@byteplus/rtc 4.47.10 → 4.49.1

package/index.d.ts

@@ -526,8 +526,8 @@ export interface IAudioMixingManager {
 	/** 
 	 * @brief Get audio file duration
 	 * @param id   Mix ID
-	 * @return + > 0: Success, audio file duration, in milliseconds.
-	 *         + < 0: Failure
+	 * @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
 	 */
@@ -536,8 +536,8 @@ export interface IAudioMixingManager {
 	/** 
 	 * @brief Get audio file playback progress
 	 * @param id   Mix ID
-	 * @return + > 0: Success, audio file playback progress, in milliseconds.
-	 *         + < 0: Failed
+	 * @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
 	 */
@@ -665,7 +665,6 @@ export interface MixOption {
 }
 declare class EnhancedEventEmitter<EventTypes extends ValidEventTypes = string | symbol> extends EventEmitter<EventTypes> {
 	safeEmit<T extends EventNames<EventTypes>>(event: T, ...args: EventArgs<EventTypes, T>): boolean;
-	safeEmitAsPromise(event: string, ...args: any[]): Promise<any>;
 }
 
 /** 
@@ -942,7 +941,7 @@ export interface onVideoFirstFrameDecodedEvent {
 /** 
  * Event related to decoding the first video frame of the public stream.
  */
-export type PublicStreamVideoFirstFrameDecodedEvent = Omit<
+export declare type PublicStreamVideoFirstFrameDecodedEvent = Omit<
 
 /** 
  * @brief Event related to the decoding of the first video frame.
@@ -1417,7 +1416,7 @@ export interface RemoteStreamStats {
 /** 
  * @brief Subscribed public stream's statistics and network condition, reference period 2s.
  */
-export type PublicStreamStats = Omit<
+export declare type PublicStreamStats = Omit<
 
 /** 
  * @brief Remote stream stats
@@ -1741,7 +1740,7 @@ export interface PlayerEvent {
 	/** 
 	 * @brief userId of the user who is joining the room
 	 */
-	userId?: string;
+	userId: string;
 	
 	/** 
 	 * @brief  The public stream's ID.
@@ -1824,20 +1823,20 @@ export interface RemoteAudioPropertiesInfo {
 export interface AudioPropertiesInfo {
 	
 	/** 
-	 * @brief linear volume. The value is in linear relation to the original volume. The higher the value, the higher the volume. The range is [0,255].
-	 *        - [0, 25]: Silence
-	 *        - [26, 75]: Low volume
-	 *        - [76, 204]: Medium volume
-	 *        - [205, 255]: High volume
+	 * @brief linear volume. The value is in linear relation to the original volume. The higher the value, the higher the volume. The range is [0,255]. <br>
+	 *        - [0, 25]: Silence <br>
+	 *        - [26, 75]: Low volume <br>
+	 *        - [76, 204]: Medium volume <br>
+	 *        - [205, 255]: High volume <br>
 	 */
 	linearVolume: number;
 	
 	/** 
-	 * @brief non-linear volume in dB. The value is in proportion to the log value of the original volume. You can use the value to recognize the Active Speaker in the room. The range is [-127, 0].
-	 *        - [-127, -60]: Silence
-	 *        - [-59, -40]: Low volume
-	 *        - [-39, -20]: Medium volume
-	 *        - [-19, 0]: High volume
+	 * @brief non-linear volume in dB. The value is in proportion to the log value of the original volume. You can use the value to recognize the Active Speaker in the room. The range is [-127, 0]. <br>
+	 *        - [-127, -60]: Silence <br>
+	 *        - [-59, -40]: Low volume <br>
+	 *        - [-39, -20]: Medium volume <br>
+	 *        - [-19, 0]: High volume <br>
 	 */
 	nonlinearVolume: number;
 }
@@ -1871,13 +1870,23 @@ export interface AutoPlayFailedEvent {
 	/** 
 	 * @brief ID of the user whose stream failed to playback automatically. Without this parameter, the notification is about the local stream.
 	 */
-	userId?: string;
+	userId: string;
 	
 	/** 
 	 * @brief Type of the media which is failed to play automatically
 	 *        For example, when both audio and video streams from the same user failed to playback automatically, you would get notified by `onAutoPlayFailed` twice with the argument of `audio` and `video` respectively.
 	 */
-	kind?: "audio" | "video";
+	kind: "audio" | "video";
+	
+	/** 
+	 * @brief The attributes of the screen-sharing stream.
+	 */
+	streamIndex: StreamIndex;
+	
+	/** 
+	 * @brief Remote stream type
+	 */
+	mediaType: MediaType.AUDIO | MediaType.VIDEO;
 }
 declare enum StreamMixingEventErrorCode {
 	
@@ -1927,14 +1936,14 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onPlayerEvent]: (event: PlayerEvent) => void;
 	
 	/** 
-	 * @brief The user joined.
+	 * @brief Callback for a remote visible user joining the room, or an invisible user in the room switching to visible.
 	 * @param event
 	 * @region Room Management Event
 	 */
 	[EngineEventsTypes.onUserJoined]: (event: onUserJoinedEvent) => void;
 	
 	/** 
-	 * @brief The user leaved.
+	 * @brief Callback for a remote visible user leaving the room, or a visible user in the room switching to invisible.
 	 * @param event
 	 * @region Room Management Event
 	 */
@@ -2168,6 +2177,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onLocalStreamStats]: (stats: LocalStreamStats) => void;
 	
 	/** 
+	 * @hidden
 	 * @brief This 2 s periodical callback notifies information of the public stream after calling [startPlayPublicStream](104478#startplaypublicstream).
 	 * @param stats
 	 */
@@ -2381,6 +2391,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onCloudProxyConnected]: (event: CloudProxyConnectedEvent) => void;
 	
 	/** 
+	 * @hidden
 	 * @brief Callback of the result of publishing the public stream.
 	 *        After calling [startPushPublicStream](104478#startpushpublicstream), you will be informed of the result through this callback.
 	 * @param event
@@ -2399,6 +2410,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onPublicStreamSEIMessageReceived]: (event: PublicStreamSEIMessageEvent) => void;
 	
 	/** 
+	 * @hidden
 	 * @brief Callback of the successful decoding of the first video frame of the public stream.
 	 *        Refer to [startPlayPublicStream](104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
@@ -2406,6 +2418,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onFirstPublicStreamVideoFrameDecoded]: (event: PublicStreamVideoFirstFrameDecodedEvent) => void;
 	
 	/** 
+	 * @hidden
 	 * @brief Callback of the successful decoding of the first audio frame of the public stream.
 	 *        Refer to [startPlayPublicStream](104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
@@ -2413,6 +2426,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onFirstPublicStreamAudioFrameDecoded]: (event: PublicStreamAudioFirstFrameDecodedEvent) => void;
 	
 	/** 
+	 * @hidden
 	 * @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
@@ -2626,7 +2640,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param userId The ID of the remote user who published the target audio/video stream.
 	 * @param mediaType Media stream type, used for specifying whether to subscribe to the audio stream or the video stream.
 	 * @notes You must first get the remote stream
-	 * information through onUserPublishStream{@link #IRTCRoomEventHandler#onUserPublishStream} before calling this API to subscribe to streams accordingly.
+	 * information through [onUserPublishStream](104479#onuserpublishstream) before calling this API to subscribe to streams accordingly.
 	 * @region Room Management
 	 */
 	subscribeStream: (userId: string, mediaType: MediaType) => Promise<void>;
@@ -2692,8 +2706,9 @@ 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.
-	 * @param deviceId ID of the device need to enable which you can call [enumerateDevices](#enumeratedevices) to obtain.
+	 * @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.
@@ -2716,7 +2731,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Start internal audio capture. The default is off.
 	 * Internal audio capture refers to: capturing audio using the built-in module.
 	 * The remote clients in the room will be informed of the state change via [onUserStartAudioCapture](104479#onuserstartaudiocapture) after the publisher starts audio capture by calling this API.
-	 * @param deviceId ID of the device need to enable which you can call [enumerateDevices](#enumeratedevices) to obtain.
+	 * @param deviceId ID of the capture device which you need to pass in to avoid exceptions such as no sound. You can call [enumerateDevices](#enumeratedevices) to obtain the list of devices.
 	 * @return
 	 * @notes + Call [stopAudioCapture](#stopaudiocapture) to stop the internal audio capture. Otherwise, the internal audio capture will sustain until you destroy the engine instance.
 	 *     + Once you create the engine instance, you can start internal audio capture regardless of the audio publishing state. The audio stream will start publishing only after the audio capture starts.
@@ -2767,7 +2782,7 @@ 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` is triggered.
+	 *     It's recommended to call this API and bind a view once the event of [onUserPublishStream](104479#onuserpublishstream) 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
@@ -2797,12 +2812,14 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * 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.
 	 * @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.
+	 * @param streamIndex Video stream properties
 	 * @return Calling this API when the audio and video are playing will also return a Promise.
 	 * @region Room Management
 	 * @note + Call this API to start playback if the audio and/or video is not playing.
 	 * + Call this API once to start both audio and video playbacks when they do not start.
 	 */
-	play: (userId?: string) => Promise<void>;
+	play: (userId?: string, mediaType?: MediaType, streamIndex?: StreamIndex) => Promise<void>;
 	
 	/** 
 	 * @brief Pause receiving remote media streams.
@@ -2884,7 +2901,8 @@ 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 + 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.
+	 * @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.
 	 *        + 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
@@ -2905,7 +2923,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @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, see RemoteVideoConfig{@link #RemoteVideoConfig}.
+	 * @param remoteVideoConfig Your expected configuration.
 	 * @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).
@@ -2930,8 +2948,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * Local client will get informed of the the volume of local user and each subscribed users via [onAudioVolumeIndication](104479#onaudiovolumeindication).
 	 * @param interval The time interval for receiving a volume prompt callback.
 	 *        It is recommended to set to greater than or equal to 200 milliseconds; when less than 10 milliseconds, the behavior is undefined.
-	 *         + ≤ 0: Disable the volume prompt function.
-	 *         + > 0: Enable the volume prompt function and set the time interval for receiving the volume prompt callback. In milliseconds.
+	 *         + `≤ 0`: Disable the volume prompt function.
+	 *         + `> 0`: Enable the volume prompt function and set the time interval for receiving the volume prompt callback. In milliseconds.
 	 * @region Room Management
 	 */
 	setAudioVolumeIndicationInterval: (interval?: number) => void;
@@ -2948,7 +2966,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Starts pushing media streams to CDN and sets the relevant configurations.
-	 * @param transcode configuration for push media streams to CDN.
+	 * @param transcode Configurations to be set when pushing media streams to CDN.
 	 * @return
 	 * @notes + After calling this API, you will receive the pushing a mixed stream to CDN events via [onStreamMixingEvent](104479#onstreammixingevent).
 	 *        + Call [stopLiveTranscoding](#stoplivetranscoding) to stop pushing mixed streams to CDN.
@@ -2958,9 +2976,9 @@ 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.
-	 * @param transcode Configurations that you want to update.
-	 * @region Advanced Feature
+	 *        After calling [startLiveTranscoding](#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
 	 */
 	updateLiveTranscoding: (transcode: Partial<LiveTranscodeConfig>) => Promise<void>;
 	
@@ -3048,18 +3066,17 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @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).
-	 * @region Audio Management
 	 */
 	setAudioCaptureConfig: (config: TrackCaptureConfig) => Promise<void>;
 	
 	/** 
 	 * @brief Configure the internal video capture. Different browser has its own default settings.
-	 * @param config Configuration of the internal video capture, For supported options please see: [MediaTrackConstraints](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints)
+	 * @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).
-	 * @region Video Management
 	 */
-	setVideoCaptureConfig: (config: TrackCaptureConfig) => Promise<void>;
+	setVideoCaptureConfig: (config: TrackCaptureConfig) => Promise<MediaTrackSettings>;
 	
 	/** 
 	 * @brief Log in to the RTM server.
@@ -3150,18 +3167,19 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param  cloudProxyInfo cloud proxy informarion list.
 	 * @notes  + Call this API before joining the room.
 	 *         + Start pre-call network detection after starting cloud proxy.
-	 *         + After starting cloud proxy and connects the cloud proxy server successfully, receives [onCloudProxyConnected](104479#onCloudProxyConnected).
-	 *         + To stop cloud proxy, call [stopCloudProxy](#stopCloudProxy).
+	 *         + After starting cloud proxy and connects the cloud proxy server successfully, receives [onCloudProxyConnected](104479#oncloudproxyconnected).
+	 *         + To stop cloud proxy, call [stopCloudProxy](#stopcloudproxy).
 	 */
 	startCloudProxy: (cloudProxyInfo: CloudProxyInfo) => void;
 	
 	/** 
 	 * @brief Stop cloud proxy
-	 * @notes To start cloud proxy, call [startCloudProxy](#startCloudProxy).
+	 * @notes To start cloud proxy, call [startCloudProxy](#startcloudproxy).
 	 */
 	stopCloudProxy: () => void;
 	
 	/** 
+	 * @hidden
 	 * @brief Starts publishing a public media stream.
 	 * @param publicStreamId The public stream's ID.
 	 * @param publicStreamParam The public stream's parameters.
@@ -3177,6 +3195,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	startPushPublicStream: (publicStreamId: string, publicStreamParam: PublicStreamConfig) => Promise<void>;
 	
 	/** 
+	 * @hidden
 	 * @brief Update the properties of the public stream published by the current user.
 	 *        Call [startPushPublicStream](#startpushpublicstream) to start publishing a public stream.
 	 * @param publicStreamId The public stream's ID.
@@ -3187,6 +3206,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	updatePublicStreamParam: (publicStreamId: string, publicStreamParam: Partial<PublicStreamConfig>) => Promise<void>;
 	
 	/** 
+	 * @hidden
 	 * @type api
 	 * @brief Stops publishing the public stream.
 	 *        Call [startPushPublicStream](#startpushpublicstream) to start publishing a public stream.
@@ -3197,6 +3217,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	stopPushPublicStream: (publicStreamId: string) => Promise<void>;
 	
 	/** 
+	 * @hidden
 	 * @brief Subscribes to a designated public stream.
 	 *        You can call this API to subscribe to a public stream whether you joined the room or not.
 	 * @param publicStreamId The public stream's ID.
@@ -3209,6 +3230,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	startPlayPublicStream: (publicStreamId: string) => Promise<void>;
 	
 	/** 
+	 * @hidden
 	 * @brief Unsubscribes from a designated public stream.
 	 *        Call [startPlayPublicStream](#startplaypublicstream) to subscribe to the public stream.
 	 * @param publicStreamId The public stream's ID.
@@ -3217,6 +3239,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	stopPlayPublicStream: (publicStreamId: string) => Promise<void>;
 	
 	/** 
+	 * @hidden
 	 * @brief Sets the player information and rendering mode for the public stream.
 	 * @param publicStreamId The public stream's ID.
 	 * @param videoPlayerOption Player information and rendering mode.
@@ -3249,7 +3272,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return + `true` Success
 	 *         + `false` Failure
 	 * @notes + You must call this API before joining the room.
-	 *        + After you enable the fallback, you will receive [onSimulcastSubscribeFallback](104479#onsimulcastsubscribefallback) and [onRemoteVideoSizeChanged](104479#onremotevideosizechanged} when the resolution of your subscribed stream is lowered or restored.
+	 *        + After you enable the fallback, you will receive [onSimulcastSubscribeFallback](104479#onsimulcastsubscribefallback) and [onRemoteVideoSizeChanged](104479#onremotevideosizechanged) when the resolution of your subscribed stream is lowered or restored.
 	 *        + You can set media stream fallback using API or cloud configuration. The latter takes a higher priority.
 	 */
 	setSubscribeFallbackOption: (option: SubscribeFallbackOption) => boolean;
@@ -3285,6 +3308,7 @@ 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.
 	 * @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.
@@ -3292,7 +3316,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * + Call [stopAudioDevicePlayTest](#stopaudiodeviceplaytest) to stop the test, including capturing and playing the recording.
 	 * + This test performs locally and does not involve network connection testing.
 	 */
-	startAudioDeviceRecordTest: (indicationInterval: number) => Promise<void>;
+	startAudioDeviceRecordTest: (indicationInterval: number, onAutoplayFailed?: (resume: () => Promise<any>) => any) => Promise<void>;
 	
 	/** 
 	 * @brief Call this API to stop recording in the test and start to play the recording.
@@ -3324,9 +3348,9 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * + Callback of the MediaStreamTrack will inform you by the event of `ended` that the local media capture stops, probably caused either by an API call or the removal of devices.
 	 * + You can get the following info from the MediaStreamTrack: video resolution, device ID, and volume. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
 	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.
-	 *   1. Get the video resolution or audio volume of the local streams: [onLocalStreamStats](70081#onlocalstreamstats)
-	 *   2. Get device IDs: [`enumerateDevices`](#enumeratedevices)
-	 *   3. stop: [`stopAudioCapture`](#stopaudiocapture), [`stopVideoCapture`](#stopvideocapture)
+	 *   1. Get the video resolution or audio volume of the local streams: [onLocalStreamStats](104479#onlocalstreamstats)
+	 *   2. Get device IDs: [enumerateDevices](#enumeratedevices)
+	 *   3. stop: [stopAudioCapture](#stopaudiocapture), [stopVideoCapture](#stopvideocapture)
 	 */
 	getLocalStreamTrack: (streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
@@ -3921,7 +3945,7 @@ export interface ScreenEncoderConfig {
 /** 
  * @brief Internal capture configuration for video & audio. Refer to [browser documentation](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints) for more details.
  */
-export type TrackCaptureConfig = Omit<MediaTrackConstraints, "deviceId">;
+export declare type TrackCaptureConfig = Omit<MediaTrackConstraints, "deviceId">;
 
 /** 
  * @brief The mirror type
@@ -3961,22 +3985,24 @@ export interface AudioPropertiesConfig {
 }
 
 /** 
- * @brief Configurations to be set when pushing media streams to CDN
+ * @brief Configurations to be set when pushing media streams to CDN.
  */
 export interface LiveTranscodeConfig {
 	
 	/** 
-	 * @brief CDN url where the mixed stream is supposed to be pushed.
+	 * @brief The URL for live transcoding. Only supports live transcoding via RTMP. The URL should match the regular expression `/^rtmps?:\/\//`.
+	 *        This parameter cannot be updated while pushing stream to the CDN.
 	 */
 	url: string;
 	
 	/** 
-	 * Audio transcoding related configurations.
+	 * @brief Audio transcoding configurations.
+	 *        These parameters cannot be updated while pushing stream to the CDN.
 	 */
 	audio?: Partial<LiveTranscodeAudioConfig>;
 	
 	/** 
-	 * @brief Video transcoding related configurations.
+	 * @brief Video transcoding configurations.
 	 */
 	video?: Partial<LiveTranscodeVideoConfig>;
 	
@@ -3999,107 +4025,112 @@ export declare enum LogChannel {
 }
 
 /** 
- * @brief Video transcoding related configurations. Assign a value exceeding the range will lead to a failure of starting or updating a live transcoding mission.
+ * @brief Video transcoding configurations. With invalid or empty input, the configurations will be set as the default values.
  */
 export interface LiveTranscodeVideoConfig {
 	
 	/** 
-	 * @brief Width in px. Range: [2, 4096]. The value must be even. The default value is 640 px.
+	 * @brief The width (pixels) to be set. The range is [2, 1920], and must be an even number. The default value is 640 pixels.
+	 *        If an odd number is set, the width will be adjusted to the next larger even number.
 	 */
 	width: number;
 	
 	/** 
-	 * @brief Height in px. Range: [2, 2160]. The value must be even. The default value is 360 px.
+	 * @brief The height (pixels) to be set. The range is [2, 1920], and must be an even number. The default value is 360 pixels.
+	 *        If an odd number is set, the height will be adjusted to the next larger even number.
 	 */
 	height: number;
 	
 	/** 
-	 * @brief The video encoding format to be set. The default value is `H264`.
+	 * @brief The video codec. The default value is `H264`.
+	 *        This parameter cannot be updated while pushing stream to the CDN.
 	 */
-	codec: string;
+	codec: TRANSCODING_VIDEO_CODEC;
 	
 	/** 
-	 * @brief Frame rate in fps. The value must be larger than 0. The default value is 30 fps.
+	 * @brief The frame rate (FPS) in range of [1, 60]. The default value is 15 FPS.
 	 */
 	fps: number;
 	
 	/** 
-	 * @brief Number of frames between two I-frames. The value must be larger than 0. The default value is 60.
+	 * @brief The time interval between I-frames (second) in range of [1, 5]. The default value is 2 seconds.
+	 *        This parameter cannot be updated while pushing stream to the CDN.
 	 */
 	gop: number;
 	
 	/** 
+	 * @hidden
 	 * @brief Whether to push streams with low latency. The default value is `true`.
 	 */
-	lowLatency: boolean;
 	
 	/** 
-	 * @brief Bitrate (Kbps) of the mixed video stream in Kbps. Range: [16, 50000000]. The default value is 600 Kbps.
+	 * @brief The bitrate (Kbps) in range of [1, 10000]. The default value is self-adaptive.
 	 */
 	kBitRate: number;
 }
 
 /** 
- * @brief Audio transcoding related configurations
+ * @brief Audio transcoding configurations. With invalid or empty input, the configurations will be set as the default values.
  */
 export interface LiveTranscodeAudioConfig {
 	
 	/** 
 	 * @brief Audio coding standard. Optional. The default value is `AAC`.
 	 */
-	codec?: string;
+	codec?: "AAC";
 	
 	/** 
-	 * @brief The bitrate (kbps) to be set. The value must be 32, 64, 128, or 192. The default value is 64 Kbps.
+	 * @brief The bitrate (Kbps) in range of [32, 192]. Defaults to 64 Kbps.
 	 */
 	kBitRate: number;
 	
 	/** 
-	 * @brief The sample rate (kHz) to be set. The value must be 32, 44.1, or 48. The default value is 48 Khz.
+	 * @brief The sample rate (Hz). Supported sample rates: 32,000Hz, 44,100Hz, 48,000Hz. Defaults to 48,000Hz.
 	 */
 	sampleRate: number;
 	
 	/** 
-	 * @brief The number of sound channels to be set. The value must be 1 or 2. The default value is 2.
+	 * @brief The number of channels. Supported channels: 1 (single channel), 2 (dual channel).  Defaults to 2.
 	 */
 	channels: number;
 	
 	/** 
-	 * @brief The AAC profile to be set. The value must be `LC`, `HEv1`, or `HEv2`. The default value is `LC`.
+	 * @brief AAC profile. Defaults to `LC`.
 	 */
-	AACProfile: "LC" | "HEv1" | "HEv2";
+	AACProfile: AAC_PROFILE;
 }
 
 /** 
- * @brief User layout information
+ * @brief The overall video layout of the mixed stream.
  */
 export interface LiveTranscodeLayoutConfig {
 	
 	/** 
-	 * @brief User layout information list
+	 * @brief List of user's video layout information. With invalid or empty input, the configurations will be set as the default values.
 	 */
 	regions: LiveTranscodeLayoutRegion[];
 	
 	/** 
-	 * @brief Data passed through from the App
+	 * @brief Data passed through from the App.
 	 */
 	appData: string;
 	
 	/** 
-	 * @brief Background color of the canvas
+	 * @brief Background-color of the mixed stream in hexadecimal values such as #FFFFFF and #000000. The default value is #000000 (black).
+	 *        With invalid or empty input, the configurations will be set as the default values.
 	 */
 	backgroundColor: string;
 }
 
 /** 
- * @brief Layout information for one of the video streams to be mixed
+ * @brief Layout information for one of the video streams to be mixed.
  */
 export interface LiveTranscodeLayoutRegion {
 	
 	/** 
-	 * @brief Transparency to be set. The range is [0.0, 1.0].
+	 * @brief The opacity in range of (0.0, 1.0]. The lower value, the more transparent. The default value is 1.0.
 	 */
-	alpha: number;
+	alpha?: number;
 	
 	/** 
 	 * @brief ID of the user who published the stream
@@ -4107,43 +4138,43 @@ export interface LiveTranscodeLayoutRegion {
 	userId: string;
 	
 	/** 
-	 * @brief The normalization ratio of the x-coordinate against the whole image which ranging from [0.0,1.0)
+	 * @brief The normalized horizontal coordinate value of the top left end vertex of the user's view, ranging within [0.0, 1.0). The default value is 0.0.
 	 */
-	x: number;
+	x?: number;
 	
 	/** 
-	 * @brief The normalized horizontal coordinate value of the top left end vertex of the user's view, ranging within [0.0, 1.0)
+	 * @brief The normalized vertical coordinate value of the top left end vertex of the user's view, ranging within [0.0, 1.0). The default value is 0.0.
 	 */
-	y: number;
+	y?: number;
 	
 	/** 
-	 * @brief The normalized vertical coordinate value of the top left end vertex of the user's view, ranging within [0.0, 1.0)
+	 * @brief The layer on which the video is rendered. The range is [0, 100]. 0 for the bottom layer, and 100 for the top layer. The default value is 0.
 	 */
-	zorder: number;
+	zorder?: number;
 	
 	/** 
-	 * @brief The normalized width of the user's view, ranging within (0.0, 1.0]
+	 * @brief The normalized width of the user's view, ranging within (0.0, 1.0]. The default value is 1.0.
 	 */
-	w: number;
+	w?: number;
 	
 	/** 
-	 * @brief The normalized height of the user's view, ranging within (0.0, 1.0]
+	 * @brief The normalized height of the user's view, ranging within (0.0, 1.0]. The default value is 1.0.
 	 */
-	h: number;
+	h?: number;
 	
 	/** 
-	 * @brief The view mode to be set. See the following options:
-	 * + 1: Fill and Crop. The video is enlarged or shrunk with fixed aspect ratio, and completely fills the windows. The region of the video which exceeds the window will be cropped.
-	 * + 2: Fit. The video is enlarged or shrunk with fixed aspect ratio, and is shown completely in the windows. The region of the window, which is not filled with video, is filled with black background.
-	 * + 3: Stretch. The video's aspect ratio is changed as it's automatically stretched to completely fill the bounding box it's contained in, but the whole image is visible.
+	 * @brief The render mode to be set. The default value is 1. See the following options:
+	 * • 1: Fill and Crop. The video is enlarged or shrunk with fixed aspect ratio, and completely fills the windows. The region of the video which exceeds the window will be cropped.
+	 * • 2: Fit. The video is enlarged or shrunk with fixed aspect ratio, and is shown completely in the windows. The region of the window, which is not filled with video, is filled with black background.
+	 * • 3: Stretch. The video's aspect ratio is changed as it's automatically stretched to completely fill the bounding box it's contained in, but the whole image is visible.
 	 */
 	renderMode?: number;
 	
 	/** 
 	 * @brief The content to be set. The default value is 0. See the following options:
-	 * + 0: Push both audio and video stream.
-	 * + 1: Push audio stream only.
-	 * + 2: Push video stream only.
+	 * • 0: Push both audio and video stream.
+	 * • 1: Push audio stream only.
+	 * • 2: Push video stream only.
 	 */
 	contentControl?: number;
 	
@@ -4153,9 +4184,10 @@ export interface LiveTranscodeLayoutRegion {
 	isScreenStream: boolean;
 	
 	/** 
+	 * @hidden
 	 * @brief Whether the source user of the stream is a local user.
 	 */
-	isLocalUser: boolean;
+	isLocalUser?: boolean;
 }
 
 /** 
@@ -4240,7 +4272,7 @@ export declare enum PublicInterpolationMode {
 /** 
  * @brief Layout configuration of a media stream
  */
-export type PublicStreamLayoutRegion = Omit<LiveTranscodeLayoutRegion, "isLocalUser" | "contentControl"> & {
+export declare type PublicStreamLayoutRegion = Omit<LiveTranscodeLayoutRegion, "isLocalUser" | "contentControl"> & {
 	
 	/** 
 	 * @brief Room ID of the user that publishes the media stream.
@@ -4626,6 +4658,44 @@ export interface EngineConfig {
 	expectedIDC: string;
 }
 
+/** 
+ * @type keytype
+ * @brief Advanced Audio Coding (AAC) profile.
+ */
+export declare enum AAC_PROFILE {
+	
+	/** 
+	 * @brief (Default) AAC Low-Complexity profile (AAC-LC).
+	 */
+	LC = "LC",
+	
+	/** 
+	 * @brief HE-AAC v1 profile (AAC LC with SBR).
+	 */
+	HEv1 = "HEv1",
+	
+	/** 
+	 * @brief HE-AAC v2 profile (AAC LC with SBR and Parametric Stereo).
+	 */
+	HEv2 = "HEv2"
+}
+
+/** 
+ * @brief The video codec.
+ */
+export declare enum TRANSCODING_VIDEO_CODEC {
+	
+	/** 
+	 * @brief (Default) H.264 format.
+	 */
+	H264 = "H264",
+	
+	/** 
+	 * @brief ByteVC1 format.
+	 */
+	H265 = "H265"
+}
+
 /** 
  * @brief Point-to-point text message
  */
@@ -4687,10 +4757,24 @@ export interface PublicStreamResultEvent {
 	
 	/** 
 	 * @brief Event details of publishing the public stream.
+	 * + `200`: Published or subscribed successfully.
+	 * + `1191`: Invalid parameter(s). Please revise the paramter(s) and retry.
+	 * + `1192`: Error for the task at the server side. The server will retry automatically.
+	 * + `1193`: Unrecoverable error of publishing the public stream. Please start the task again.
+	 * + `1195`: Failed to publish the public stream. The SDK  will retry that you have no need to handle.
+	 * + `1196`: Time-out error of publishing the public stream. The SDK will retry in 10 s. The SDK will stop trying again after three times of failures.
 	 */
 	message: string;
 }
+
+/** 
+ * @brief SDK successful connected to the cloud proxy server event.
+ */
 export interface CloudProxyConnectedEvent {
+	
+	/** 
+	 * @brief The time interval to join the channel after SDK turned on cloud proxy.
+	 */
 	interval: number;
 }
 declare const Config: {
@@ -4713,7 +4797,7 @@ declare function getParameter(key: keyof typeof Config): string | boolean | stri
  * @detail 104478
  */
 /** 
- * @id VERTC
+ * @id BytePlusRTC
  * @list 104477
  * @detail 104478
  */
@@ -4819,6 +4903,34 @@ export interface IVERTC {
 	 * @region Engine Management
 	 */
 	getSupportedCodecs: () => Promise<string[]>;
+	
+	/** 
+	 * @brief Configures the logging level of local cache, and the maximum cache space that can be used.
+	 * @notes This API must be called before [createEngine](#createengine). If called after [createEngine](#createengine), it will take effect after the next [createEngine](#createengine) after [destroyEngine](#destroyengine).
+	 */
+	setLogConfig: ({ logLevel, LogfileSize }: {
+		
+		/** 
+		* @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.
+		*/
+		LogfileSize?: number;
+	}) => void;
+	
+	/** 
+	 * @brief Download local cache generated file.
+	 * @param key The key of the local cache. If empty, download the logs from the current cache.
+	 */
+	downloadLog: (key?: string) => void;
 }
 declare class VERTC implements IVERTC {
 	getSdkVersion: () => string;
@@ -4843,6 +4955,11 @@ declare class VERTC implements IVERTC {
 	ErrorCode: typeof ErrorCode;
 	platform: string;
 	private _monitor;
+	downloadLog: (key?: string | undefined) => void;
+	setLogConfig: ({ logLevel, LogfileSize }: {
+		logLevel?: "none" | "error" | "warning" | "info" | "trace" | undefined;
+		LogfileSize?: number | undefined;
+	}) => void;
 }
 declare const _default: VERTC;