio file, the audio file will be automatically uninstalled.
-	 * @region Advanced Feature
-	 */
-	stopAudioMixing: (id: number) => void;
-	
-	/** 
-	 * @brief Pause playing audio files and mixes
-	 * @param id  ID of the mixing
-	 * @notes + After calling [startAudioMixing](#startaudiomixing) method to start playing audio files and mixes, you can call this method to pause playing audio files and mixes.
-	 *        + After calling this method to pause playing audio files and mixing, you can call the [resumeAudioMixing](#resumeaudiomixing) method to resume playing and mixing.
-	 * @region Advanced Feature
-	 */
-	pauseAudioMixing: (id: number) => void;
-	
-	/** 
-	 * @brief Resume playing audio files and mixing
-	 * @param id  Mixing ID
-	 * @notes After calling the [pauseAudioMixing](#pauseaudiomixing) method to pause playing audio files and mixing, you can resume playing and mixing by calling this method.
-	 * @region Advanced Feature
-	 */
-	resumeAudioMixing: (id: number) => void;
-	
-	/** 
-	 * @brief Preload the specified audio file into memory
-	 * @param id Mix ID. For identifying the mix, please ensure that the mix ID is unique.
-	 *         If this method is called repeatedly with the same ID, the latter overrides the previous.
-	 *         If you call [startAudioMixing](#startaudiomixing) and then call this method with the same ID, the previous mix stopped, and then the next mix was loaded.
-	 *         After calling this method with an ID to preload A.mp3, if you need to call [startAudioMixing](#startaudiomixing) to play B.mp3 with the same ID, please call [unloadAudioMixing](#unloadaudiomixing) to uninstall A.mp3 first, otherwise you will report an error AUDIO_MIXING_ERROR_LOAD_CONFLICT.
-	 * @param filePath  Used for mixing file paths. It can be the absolute path of the local file or the url of the online audio.
-	 * @return
-	 * @notes + When you need to play an audio file frequently, call this method to preload the file. When playing, you can only load the file once to reduce CPU usage.
-	 *        + This method pre-loads the specified audio file, and only calling [startAudioMixing](#startaudiomixing) API can you start playing the specified audio file.
-	 *        + The specified audio file preloaded by calling this method can be unloaded through [unloadAudioMixing](#unloadaudiomixing).
-	 * @region Advanced Feature
-	 */
-	preloadAudioMixing: (id: number, filePath: string) => Promise<void>;
-	
-	/** 
-	 * @brief Uninstall the specified audio file
-	 * @param id Mix ID
-	 * @region Advanced Feature
-	 */
-	unloadAudioMixing: (id: number) => void;
-	
-	/** 
-	 * @brief Adjust the volume of audio mixing, including audio files and PCM data.
-	 * @param id Mixing ID
-	 * @param volume The ratio of the mixing volume to the original volume. The range is `[0, 400]`. The recommended range is `[0, 100]`.
-	 *         + 0: Mute
-	 *         + 100: Original volume
-	 *         + 400: Maximum volume (with overflow protection)
-	 * @region Advanced Feature
-	 */
-	setAudioMixingVolume: (id: number, volume: number) => void;
-	
-	/** 
-	 * @brief Get volume of the audio files.
-	 * @param id Audio mix id
-	 * @return + `0`: Failure
-	 *         + `>0`: Successful. Volume
-	 * @region Advanced Feature
-	 */
-	getAudioMixingVolume: (id: number) => number;
-	
-	/** 
-	 * @brief Get audio file duration
-	 * @param id   Mix ID
-	 * @return + `> 0`: Success, audio file duration, in milliseconds.
-	 *         + `< 0`: Failure
-	 * @notes  Before calling this method to obtain the length of the audio file, you need to call [preloadAudioMixing](#preloadaudiomixing) or [startAudioMixing](#startaudiomixing).
-	 * @region Advanced Feature
+	
+	/** 
+	 * @hidden
+	 * @deprecated
+	 * @brief Connection has not initiated.
 	 */
-	getAudioMixingDuration: (id: number) => number;
+	USER_MESSAGE_INIT = "USER_MESSAGE_INIT",
 	
 	/** 
-	 * @brief Get audio file playback progress
-	 * @param id   Mix ID
-	 * @return + `> 0`: Success, audio file playback progress, in milliseconds.
-	 *         + `< 0`: Failed
-	 * @notes Before calling this method to get the audio file playback progress, you need to call [startAudioMixing](#startaudiomixing) to start playing the audio file.
-	 * @region Advanced Feature
+	 * @hidden
+	 * @deprecated since 4.49, use USER_MESSAGE_UNKNOWN instead.
+	 * @brief No available data connection
 	 */
-	getAudioMixingCurrentPosition: (id: number) => number;
+	USER_MESSAGE_NO_CONNECTION = "USER_MESSAGE_NO_CONNECTION",
 	
 	/** 
-	 * @brief Set the starting playback position of the audio file
-	 * @param id Mixing ID
-	 * @param position The starting playback position of the audio file in milliseconds.
-	 *                 Call [getAudioMixingDuration](#getAudioMixingDuration) to get the duration of the audio file. The value of  `position` must not be greater than the duration of the audio file.
-	 * @notes + This method allows you to play an audio file from a specified position as appropriate, without having to play an audio file in its entirety from the beginning to finish.
-	 *       + Before calling this method to set the playback position of the audio file, you need to call [startAudioMixing](#startaudiomixing) to start playing the audio file.
-	 * @region Advanced Feature
+	 * @brief Sending a message before login to the IM server
 	 */
-	setAudioMixingPosition: (id: number, position: number) => void;
+	USER_MESSAGE_NOT_LOGIN = "USER_MESSAGE_NOT_LOGIN",
 	
 	/** 
-	 * @brief Start PCM mixing.
-	 *        To mix multiple PCM audio data, call this API multiple times with different ids.
-	 * @param type Mixing type.
-	 *        Whether the PCM data is mixed locally and sent to the remotes.
-	 * @notes + After this API, you must call [pushAudioMixingFrame](#pushaudiomixingframe) to start mixing.
-	 *        + To end PCM mixing, use [disableAudioMixingFrame](#disableaudiomixingframe).
-	 * @region Advanced Feature
+	 * @brief Lack of params of the IM server
 	 */
-	enableAudioMixingBuffer: (type: AudioMixingType) => void;
+	USER_MESSAGE_SERVER_PARAMS_NOTSET = "USER_MESSAGE_SERVER_PARAMS_NOTSET",
 	
 	/** 
-	 * @brief Disable PCM mixing.
-	 * @region Advanced Feature
+	 * @brief Unknown error of messaging
 	 */
-	disableAudioMixingBuffer: () => void;
+	USER_MESSAGE_UNKNOWN = "USER_MESSAGE_UNKNOWN",
 	
 	/** 
-	 * @brief Push PCM audio frame data for mixing
-	 * @param buffer Refer to [Browser documentation](https://developer.mozilla.org/en-US/docs/Web/API/AudioBuffer)
-	 * @notes Before calling this method, you must start the external audio stream mixing through [enableAudioMixingFrame](#enableaudiomixingframe).
-	 * @region Advanced Feature
+	 * @brief The user publishes a public stream before joining room.
 	 */
-	pushAudioMixingBuffer: (buffer: AudioBuffer) => void;
-}
-
-/** 
- * @brief Mix playback channel type
- */
-export declare enum AudioMixingDualMonoMode {
+	START_PUBLIC_STREAM_BEFORE_JOIN = "START_PUBLIC_STREAM_BEFORE_JOIN",
 	
 	/** 
-	 * @brief Consistent with audio files
+	 * @brief Reconnection with the server fails and SDK will no longer automatically retry.
+	 *        Please leave room first and join room again, or contact technical support.
 	 */
-	AUTO = 0,
+	RECONNECT_FAILED = "RECONNECT_FAILED",
 	
 	/** 
-	 * @brief Only the left channel audio in the audio file can be heard
+	 * @brief Subtitles are already on. There is no need to call [startSubtitle](104478#startsubtitle) again.
 	 */
-	MODE_L = 1,
+	SUBTITLE_ALREADY_ON = "SUBTITLE_ALREADY_ON",
 	
 	/** 
-	 * @brief Only the right channel audio in the audio file can be heard
+	 * @hidden currently not available
+	 * @brief The substitling task didn't start, or call an API to update subtitles without turning on subtitles.
 	 */
-	MODE_R = 2,
+	SUBTITLE_NOT_TURNED_ON = "SUBTITLE_NOT_TURNED_ON",
 	
 	/** 
-	 * @brief Can hear the left and right audio channels in the audio file at the same time
+	 * @brief Errors occurred concerning cloud media processing. Please contact the technical support.
 	 */
-	MODE_MIX = 3
-}
-
-/** 
- * @brief Mixing type
- */
-export declare enum AudioMixingType {
+	SUBTITLE_ERR_POSTPROCESS = "SUBTITLE_ERR_POSTPROCESS",
 	
 	/** 
-	 * @brief Played at the local device only
+	 * @brief Failed to connect to the thrid-party service. Please contact the technical support.
 	 */
-	PLAYOUT = 0,
+	SUBTITLE_ERR_CONNECTION_ERROR = "SUBTITLE_ERR_CONNECTION_ERROR",
 	
 	/** 
-	 * @brief Sent to the remote devices only
+	 * @brief Cloud media processing is timeout. Please contact the technical support.
 	 */
-	PUBLISH = 1,
+	SUBTITLE_ERR_PROCESS_ERROR = "SUBTITLE_ERR_PROCESS_ERROR",
 	
 	/** 
-	 * @brief Played at the local device and sent to the remote devices.
+	 * @brief The client side failed to identity error codes sent by cloud media processing. Please contact the technical support.
 	 */
-	PLAYOUT_AND_PUBLISH = 2
+	SUBTITLE_ERR_UNKNOWN = "SUBTITLE_ERR_UNKNOWN"
 }
-declare enum AudioMixingState {
-	
-	AUDIO_MIXING_STATE_PRELOADED = 0,
-	
-	AUDIO_MIXING_STATE_PLAYING = 1,
+declare enum AudioMixingError {
 	
-	AUDIO_MIXING_STATE_PAUSED = 2,
+	/** 
+	 * @brief Normal
+	 */
+	AUDIO_MIXING_ERROR_OK = 0,
 	
-	AUDIO_MIXING_STATE_STOPPED = 3,
+	/** 
+	 * @brief Pre-load fails because of missing audio file or the length of the audio file exceeding 20s.
+	 */
+	AUDIO_MIXING_ERROR_PRELOAD_FAILED = 1,
 	
-	AUDIO_MIXING_STATE_FAILED = 4,
+	/** 
+	 * @brief Audio mixing starts fails because of missing audio file or failure of reading the audio file.
+	 */
+	AUDIO_MIXING_ERROR_START_FAILED = 2,
 	
-	AUDIO_MIXING_STATE_FINISHED = 5,
+	/** 
+	 * @brief Invalid audio mixing ID
+	 */
+	AUDIO_MIXING_ERROR_ID_NOT_FOUND = 3,
 	
-	AUDIO_MIXING_STATE_PCM_ENABLED = 6,
+	/** 
+	 * @brief Invalid start position
+	 */
+	AUDIO_MIXING_ERROR_SET_POSITION_FAILED = 4,
 	
-	AUDIO_MIXING_STATE_PCM_DISABLED = 7
-}
-
-/** 
- * @brief Mixing configuration
- */
-export interface MixOption {
+	/** 
+	 * @brief Invalid volume argument. The volume value ranges [0, 400].
+	 */
+	AUDIO_MIXING_ERROR_INVALID_VOLUME = 5,
 	
 	/** 
-	 * @brief Mix playback times
-	 * @notes + Play_count < = 0: Infinite loop
-	 *        + Play_count == 1: Play once (default)
-	 *        + Play_count > 1: Play play_count times
+	 * @brief The audio file to be played does not match to the preload one. Call `unloadAudioMixing` to unload file and proceed.
 	 */
-	playCount: number;
+	AUDIO_MIXING_ERROR_LOAD_CONFLICT = 6,
 	
 	/** 
-	 * @brief For mixing playback types.
+	 * @brief Type of the audio file to be played is not supported.
 	 */
-	type?: AudioMixingType;
-}
-declare class EnhancedEventEmitter<EventTypes extends ValidEventTypes = string | symbol> extends EventEmitter<EventTypes> {
-	safeEmit<T extends EventNames<EventTypes>>(event: T, ...args: EventArgs<EventTypes, T>): boolean;
-}
-
-/** 
- * @brief Information and state of a media device.
- */
-export interface DeviceInfo {
+	AUDIO_MIXING_ERROR_ID_TYPE_NOT_MATCH = 7,
 	
 	/** 
-	 * @brief Information of a media device. See [MediaDeviceInfo](https://developer.mozilla.org/en-US/docs/Web/API/MediaDeviceInfo).
+	 * @brief Invalid audio pitch
 	 */
-	mediaDeviceInfo: MediaDeviceInfo;
+	AUDIO_MIXING_ERROR_ID_TYPE_INVALID_PITCH = 8,
 	
 	/** 
-	 * @brief Device state.
+	 * @brief Invalid audio track
 	 */
-	deviceState: "active" | "inactive";
-}
-/**
- * {zh}
- * 音频编码参数细节
- */
-export interface AudioCodecProfile {
-	chn: number;
-	sr: number;
-}
-export interface IEngineBAC {
-	codec?: {
-		enc_fs?: number;
-	};
-	adm?: {
-		play?: AudioCodecProfile;
-		record?: AudioCodecProfile;
-	};
-	agc?: any;
-	aec?: any;
-	ans?: any;
-}
-export interface IEngineANM {
-	enc_param?: {
-		bitrate: number;
-		chn: number;
-		sr: number;
-	};
-}
-export interface IAudioConfig {
-	bitrate?: number;
-	encChannel?: number;
-	channel?: number;
-	playChannel?: number;
-	sampleRate?: number;
-	encSampleRate?: number;
-	playbackRate?: number;
-	agc?: boolean;
-	aec?: boolean;
-	ans?: boolean;
+	AUDIO_MIXING_ERROR_INVALID_AUDIO_TRACK = 9
 }
 declare class AudioProfileManager {
-	engineBAC: Partial<IEngineBAC>;
-	engineANM: IEngineANM;
-	bitrate?: number;
-	sampleRate?: number;
-	channel?: number;
-	aec?: boolean;
-	agc?: boolean;
-	ans?: boolean;
-	constructor();
-	private _init;
-	updateEngineBAC(newProfile?: IEngineBAC): void;
-	updateEngineANM(newProfile?: IEngineANM): void;
+	private _appId;
+	private _roomId?;
+	private _constraints;
+	private _profile?;
+	constructor(_appId: string);
+	setRoomId(roomId?: string): 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;
+	getOpusConfigStr(conf?: string): string;
+	updateConstraints(conf: MediaTrackConstraints): void;
+	getConstraints(): MediaTrackConstraints;
+	private _getConfigByAudioProfile;
 }
 /** 
  * @brief Cloud proxy informatio
@@ -763,12 +782,13 @@ export interface CloudProxyInfo {
 	configProxy: string;
 }
 export interface RTCContext {
+	id: string;
 	audioProfileManager?: AudioProfileManager;
 	serverConfig?: {
 		videoCodec?: string;
-		audioRed: boolean;
-		muteReplaceUnsub: boolean;
-		simulcastOnDemand: boolean;
+		audioRed?: boolean;
+		muteReplaceUnsub?: boolean;
+		simulcastOnDemand?: boolean;
 	};
 	streamRTT?: {
 		[streamId: string]: {
@@ -776,11 +796,16 @@ export interface RTCContext {
 			video: number;
 		};
 	};
+	activeSimStreams?: boolean[];
 	subscribeFallbackOption?: SubscribeFallbackOption;
 	userPriority?: Map<string, RemoteUserPriority>;
 	avSync: boolean;
 }
 export declare type RTCExtension = any;
+/**
+ * 设置到pc上的视频编码参数
+ */
+export declare type VideoEncoderConfig_PC = Pick<VideoEncoderConfig, "maxKbps">;
 declare enum StreamMixingEventType {
 	
 	/** 
@@ -1160,7 +1185,9 @@ export interface LocalVideoStats {
 	encodedFrameCount: number;
 	
 	/** 
-	 * @brief Encoding type
+	 * @brief Video codec.
+	 *        "H264": H.264 video codec.
+	 *        "VP8": VP8 video codec.
 	 */
 	codecType: string;
 	
@@ -1611,6 +1638,8 @@ declare enum EngineEventsTypes {
 	onVideoStreamBanned = "onVideoStreamBanned",
 	onAudioStreamBanned = "onAudioStreamBanned",
 	onLocalVideoSizeChanged = "onLocalVideoSizeChanged",
+	onSubtitleStateChanged = "onSubtitleStateChanged",
+	onSubtitleMessageReceived = "onSubtitleMessageReceived",
 	onServerParamsSetResult = "onServerParamsSetResult"
 }
 
@@ -1798,7 +1827,6 @@ export interface LocalAudioPropertiesInfo {
 	streamIndex: StreamIndex;
 	
 	/** 
-	 * @type keytype
 	 * @brief Audio information
 	 */
 	audioPropertiesInfo: AudioPropertiesInfo;
@@ -1926,8 +1954,8 @@ declare enum StreamMixingEventErrorCode {
 
 /** 
  * @id EngineEvents
- * @list 104477
- * @detail 104479
+ * @list overview
+ * @detail callback
  */
 export interface IEngineEvents {
 	
@@ -1961,7 +1989,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Callback on new media streams captured by camera/microphone in the room.
-	 * @notes You will receive this callback after a remote user successfully publishes media streams captured by camera/microphone in the room with [publishStream](#publishstream). Then you can choose whether to call [subscribeStream](#subscribeStream) to subscribe to the streams or not.
+	 * @notes You will receive this callback after a remote user successfully publishes media streams captured by camera/microphone in the room with [publishStream](104478#publishstream). Then you can choose whether to call [subscribeStream](104478#subscribestream) to subscribe to the streams or not.
 	 */
 	[EngineEventsTypes.onUserPublishStream]: (event: {
 		
@@ -1978,7 +2006,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Callback on removal of remote media stream captured by camera/microphone.
-	 * @notes You will receive this callback after a remote user successfully unpublishes media streams captured by camera/microphone in the room with [publishStream](#publishstream). Then you can choose whether to call [unsubscribeStream](#unsubscribeStream) to unsubscribe from the streams or not.
+	 * @notes You will receive this callback after a remote user successfully unpublishes media streams captured by camera/microphone in the room with [unpublishStream](104478#unpublishstream). Then you can choose whether to call [unsubscribeStream](104478#unsubscribestream) to unsubscribe from the streams or not.
 	 */
 	[EngineEventsTypes.onUserUnpublishStream]: (event: {
 		
@@ -2000,7 +2028,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Callback on new screen sharing media streams from remote users in the room.
-	 * @notes You will receive this callback after a remote user successfully published screen sharing streams in the room with [publishScreen](#publishscreen). Then you can choose whether to call [subscribeScreen](#subscribeScreen) to subscribe to the streams or not.
+	 * @notes You will receive this callback after a remote user successfully published screen sharing streams in the room with [publishScreen](104478#publishscreen). Then you can choose whether to call [subscribeScreen](104478#subscribescreen) to subscribe to the streams or not.
 	 */
 	[EngineEventsTypes.onUserPublishScreen]: (event: {
 		
@@ -2017,7 +2045,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Callback on removal of screen sharing media streams from remote users in the room.
-	 * @notes You will receive this callback after a remote user successfully unpublishes media streams captured by camera/microphone in the room with [publishScreen](#publishscreen). Then you can choose whether to call [unsubscribeScreen](#unsubscribeScreen) to unsubscribe from the streams or not.
+	 * @notes You will receive this callback after a remote user successfully unpublishes media streams captured by camera/microphone in the room with [unpublishScreen](104478#unpublishscreen). Then you can choose whether to call [unsubscribeScreen](104478#unsubscribescreen) to unsubscribe from the streams or not.
 	 */
 	[EngineEventsTypes.onUserUnpublishScreen]: (event: {
 		
@@ -2152,7 +2180,7 @@ export interface IEngineEvents {
 	}) => void;
 	
 	/** 
-	 * @brief Receiving a point-to-point text message from one of the users in the room by calling [sendUserMessage](104478#sendUserMessage)
+	 * @brief Receiving a point-to-point text message from one of the users in the room by calling [sendUserMessage](104478#sendusermessage)
 	 * @param event
 	 * @region Message Event
 	 */
@@ -2203,7 +2231,7 @@ export interface IEngineEvents {
 	}) => void;
 	
 	/** 
-	 * @brief After calling [enableAudioPropertiesReport](104478#enableAudioPropertiesReport), you will periodically receive this callback for the information about local audio.
+	 * @brief After calling [enableAudioPropertiesReport](104478#enableaudiopropertiesreport), you will periodically receive this callback for the information about local audio.
 	 *        Local audio includes the microphone audio and the screen audio captured using RTC SDK internal mechanisms.
 	 * @param event Information about local audio.
 	 * @region Audio Managment
@@ -2211,7 +2239,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onLocalAudioPropertiesReport]: (event: LocalAudioPropertiesInfo[]) => void;
 	
 	/** 
-	 * @brief After calling [enableAudioPropertiesReport](104478#enableAudioPropertiesReport), you will periodically receive this callback for the information about the subscribed remote audio streams.
+	 * @brief After calling [enableAudioPropertiesReport](104478#enableaudiopropertiesreport), you will periodically receive this callback for the information about the subscribed remote audio streams.
 	 *        The remote audio streams includes the microphone audio and screen audio collected using the RTC SDK internal mechanism/custom mechanism.
 	 * @param event Information about remote audio.
 	 * @region Audio Management
@@ -2232,7 +2260,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onAudioPlaybackDeviceChanged]: (event: MediaDeviceInfo) => void;
 	
 	/** 
-	 * @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).
+	 * @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
 	 * @region Audio Management Event
 	 */
@@ -2245,7 +2273,7 @@ export interface IEngineEvents {
 	}) => 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).
+	 * @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
 	 * @region Room Management Event
 	 */
@@ -2258,7 +2286,7 @@ export interface IEngineEvents {
 	}) => 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).
+	 * @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
 	 * @region Room Management Event
 	 */
@@ -2271,7 +2299,7 @@ export interface IEngineEvents {
 	}) => 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).
+	 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
 	 * @region Room Management Event
 	 */
@@ -2388,7 +2416,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onTokenWillExpire]: () => void;
 	
 	/** 
-	 * @brief Receives the callback when you call [startCloudProxy](104478#startCloudProxy) to start cloud proxy, and the SDK connects the proxy server successfully.
+	 * @brief Receives the callback when you call [startCloudProxy](104478#startcloudproxy) to start cloud proxy, and the SDK connects the proxy server successfully.
 	 * @param event
 	 */
 	[EngineEventsTypes.onCloudProxyConnected]: (event: CloudProxyConnectedEvent) => void;
@@ -2538,19 +2566,36 @@ export interface IEngineEvents {
 		};
 	}) => void;
 	
+	/** 
+	 * @hidden currently not available
+	 * @brief  Callback on subtitle states.
+	 *         After you call [startSubtitle](104478#startsubtitle) and [stopSubtitle](104478#stopsubtitle), you will receive this callback which informs you of the states and error codes of the subtitling task, as well as detailed information on the third party services' errors.
+	 * @param event
+	 */
+	[EngineEventsTypes.onSubtitleStateChanged]: (e: SubtitleEvent) => void;
+	
+	/** 
+	 * @hidden currently not available
+	 * @brief  Callback on subtitle messages.
+	 *         After calling [startSubtitle](104478#startsubtitle) successfully, you will receive this callback which informs you of the related information on subtitles.
+	 * @param event
+	 */
+	[EngineEventsTypes.onSubtitleMessageReceived]: (e: SubtitleMessage[]) => void;
+	
 	/** 
 	 * @brief  Event of whether the setting for the message server successes
 	 * @param error Error code
-	 * @notes + This event notifies you the result after the call of [setServerParams](104478#setServerParams).
-	 * + If the client losts connection after the call of [setServerParams](104478#setServerParams) this event notifies once the client reconnects.
+	 * @notes + This event notifies you the result after the call of [setServerParams](104478#setserverparams).
+	 * + If the client losts connection after the call of [setServerParams](104478#setserverparams) this event notifies once the client reconnects.
 	 */
 	[EngineEventsTypes.onServerParamsSetResult]: (error?: ErrorCode) => void;
 }
 
 /** 
  * @id RTCEngine
- * @list 104477
- * @detail 104478
+ *     <span id="irtcengine"></span>
+ * @list overview
+ * @detail api
  */
 export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
@@ -2558,7 +2603,7 @@ 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.
 	 * @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.
+	 *         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](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 ID of the room to be added.
@@ -2572,10 +2617,11 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return
 	 * @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. The remote user will receive [onUserJoined](104479#onuserjoined) if a visible user reconnected successfully.
+	 *        + 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.
 	 * @region Room Management
 	 */
 	joinRoom: (token: string | null, roomId: string, userInfo: UserInfo, roomConfig?: RoomConfig) => Promise<void>;
+	setConnectionMode: (mode: ConnectionMode) => void;
 	
 	/** 
 	 * @brief Leave the room.
@@ -2589,7 +2635,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @hidden
 	 * @brief Destroy engine
-	 * Destroy the engine instances created by calling [createEngine](#createengine) and release all related resources.
+	 * Destroy the engine instances created by calling [createEngine](104478#createengine) and release all related resources.
 	 * @notes + Make sure that the business scenarios related to the `RTCEngine` instances that need to be destroyed are all finished before calling this API.
 	 *       + Calling this API will destroy all memory related to this `RTCEngine` instance, and stop any interaction with the media server.
 	 *       + This API is a blocking call which will block the current thread until the SDK completes the exit logic. Therefore, do NOT to call this API directly in the callback thread. Besides do NOT wait for the execution of the main thread including this API in any callback, thus causing a deadlock.
@@ -2600,8 +2646,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @brief Publishes media streams captured by camera/microphone in the current room.
 	 * @param mediaType Media stream type, used for specifying whether to publish audio stream or video stream.
-	 * @notes + Streams are not published automatically in multi-room mode by default, you need to call this API to publish streams manually.
-	 *        + An invisible user cannot publish media streams. Call [setUserVisibility](#setuservisibility) to change your visibility in the room.
+	 * @notes + An invisible user cannot publish media streams. Call [setUserVisibility](#setuservisibility) to change your visibility in the room.
 	 *        + Call [publishScreen](#publishscreen) to start screen sharing.
 	 *        + After you call this API, the other users in the room will receive [onUserPublishStream](104479#onuserpublishstream).
 	 *        + Call [unpublishStream](#unpublishstream) to stop publishing streams.
@@ -2707,15 +2752,13 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Enable internal video capture. The default state is off.
-	 * Internal video capture refers to: capturing video using the built-in module.
-	 * The remote clients in the room will be informed of the state change via [onUserStartVideoCapture](104479#onuserstartvideocapture) after the publisher starts video capture by calling this API.
+	 *        Internal video capture refers to: capturing video using the built-in module.
+	 *        The remote clients in the room will be informed of the state change via [onUserStartVideoCapture](104479#onuserstartvideocapture) after the publisher starts video capture by calling this API.
 	 * @param deviceId ID of the capture device which you need to pass in to avoid exceptions such as no screen or a screen that is not the desired one. You can call [enumerateDevices](#enumeratedevices) to obtain the list of devices.
 	 * @return The actual effective video capture parameters. See [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints). Pay attention to the properties: `frameRate`, `height`, and `width`.
-	 *         The `frameRate`, `height`, `width` set in [setVideoEncoderConfig](104478#setvideoencoderconfig) for the max stream must match this return value accordingly.
 	 * @notes + Call [stopVideoCapture](#stopvideocapture) to stop the internal video capture. Otherwise, the internal video capture will sustain until you destroy the engine instance.
-	 *     + Once you create the engine instance, you can start internal video capture regardless of the video publishing state. The video stream will start publishing only after the video capture starts.
-	 *     + Call [setVideoCaptureDevice](#setvideocapturedevice) to switch the camera used by the internal video capture module.
-	 * @region Video Management
+	 *        + Once you create the engine instance, you can start internal video capture regardless of the video publishing state. The video stream will start publishing only after the video capture starts.
+	 *        + Call [setVideoCaptureDevice](#setvideocapturedevice) to switch the camera used by the internal video capture module.
 	 */
 	startVideoCapture: (deviceId?: string) => Promise<MediaTrackSettings>;
 	
@@ -2757,8 +2800,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @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.
 	 * @return
-	 * @notes + Call this API will start screen stream video capture but will not publish the captured video. Publishing a screen-sharing stream by calling [publishScreen](#publishScreen).
-	 *     + To turn off screen video source capture, call [stopScreenVideoCapture](#stopScreenVideoCapture).
+	 * @notes + Call this API will start screen stream video capture but will not publish the captured video. Publishing a screen-sharing stream by calling [publishScreen](#publishscreen).
+	 *     + To turn off screen video source capture, call [stopScreenCapture](#stopscreencapture).
 	 *     + Before calling this API, you can call [setVideoEncoderConfig](#setvideoencoderconfig) to set the capture frame rate and encoding resolution of the screen video stream.
 	 * @region Advanced Feature
 	 */
@@ -2767,7 +2810,6 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @brief Stop screen sharing.
 	 * @return
-	 * @notes If the stream has been published, the other users in the room will receive the [onStreamRemove](104479#onstreamremove) event after the user stop screen sharing by calling this API.
 	 * @region Advanced Feature
 	 */
 	stopScreenCapture: () => Promise<void>;
@@ -2785,7 +2827,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @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) is triggered.
+	 *     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
@@ -2812,7 +2854,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Play audio and video stream manually.
-	 * Call this API when user need to take part in the audio and video reaction after calling [createEngine](#createengine) and set to autoPlay to False.
+	 * Call this API when user need to take part in the audio and video reaction after calling [createEngine](104478#createengine) and set to autoPlay to False.
 	 * @param userId ID of the user whose stream is intended to play
 	 *               Call this API without a parameter to start playing the local medias.
 	 * @param mediaType Media stream type, used for specifying whether to publish audio stream or video stream.
@@ -2830,7 +2872,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return
 	 * @notes + Calling this API does not change the capture state and the transmission state of the remote clients.
 	 *         + Calling this API does not cancel the subscription or change any subscription configuration.
-	 *         + To resume, call [resumeAllSubscribedStream](#resumeAllSubscribedStream).
+	 *         + To resume, call [resumeAllSubscribedStream](#resumeallsubscribedstream).
 	 * @region Media Stream Transmission
 	 */
 	pauseAllSubscribedStream: (mediaType: MediaType) => Promise<void>;
@@ -2904,29 +2946,28 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        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 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 + The `frameRate`, `height`, `width` set by the max stream must match the return value of [startVideoCapture](104478#startvideocapture).
-	 *        + 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.
+	 * @notes + The default parameters are 640 × 480 px, 15 FPS, with 600 Kbps max bitrate.
+	 *        + You can call this API before and after publishing video streams. After the video stream is published, you cannot switch from single stream to multi-stream, or modify the number of published multi-stream layers.
+	 *        + When simulcasting, it is recommended to cancel automatic publishing and publish audio/video streams manually by calling [publishStream](#publishstream), otherwise simulcast will not take effect.
+	 *        + If the set parameters are not supported by the browser, RTC SDK will capture and encode according to the range of parameters supported by the browser, and report the actual parameters of the capture by [onLocalVideoSizeChanged](104479#onlocalvideosizechanged).
+	 *        + You need to call [enableSimulcastMode](#enablesimulcastmode) to start simulcasting. If you set the parameters for multiple streams without calling [enableSimulcastMode](#enablesimulcastmode), only the stream with the largest resolution will be published, and the setting will take effect after simulcasting is enabled. If simulcasting is enabled, but only one stream's parameters are set, RTC SDK will automatically fill 1~2 groups of smaller stream parameters.
 	 *        + This API is applicable to the video stream captured by the camera, see [setScreenEncoderConfig](#setscreenencoderconfig) for setting parameters for screen sharing video stream.
-	 * @region Video Management
 	 */
 	setVideoEncoderConfig: (descriptions: VideoEncoderConfig | VideoEncoderConfig[]) => Promise<void>;
 	
 	/** 
-	 * @type api
-	 * @author changzhenan
-	 * @region Video Management
 	 * @brief Set the encoding configuration of the screen-sharing stream，which is also applied in video capture.
 	 *        If the configuration is not supported by your browser, the video will be captured based on the supported configuration. You will also receive the callback [onLocalVideoSizeChanged](104479#onlocalvideosizechanged).
-	 * @param VideoEncoderConfig The encoding configuration of the screen-sharing stream.
-	 * @notes + We recommend that you set the encoding configuration before video capture. Otherwise, the video will be captured based on the default configuration(1080p@15fps).
+	 * @param description The encoding configuration of the screen-sharing stream.
+	 * @notes + The default parameters are 1920 × 1080 px, 15 FPS, with 3000 Kbps max bitrate.
+	 *        + We recommend that you set the encoding configuration before video capture. Otherwise, the video will be captured based on the default configuration.
 	 */
 	setScreenEncoderConfig: (description: ScreenEncoderConfig) => Promise<void>;
 	
 	/** 
 	 * @brief Sets your expected configuration of the remote video stream that you want to subscribe to or have subscribed to.
-	 * @param userId The ID of the target stream publisher
-	 * @param remoteVideoConfig Your expected configuration.
+	 * @param userId ID of the remote video stream publisher you expect to configure subscription parameters for.
+	 * @param remoteVideoConfig The parameters you expect to configure for the remote video stream.
 	 * @notes + This API only works after the publisher calls [enableSimulcastMode](#enablesimulcastmode) to enable publishing multiple video streams, in which case the subscriber will receive a stream that is closest to the set configuration.
 	 *        + If you don't call this API after the publisher enables the function of publishing multiple streams, you will receive by default the video stream with the largest resolution set by the publisher.
 	 *        + You should call this API in the room. If you want to call it before entering the room, you should set the `remoteVideoConfig` in the roomConfig` when calling [joinRoom](#joinroom).
@@ -2939,7 +2980,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Enables the simulcast mode of the video stream captured by the camera at the publishing end.imulcasting, including resolution, frame rate, bitrate, scale mode, fallback strategy under poor network conditions.
 	 * @param enabled
 	 * @param + You must call this API before publishing the video streams.
-	 *    + After enabling the simulcast mode, SDK automatically sets the encoding parameters for the video streams. For manual setting, call [setVideoEncoderConfig](#setVideoEncoderConfig).
+	 *    + After enabling the simulcast mode, SDK automatically sets the encoding parameters for the video streams. For manual setting, call [setVideoEncoderConfig](#setvideoencoderconfig).
 	 */
 	enableSimulcastMode: (enabled: boolean) => boolean;
 	
@@ -2961,8 +3002,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Enable audio information prompts.
 	 * @param config Audio information prompts-related configurations.
 	 * @notes With the prompt on, you can:
-	 *       + Get the information of the audio stream captured by the local microphone and screen audio stream through [onLocalAudioPropertiesReport](104479#onLocalAudioPropertiesReport).
-	 *       + Get the information of the subscribed remote audio streams through [onRemoteAudioPropertiesReport](104479#onRemoteAudioPropertiesReport).
+	 *       + Get the information of the audio stream captured by the local microphone and screen audio stream through [onLocalAudioPropertiesReport](104479#onlocalaudiopropertiesreport).
+	 *       + Get the information of the subscribed remote audio streams through [onRemoteAudioPropertiesReport](104479#onremoteaudiopropertiesreport).
 	 * @region Audio Management
 	 */
 	enableAudioPropertiesReport: (config?: AudioPropertiesConfig) => void;
@@ -2979,7 +3020,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Update configuration needed when pushing media streams to CDN.
-	 *        After calling [startLiveTranscoding](#startlivetranscoding) to enable the function of pushing streams to CDN, you can call this API to update the relevant configurations.
+	 *        After calling [startLiveTranscoding](104478#startlivetranscoding) to enable the function of pushing streams to CDN, you can call this API to update the relevant configurations.
 	 * @param transcode Updated configurations. Some of the configurations cannot be updated. See [LiveTranscodeConfig](104481#livetranscodeconfig) for specific indications.
 	 * @return
 	 */
@@ -2987,7 +3028,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Stop pushing media streams to CDN
-	 * For starting pushing media streams to CDN. Refer to [startLiveTranscoding](#startlivetranscoding).
+	 * For starting pushing media streams to CDN. Refer to [startLiveTranscoding](104478#startlivetranscoding).
 	 * @return
 	 * @region Advanced Feature
 	 */
@@ -3004,7 +3045,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	setBusinessId: (businessId: string) => boolean;
 	
 	/** 
-	 * @brief Set user visibility. Visible by default.
+	 * @brief Set user visibility. The local user is visible to others by default before calling this API.
 	 * @param enable  Visibility of the user in the room:
 	 *         + `true`: The user can publish media streams. And the other users in the room get informed of the behaviors of the user, such as Joining room, starting video capture, and Leaving room.
 	 *         + `false`: The user cannot publish media streams. And the other users in the room do not get informed of the behaviors of the user, such as joining, starting video capture, or leaving.
@@ -3019,7 +3060,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Set a custom videoTrack as the video source.
-	 *        Before calling this API, you must call [setVideoSourceType](#setvideosourcetype) to enable custom video capture.
+	 *        Before calling this API, you must call [setVideoSourceType](104478#setvideosourcetype) to enable custom video capture.
 	 * @param index specify a stream index.
 	 * @param track specify a video track. See [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack).
 	 * @return
@@ -3029,7 +3070,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Set a custom audioTrack as the audio source.
-	 *        Before calling this API, you must call [setAudioSourceType](#setaudiosourcetype) to enable custom video capture.
+	 *        Before calling this API, you must call [setAudioSourceType](104478#setaudiosourcetype) to enable custom video capture.
 	 * @param index specify a stream index.
 	 * @param track specify an audio track. See [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack).
 	 * @return
@@ -3053,7 +3094,6 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param index audio stream index
 	 * @param audioSourceType Type of the module for video capture
 	 * @return
-	 * @region Audio Management
 	 */
 	setAudioSourceType: (index: StreamIndex, audioSourceType: AudioSourceType) => Promise<void>;
 	
@@ -3068,16 +3108,19 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @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.
 	 * @return
-	 * @note You can call this API after creating the engine. We recommend to call this API before [startAudioCapture](#startaudiocapture).
+	 * @notes You should call this API before starting audio capture and publishing streams.
 	 */
 	setAudioCaptureConfig: (config: TrackCaptureConfig) => Promise<void>;
 	
 	/** 
+	 * @hidden
+	 * @deprecated since v4.51, use setVideoEncoderConfig instead.
 	 * @brief Configure the internal video capture. Different browser has its own default settings.
 	 * @param config Configuration for internal video capture. See [MediaTrackConstraints](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints).
 	 *               The input `frameRate`, `height`, `width` are only desired values. The true resolution and framerate of the internal capture are the return values of [startVideoCapture](104478#startvideocapture).
 	 * @return
-	 * @note You can call this API after creating the engine. We recommend to call this API before [startVideoCapture](#startvideocapture).
+	 * @note + You can call this API after creating the engine. We recommend to call this API before [startVideoCapture](#startvideocapture).
+	 *       + Since v4.51, the scaling ratio of capturing resolution to encoding resolution is fixed to 1. When the capturing resolution is modified, the actual sending resolution will be changed at the same time.
 	 */
 	setVideoCaptureConfig: (config: TrackCaptureConfig) => Promise<MediaTrackSettings>;
 	
@@ -3090,7 +3133,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param userInfo  User information.
 	 * @return
 	 * @notes
-	 *        + Call [logout](104478@#logout) to log out the IM server.
+	 *        + Call [logout](#logout) to log out the IM server.
 	 *        + In the same App ID, the user ID of each user must be unique. The user will be logged out when a user with the same userID joins the room because of the error of `RTM_DUPLICATE_LOGIN`.
 	 * @region Room Management
 	 */
@@ -3115,7 +3158,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Get the status of a user before sending him/her a P2P message. You can also call this API to check the status of the current user.
 	 * @param userId ID of the user whose status intended to know
 	 * @return
-	 * @notes Call [login](104478@#login) before calling this API.
+	 * @notes Call [login](#login) before calling this API.
 	 */
 	getPeerOnlineStatus: (userId: string) => Promise<USER_ONLINE_STATUS>;
 	
@@ -3124,8 +3167,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param userId ID of the recipient
 	 * @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.
+	 * @notes + Call [login](#login) before calling this API.
+	 * + The recipient must be logged into the server and will receive the text message via [onUserMessageReceivedOutsideRoom](104479#onusermessagereceivedoutsideroom) if the message has been sent successfully.
 	 */
 	sendUserMessageOutsideRoom: (userId: string, message: string) => Promise<any>;
 	
@@ -3140,7 +3183,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	sendUserBinaryMessageOutsideRoom: (userId: string, message: ArrayBuffer) => Promise<any>;
 	
 	/** 
-	 * @brief Set the params of the server before calling [sendServerMessage](#sendServerMessage) or [sendServerBinaryMessage](#sendserverbinarymessage) to send messages to the server.
+	 * @brief Set the params of the server before calling [sendServerMessage](104478#sendservermessage) or [sendServerBinaryMessage](104478#sendserverbinarymessage) to send messages to the server.
 	 * @param signature Dynamic signature.
 	 * @param url Address of the IM server
 	 * @return
@@ -3200,7 +3243,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @hidden
 	 * @brief Update the properties of the public stream published by the current user.
-	 *        Call [startPushPublicStream](#startpushpublicstream) to start publishing a public stream.
+	 *        Call [startPushPublicStream](104478#startpushpublicstream) to start publishing a public stream.
 	 * @param publicStreamId The public stream's ID.
 	 *                       The public stream must be published by the current user.
 	 * @param publicStreamParam The public stream's parameters.
@@ -3212,7 +3255,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @hidden
 	 * @type api
 	 * @brief Stops publishing the public stream.
-	 *        Call [startPushPublicStream](#startpushpublicstream) to start publishing a public stream.
+	 *        Call [startPushPublicStream](104478#startpushpublicstream) to start publishing a public stream.
 	 * @param publicStreamId The public stream's ID.
 	 *                       The public stream must be published by the current user.
 	 * @return
@@ -3228,6 +3271,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return
 	 * @notes + Before calling this API, we recommend that you call [setPublicStreamVideoPlayer](#setpublicstreamvideoplayer) first to set the player information and rendering mode for the public stream.
 	 *        + After calling this API, you will receive [onFirstPublicStreamVideoFrameDecoded](104479#onfirstpublicstreamvideoframedecoded) and [onFirstPublicStreamAudioFrameDecoded](104479#onfirstpublicstreamaudioframedecoded) once the first frame has been decoded.
+	 *        + If the public stream contains SEI message, you will receive the message by be informed through [onPublicStreamSEIMessageReceived](104479#onpublicstreamseimessagereceived).
 	 *        + Call [stopPlayPublicStream](#stopplaypublicstream) to unsubscribing from the public stream.
 	 */
 	startPlayPublicStream: (publicStreamId: string) => Promise<void>;
@@ -3235,7 +3279,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @hidden
 	 * @brief Unsubscribes from a designated public stream.
-	 *        Call [startPlayPublicStream](#startplaypublicstream) to subscribe to the public stream.
+	 *        Call [startPlayPublicStream](104478#startplaypublicstream) to subscribe to the public stream.
 	 * @param publicStreamId The public stream's ID.
 	 * @return
 	 */
@@ -3282,7 +3326,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Set the priority of the remote user's media stream.
-	 *        Use this API with [setSubscribeFallbackOption](#setsubscribefallbackoption).
+	 *        Use this API with [setSubscribeFallbackOption](104478#setsubscribefallbackoption).
 	 *        With video stream fallback enabled and user priority specified, the stream quality of higher priority user is the last to be compromised.
 	 * @param userId The ID of the remote user
 	 * @param priority The priority of the remote user's media stream
@@ -3309,12 +3353,12 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Start the capture and playback test for local audio devices.
-	 * We recommend to register [onLocalAudioPropertiesReport](104479#onlocalaudiopropertiesreport) callback via [enableAudioPropertiesReport](enableaudiopropertiesreport). Once the test starts, you can get the volume of the capturing audio by this callback.
+	 * We recommend to register [onLocalAudioPropertiesReport](104479#onlocalaudiopropertiesreport) callback via [enableAudioPropertiesReport](104478#enableaudiopropertiesreport). Once the test starts, you can get the volume of the capturing audio by this callback.
 	 * @param indicationInterval The time interval between each callback of [onAudioPlaybackDeviceTestVolume](104479#onaudioplaybackdevicetestvolume) via which you get informed of the playback volume during the test.  We recommend setting it to 200 ms. The minimal value is 100 ms.
 	 * @param onAutoplayFailed Autoplay fails due to the limitation of the browser. We recommend to execute `resume` as soon as the user clicks on the UI element.
 	 * @return Resolve when the test ends.
 	 * @notes + You can call this API whether the user is in a room.
-	 * + The audio capturing will stop and begins to play the recording in 30 s after calling this API. Before that, call [stopAudioDeviceRecordAndPlayTest](#stopaudiodevicerecordandplayTest) to stop capturing and start playing the recording.
+	 * + The audio capturing will stop and begins to play the recording in 30 s after calling this API. Before that, call [stopAudioDeviceRecordAndPlayTest](#stopaudiodevicerecordandplaytest) to stop capturing and start playing the recording.
 	 * + You must stop the test before starting another test for audio devices.
 	 * + Call [stopAudioDevicePlayTest](#stopaudiodeviceplaytest) to stop the test, including capturing and playing the recording.
 	 * + This test performs locally and does not involve network connection testing.
@@ -3323,12 +3367,12 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Call this API to stop recording in the test and start to play the recording.
-	 * @notes After calling this API, the recording starts playing during which you can call [stopAudioDevicePlayTest](#stopaudiodeviceplaytest) to stop playing.
+	 * @notes After calling this API, the recording starts playing during which you can call [stopAudioDevicePlayTest](104478#stopaudiodeviceplaytest) to stop playing.
 	 */
 	stopAudioDeviceRecordAndPlayTest: () => void;
 	
 	/** 
-	 * @brief Stop the capture and playback test for local audio devices which is started by calling [startAudioDeviceRecordTest](#startaudiodevicerecordtest).
+	 * @brief Stop the capture and playback test for local audio devices which is started by calling [startAudioDeviceRecordTest](104478#startaudiodevicerecordtest).
 	 * Before the test ends by itself, you can call this API to stop the recording or playing.
 	 */
 	stopAudioDevicePlayTest: () => void;
@@ -3336,9 +3380,10 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @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.
+	 * @notes You should call this API before starting audio capture and publishing streams.
+	 * @return
 	 */
-	setAudioProfile: (profile: AudioProfileType) => void;
+	setAudioProfile: (profile: AudioProfileType) => Promise<void>;
 	
 	/** 
 	 * @brief Get the [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack) of the local audios or videos.
@@ -3364,11 +3409,11 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @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).
+	 * + 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)
+	 *   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;
 	
@@ -3406,7 +3451,41 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        • [AI Noise-Cancelling Extension](https://docs.byteplus.com/byteplus-rtc/docs/148647).
 	 */
 	registerExtension: (extension: RTCExtension) => Promise<void>;
-	setRemoteStreamRenderSync(sync: boolean): boolean;
+	
+	/** 
+	 * @hidden currently not available
+	 * @brief Turns on subtitles.
+	 *        After calling this method, you can choose between the recognition and translation modes based on your needs. In the recognition mode, the speech of the speaker in the room will be recognized and converted into captions. In the translation mode, the transcribed text will be translated.You will receive the [onSubtitleMessageReceived](104479#onsubtitlemessagereceived) callback to informs you of related information about subtitles.
+	 *        After calling this method, you will receive the [onSubtitleStateChanged](104479#onsubtitlestatechanged) to inform you of whether subtitles are on.
+	 * @param config Subtitle configrations.
+	 * @return +  `0`: Success.
+	 *         + `!0`: Failure.
+	 * @notes
+	 *        You must call this method after joining the room.
+	 *        You can set your source language by calling `joinRoom`  and importing a json formatted string `"SourceLanguage": "zh"` through the paramater of extraInfo. Only chinese, English and Japanese are supported.
+	 *        The language of your browser will be automatically set as the source language if you don't set it.
+	 */
+	startSubtitle: (config: RTCSubtitleConfig) => Promise<void>;
+	
+	/** 
+	 * @hidden currently not available
+	 * @brief Turns off subtitles.
+	 *        After calling this method, you will receive the [onSubtitleStateChanged](104479#onsubtitlestatechanged) to inform you of whether subtitles are off.
+	 * @return  +  `0`: Success.
+	 *          + `!0`: Failure.
+	 */
+	stopSubtitle: () => void;
+	
+	/** 
+	 * @brief Set whether the remote audio and video streams are rendered synchronously.
+	 *        You can achieve ultra low latency by turning off the audio and video synchronization.
+	 * @param sync `true`: (Default) Turn on sync.
+	 *             `false`: Turn off sync.
+	 * @return + `true`: Success.
+	 *         + `false`: Failure.
+	 * @notes You must call this API before joining the room. Calling it afterward is invalid.
+	 */
+	setRemoteStreamRenderSync: (sync: boolean) => boolean;
 }
 
 /** 
@@ -3497,32 +3576,34 @@ export declare enum ChannelProfile {
 }
 
 /** 
- * @brief Room configure
+ * @brief Room configurations.
  */
 export interface RoomConfig {
 	
 	/** 
 	 * @brief Enable/disable auto-subscribe audio. Enable by default.
 	 *        This does not include the screen stream.
-	 *        Call publish[104481#publish] to publish media streams if you disable `isAutoPublish`.
+	 *        Call [publishStream](104478#publishstream) to publish media streams if you disable `isAutoPublish`.
 	 */
 	isAutoPublish?: boolean;
 	
 	/** 
 	 * @brief Enable/disable auto-subscribe audio. Enable by default.
 	 *        This does not include the screen stream.
-	 * Call subscribeUserStream, unsubscribe to revise the subscription.
+	 *        You can call [subscribeStream](104478#subscribestream) and [unsubscribeStream](104478#unsubscribestream) to revise the subscription after entering the room.
 	 */
 	isAutoSubscribeAudio?: boolean;
 	
 	/** 
 	 * @brief Enable/disable auto-subscribe video. Enable by default.
 	 *        This does not include the screen stream.
-	 * Call subscribeUserStream, unsubscribe to revise the subscription.
+	 *        You can call [subscribeStream](104478#subscribestream) and [unsubscribeStream](104478#unsubscribestream) to revise the subscription after entering the room.
 	 */
 	isAutoSubscribeVideo?: boolean;
 	
 	/** 
+	 * @hidden
+	 * @deprecated since 4.51, use setAudioProfile instead.
 	 * @brief  Room profile
 	 *        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.
@@ -3891,22 +3972,25 @@ export interface ScreenConfig {
 export interface VideoEncoderConfig {
 	
 	/** 
-	 * @brief The width of the shared video (px).
+	 * @brief Video width in pixels. The default value is 640 px。
+	 *        The input can be a number or a [ConstrainULong](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints#constrainulong).
 	 */
-	width: number;
+	width: ConstrainULong;
 	
 	/** 
-	 * @brief The height of the shared video (px).
+	 * @brief Video height in pixels. The default value is 480 px。
+	 *        The input can be a number or a [ConstrainULong](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints#constrainulong).
 	 */
-	height: number;
+	height: ConstrainULong;
 	
 	/** 
-	 * @brief The framerate of the shared video (fps).
+	 * @brief Video framerate in FPS. The default value is 15 FPS。
+	 *        The input can be a number or a [ConstrainDouble](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints#constraindouble).
 	 */
-	frameRate: number;
+	frameRate: ConstrainDouble;
 	
 	/** 
-	 * @brief The maximum bitrate(>0/kbps). The default value is 3000kbps.
+	 * @brief The maximum bitrate in Kbps (>0). The default value is 600 Kbps.
 	 */
 	maxKbps: number;
 }
@@ -3917,22 +4001,25 @@ export interface VideoEncoderConfig {
 export interface ScreenEncoderConfig {
 	
 	/** 
-	 * @brief The width of the shared video (px).
+	 * @brief Video width in pixels. The default value is 1920 px。
+	 *        The input can be a number or a [ConstrainULong](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints#constrainulong).
 	 */
-	width: number;
+	width: ConstrainULong;
 	
 	/** 
-	 * @brief The height of the shared video (px).
+	 * @brief Video height in pixels. The default value is 1080 px。
+	 *        The input can be a number or a [ConstrainULong](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints#constrainulong).
 	 */
-	height: number;
+	height: ConstrainULong;
 	
 	/** 
-	 * @brief The framerate of the shared video (fps).
+	 * @brief Video framerate in FPS. The default value is 15 FPS。
+	 *        The input can be a number or a [ConstrainDouble](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints#constraindouble).
 	 */
-	frameRate: number;
+	frameRate: ConstrainDouble;
 	
 	/** 
-	 * @brief The maximum bitrate(>0/kbps). The default value is 3000kbps.
+	 * @brief The maximum bitrate in Kbps (>0). The default value is 3000 Kbps.
 	 */
 	maxKbps: number;
 	
@@ -4115,7 +4202,7 @@ export interface LiveTranscodeLayoutConfig {
 	regions: LiveTranscodeLayoutRegion[];
 	
 	/** 
-	 * @brief Data passed through from the App.
+	 * @brief The data you need, up to 4KB in length.
 	 */
 	appData: string;
 	
@@ -4579,7 +4666,7 @@ export declare enum AudioProfileType {
 	
 	/** 
 	 * @brief Default sound quality
-	 *        The sound quality configuration of RoomProfileType{@link #RoomProfileType} set by the server or client side
+	 *        The sound quality set by the server
 	 */
 	default = 0,
 	
@@ -4663,7 +4750,6 @@ export interface EngineConfig {
 }
 
 /** 
- * @type keytype
  * @brief Advanced Audio Coding (AAC) profile.
  */
 export declare enum AAC_PROFILE {
@@ -4700,6 +4786,111 @@ export declare enum TRANSCODING_VIDEO_CODEC {
 	H265 = "H265"
 }
 
+/** 
+ * @brief Subtitle configrations.
+ */
+export interface RTCSubtitleConfig {
+	
+	/** 
+	 * @brief Target language.
+	 */
+	targetLanguage?: string;
+	
+	/** 
+	 * @brief Subtitle mode.
+	 */
+	mode: SUBTITLE_MODE;
+}
+
+/** 
+ * @brief Subtitle mode.
+ */
+export declare enum SUBTITLE_MODE {
+	
+	/** 
+	 * @brief The recognition mode. In this mode, the speech of a users in the room will be recognized and converted into captions.
+	 */
+	ASR_ONLY = 0,
+	
+	/** 
+	 * @brief The translation mode. In this mode, the speech of a users in the room will be converted into captions and then translated.
+	 */
+	ASR_AND_TRANSLATION = 1
+}
+
+/** 
+ * @brief The states of the subtitling task.
+ */
+export interface SubtitleEvent {
+	
+	/** 
+	 * @brief Types of the states of the subtitling task.
+	 */
+	event: SubtitleEventType;
+	
+	/** 
+	 * @brief Error codes of the subtitling task.
+	 */
+	errorCode?: ErrorCode;
+	
+	/** 
+	 * @brief Related information about subtitles.
+	 */
+	errorMessage?: string;
+}
+
+/** 
+ * @brief Types of the states of the subtitling task.
+ */
+export declare enum SubtitleEventType {
+	
+	/** 
+	 * @brief Subtitles are on.
+	 */
+	STARTED = 0,
+	
+	/** 
+	 * @brief Subtitles are off.
+	 */
+	STOPPED = 1,
+	
+	/** 
+	 * @brief Errors occurred concerning the subtitling task.
+	 */
+	ERROR = 2
+}
+
+/** 
+ * @brief Related information about subtitles.
+ */
+export interface SubtitleMessage {
+	
+	/** 
+	 * @brief The speaker's ID.
+	 */
+	userId: string;
+	
+	/** 
+	 * @brief Transcribed or translated texts of the speaker's speech, encoded in UTF-8 format.
+	 */
+	text: string;
+	
+	/** 
+	 * @brief Whether transcribed texts are complete sentences. True means yes and False means no.
+	 */
+	definite: boolean;
+	
+	/** 
+	 * @brief Incremental sequence numbers assigned to transcribed or translated texts of the speaker's speech. Complete and incomplete sentences are numbered individually.
+	 */
+	sequence: number;
+}
+export declare enum ConnectionMode {
+	AUTO = "auto",
+	FORCE_TCP = "forceTCP",
+	FORCE_UDP = "forceUDP"
+}
+
 /** 
  * @brief Point-to-point text message
  */
@@ -4786,15 +4977,11 @@ declare const Config: {
 };
 declare function setParameter(key: keyof typeof Config, value: any): void;
 declare function getParameter(key: keyof typeof Config): any;
-/**
- * @id VERTC
- * @list 104477
- * @detail 104478
- */
+
 /** 
  * @id BytePlusRTC
- * @list 104477
- * @detail 104478
+ * @list overview
+ * @detail api
  */
 export interface IVERTC {
 	
@@ -4813,7 +5000,7 @@ export interface IVERTC {
 	createEngine: (appId: string, config?: EngineConfig) => IRTCEngine;
 	
 	/** 
-	 * @brief Call this API to destroy the IRTCEngine instance created by [createEngine](#createengine).
+	 * @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.
 	 * @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.
@@ -4830,7 +5017,7 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to enumerate media devices and their IDs, such as microphones, cameras, and headsets.
-	 *        Call this API to get deviceId used to specify the device when calling [setAudioPlaybackDevice](#setaudioplaybackdevice), [startVideoCapture](#startvideocapture), and [startAudioCapture](#startaudiocapture).
+	 *        Call this API to get deviceId used to specify the device when calling [setAudioPlaybackDevice](104478#setaudioplaybackdevice), [startVideoCapture](104478#startvideocapture), and [startAudioCapture](104478#startaudiocapture).
 	 * @return A list of all media devices.
 	 * @notes Only when the browser has permission to the media devices can it enumerate the full device list. Make sure the media device access is turned on before calling this API.
 	 */
@@ -4861,7 +5048,7 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to enumerate microphone devices.
-	 *        Call this API to get Device IDs used to specify the device when calling [startAudioCapture](#startaudiocapture).
+	 *        Call this API to get Device IDs used to specify the device when calling [startAudioCapture](104478#startaudiocapture) and [setAudioCaptureDevice](104478#setaudiocapturedevice).
 	 * @return A list including all microphone devices.
 	 * @notes Only when the browser has permission to the media devices can it enumerate the full device list. Make sure the media device access is turned on before calling this API.
 	 */
@@ -4869,7 +5056,7 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to enumerate audio and video devices and their IDs, such as microphones, cameras, and headsets.
-	 *        Call this API to get Device IDs used to specify the device when calling [startVideoCapture](#startvideocapture).
+	 *        Call this API to get Device IDs used to specify the device when calling [startVideoCapture](104478#startvideocapture) and [setVideoCaptureDevice](104478#setvideocapturedevice).
 	 * @return A list including all camera devices.
 	 * @notes Only when the browser has permission to the media devices can it enumerate the full device list. Make sure the media device access is turned on before calling this API.
 	 */
@@ -4877,7 +5064,7 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to enumerate audio playback devices and their IDs, such as speakers and headsets.
-	 *        Call this API to get Device IDs used to specify the device when calling [setAudioPlaybackDevice](#setaudioplaybackdevice).
+	 *        Call this API to get Device IDs used to specify the device when calling [setAudioPlaybackDevice](104478#setaudioplaybackdevice).
 	 * @return A list including all audio playback devices.
 	 * @notes Only when the browser has permission to the media devices can it enumerate the full device list. Make sure the media device access is turned on before calling this API.
 	 */
@@ -4906,18 +5093,18 @@ export interface IVERTC {
 	setLogConfig: ({ logLevel, LogfileSize }: {
 		
 		/** 
-		* @brief The logging level of local cache.
-		* • `none`: (Default) Turn off logging.
-		* • `error`: Error.
-		* • `warning`: Warning.
-		* • `info`: Information.
-		* • `trace`: Trace.
-		*/
+		 * @brief The logging level of local cache.
+		 * • `none`: (Default) Turn off logging.
+		 * • `error`: Error.
+		 * • `warning`: Warning.
+		 * • `info`: Information.
+		 * • `trace`: Trace.
+		 */
 		logLevel?: "none" | "error" | "warning" | "info" | "trace";
 		
 		/** 
-		* @brief The maximum cache space available for logs in MB. the range is 1 to 100 MB, and the default value is 10 MB.
-		*/
+		 * @brief The maximum cache space available for logs in MB. the range is 1 to 100 MB, and the default value is 10 MB.
+		 */
 		LogfileSize?: number;
 	}) => void;
 	
@@ -4946,10 +5133,12 @@ declare class VERTC implements IVERTC {
 	setParameter: typeof setParameter;
 	isSupported: () => Promise<boolean>;
 	getSupportedCodecs: () => Promise<string[]>;
+	ConnectionMode: typeof ConnectionMode;
 	events: typeof EngineEventsTypes;
 	ErrorCode: typeof ErrorCode;
 	platform: string;
 	private _monitor;
+	private _rtmMonitor;
 	downloadLog: (key?: string | undefined) => void;
 	setLogConfig: ({ logLevel, LogfileSize }: {
 		logLevel?: "none" | "error" | "warning" | "info" | "trace" | undefined;