agora-electron-sdk 4.2.6 → 4.3.0-dev.1

package/ts/Private/IAgoraRtcEngineEx.ts

@@ -50,7 +50,7 @@ export abstract class IRtcEngineEx extends IRtcEngine {
    * You can call this method multiple times to join more than one channel.
    *  If you are already in a channel, you cannot rejoin it with the same user ID.
    *  If you want to join the same channel from different devices, ensure that the user IDs are different for all devices.
-   *  Ensure that the app ID you use to generate the token is the same as the app ID used when creating the IRtcEngine instance.
+   *  Ensure that the App ID you use to generate the token is the same as the App ID used when creating the IRtcEngine instance.
    *
    * @param token The token generated on your server for authentication. If you need to join different channels at the same time or switch between channels, Agora recommends using a wildcard token so that you don't need to apply for a new token every time joining a channel.
    * @param connection The connection information. See RtcConnection.
@@ -111,9 +111,9 @@ export abstract class IRtcEngineEx extends IRtcEngine {
   ): number;
 
   /**
-   * Sets the encoder configuration for the local video.
+   * Sets the video encoder configuration.
    *
-   * Each configuration profile corresponds to a set of video parameters, including the resolution, frame rate, and bitrate. The config specified in this method is the maximum value under ideal network conditions. If the video engine cannot render the video using the specified config due to unreliable network conditions, the parameters further down the list are considered until a successful configuration is found.
+   * Sets the encoder configuration for the local video. Each configuration profile corresponds to a set of video parameters, including the resolution, frame rate, and bitrate. The config specified in this method is the maximum value under ideal network conditions. If the video engine cannot render the video using the specified config due to unreliable network conditions, the parameters further down the list are considered until a successful configuration is found.
    *
    * @param config Video profile. See VideoEncoderConfiguration.
    * @param connection The connection information. See RtcConnection.
@@ -181,12 +181,12 @@ export abstract class IRtcEngineEx extends IRtcEngine {
   ): number;
 
   /**
-   * Sets the stream type of the remote video.
+   * Sets the video stream type to subscribe to.
    *
-   * Under limited network conditions, if the publisher has not disabled the dual-stream mode using enableDualStreamModeEx (false), the receiver can choose to receive either the high-quality video stream or the low-quality video stream. The high-quality video stream has a higher resolution and bitrate, and the low-quality video stream has a lower resolution and bitrate. By default, users receive the high-quality video stream. Call this method if you want to switch to the low-quality video stream. This method allows the app to adjust the corresponding video stream type based on the size of the video window to reduce the bandwidth and resources. The aspect ratio of the low-quality video stream is the same as the high-quality video stream. Once the resolution of the high-quality video stream is set, the system automatically sets the resolution, frame rate, and bitrate of the low-quality video stream. The SDK enables the low-quality video stream auto mode on the sender by default (not actively sending low-quality video streams). The host at the receiving end can call this method to initiate a low-quality video stream stream request on the receiving end, and the sender automatically switches to the low-quality video stream mode after receiving the request.
+   * The SDK defaults to enabling low-quality video stream adaptive mode (AutoSimulcastStream) on the sender side, which means the sender does not actively send low-quality video stream. The receiver can initiate a low-quality video stream request by calling this method, and the sender will automatically start sending low-quality video stream upon receiving the request. By default, users receive the high-quality video stream. Call this method if you want to switch to the low-quality video stream. The SDK will dynamically adjust the size of the corresponding video stream based on the size of the video window to save bandwidth and computing resources. The default aspect ratio of the low-quality video stream is the same as that of the high-quality video stream. According to the current aspect ratio of the high-quality video stream, the system will automatically allocate the resolution, frame rate, and bitrate of the low-quality video stream. Under limited network conditions, if the publisher does not disable the dual-stream mode using enableDualStreamModeEx (false), the receiver can choose to receive either the high-quality video stream, or the low-quality video stream. The high-quality video stream has a higher resolution and bitrate, while the low-quality video stream has a lower resolution and bitrate.
    *
    * @param uid The user ID.
-   * @param streamType The video stream type: VideoStreamType.
+   * @param streamType The video stream type, see VideoStreamType.
    * @param connection The connection information. See RtcConnection.
    *
    * @returns
@@ -542,8 +542,7 @@ export abstract class IRtcEngineEx extends IRtcEngine {
    * After calling createDataStreamEx, you can call this method to send data stream messages to all users in the channel. The SDK has the following restrictions on this method:
    *  Up to 60 packets can be sent per second in a channel with each packet having a maximum size of 1 KB.
    *  Each client can send up to 30 KB of data per second.
-   *  Each user can have up to five data streams simultaneously. A successful method call triggers the onStreamMessage callback on the remote client, from which the remote user gets the stream message.
-   * A failed method call triggers the onStreamMessageError callback on the remote client.
+   *  Each user can have up to five data streams simultaneously. A successful method call triggers the onStreamMessage callback on the remote client, from which the remote user gets the stream message. A failed method call triggers the onStreamMessageError callback on the remote client.
    *  Ensure that you call createDataStreamEx to create a data channel before calling this method.
    *  This method applies only to the COMMUNICATION profile or to the hosts in the LIVE_BROADCASTING profile. If an audience in the LIVE_BROADCASTING profile calls this method, the audience may be switched to a host.
    *
@@ -622,7 +621,7 @@ export abstract class IRtcEngineEx extends IRtcEngine {
    *
    * @param interval Sets the time interval between two consecutive volume indications:
    *  ≤ 0: Disables the volume indication.
-   *  > 0: Time interval (ms) between two consecutive volume indications. The lowest value is 50.
+   *  > 0: Time interval (ms) between two consecutive volume indications. Ensure this parameter is set to a value greater than 10, otherwise you will not receive the onAudioVolumeIndication callback. Agora recommends that this value is set as greater than 100.
    * @param smooth The smoothing factor that sets the sensitivity of the audio volume indicator. The value ranges between 0 and 10. The recommended value is 3. The greater the value, the more sensitive the indicator.
    * @param reportVad true : Enables the voice activity detection of the local user. Once it is enabled, the vad parameter of the onAudioVolumeIndication callback reports the voice activity status of the local user. false : (Default) Disables the voice activity detection of the local user. Once it is disabled, the vad parameter of the onAudioVolumeIndication callback does not report the voice activity status of the local user, except for the scenario where the engine automatically detects the voice activity of the local user.
    * @param connection The connection information. See RtcConnection.
@@ -651,9 +650,9 @@ export abstract class IRtcEngineEx extends IRtcEngine {
    * @returns
    * 0: Success.
    *  < 0: Failure.
-   *  -2: The URL is null or the string length is 0.
+   *  -2: The URL or configuration of transcoding is invalid; check your URL and transcoding configurations.
    *  -7: The SDK is not initialized before calling this method.
-   *  -19: The Media Push URL is already in use, use another URL instead.
+   *  -19: The Media Push URL is already in use; use another URL instead.
    */
   abstract startRtmpStreamWithoutTranscodingEx(
     url: string,
@@ -676,9 +675,9 @@ export abstract class IRtcEngineEx extends IRtcEngine {
    * @returns
    * 0: Success.
    *  < 0: Failure.
-   *  -2: The URL is null or the string length is 0.
+   *  -2: The URL or configuration of transcoding is invalid; check your URL and transcoding configurations.
    *  -7: The SDK is not initialized before calling this method.
-   *  -19: The Media Push URL is already in use, use another URL instead.
+   *  -19: The Media Push URL is already in use; use another URL instead.
    */
   abstract startRtmpStreamWithTranscodingEx(
     url: string,
@@ -743,51 +742,6 @@ export abstract class IRtcEngineEx extends IRtcEngine {
     connection: RtcConnection
   ): number;
 
-  /**
-   * Starts relaying media streams across channels. This method can be used to implement scenarios such as co-host across channels.
-   *
-   * Deprecated: This method is deprecated. Use startOrUpdateChannelMediaRelayEx instead. After a successful method call, the SDK triggers the onChannelMediaRelayStateChanged and onChannelMediaRelayEvent callbacks, and these callbacks return the state and events of the media stream relay.
-   *  If the onChannelMediaRelayStateChanged callback returns RelayStateRunning (2) and RelayOk (0), and the onChannelMediaRelayEvent callback returns RelayEventPacketSentToDestChannel (4), it means that the SDK starts relaying media streams between the source channel and the target channel.
-   *  If the onChannelMediaRelayStateChanged callback returns RelayStateFailure (3), an exception occurs during the media stream relay.
-   *  Call this method after joining the channel.
-   *  This method takes effect only when you are a host in a live streaming channel.
-   *  After a successful method call, if you want to call this method again, ensure that you call the stopChannelMediaRelayEx method to quit the current relay.
-   *  The relaying media streams across channels function needs to be enabled by contacting.
-   *  Agora does not support string user accounts in this API.
-   *
-   * @param configuration The configuration of the media stream relay. See ChannelMediaRelayConfiguration.
-   * @param connection The connection information. See RtcConnection.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
-   *  -1: A general error occurs (no specified reason).
-   *  -2: The parameter is invalid.
-   *  -7: The method call was rejected. It may be because the SDK has not been initialized successfully, or the user role is not a host.
-   *  -8: Internal state error. Probably because the user is not a broadcaster.
-   */
-  abstract startChannelMediaRelayEx(
-    configuration: ChannelMediaRelayConfiguration,
-    connection: RtcConnection
-  ): number;
-
-  /**
-   * Updates the channels for media stream relay.
-   *
-   * Deprecated: This method is deprecated. Use startOrUpdateChannelMediaRelayEx instead. After the media relay starts, if you want to relay the media stream to more channels, or leave the current relay channel, you can call this method. After a successful method call, the SDK triggers the onChannelMediaRelayEvent callback with the RelayEventPacketUpdateDestChannel (7) state code. Call the method after successfully calling the startChannelMediaRelayEx method and receiving onChannelMediaRelayStateChanged (RelayStateRunning, RelayOk); otherwise, the method call fails.
-   *
-   * @param configuration The configuration of the media stream relay. See ChannelMediaRelayConfiguration.
-   * @param connection The connection information. See RtcConnection.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
-   */
-  abstract updateChannelMediaRelayEx(
-    configuration: ChannelMediaRelayConfiguration,
-    connection: RtcConnection
-  ): number;
-
   /**
    * Stops the media stream relay. Once the relay stops, the host quits all the target channels.
    *
@@ -848,7 +802,7 @@ export abstract class IRtcEngineEx extends IRtcEngine {
    *  Low-quality video stream: Low bitrate, low resolution. This method is applicable to all types of streams from the sender, including but not limited to video streams collected from cameras, screen sharing streams, and custom-collected video streams.
    *
    * @param enabled Whether to enable dual-stream mode: true : Enable dual-stream mode. false : (Default) Disable dual-stream mode.
-   * @param streamConfig The configuration of the low-quality video stream. See SimulcastStreamConfig.
+   * @param streamConfig The configuration of the low-quality video stream. See SimulcastStreamConfig. When setting mode to DisableSimulcastStream, setting streamConfig will not take effect.
    * @param connection The connection information. See RtcConnection.
    *
    * @returns
@@ -864,13 +818,15 @@ export abstract class IRtcEngineEx extends IRtcEngine {
   /**
    * Sets the dual-stream mode on the sender side.
    *
-   * The SDK enables the low-quality video stream auto mode on the sender by default, which is equivalent to calling this method and setting the mode to AutoSimulcastStream. If you want to modify this behavior, you can call this method and modify the mode to DisableSimulcastStream (never send low-quality video streams) or EnableSimulcastStream (always send low-quality video streams). The difference and connection between this method and enableDualStreamModeEx is as follows:
+   * The SDK defaults to enabling low-quality video stream adaptive mode (AutoSimulcastStream) on the sender side, which means the sender does not actively send low-quality video stream. The receiver can initiate a low-quality video stream request by calling setRemoteVideoStreamTypeEx, and the sender will automatically start sending low-quality video stream upon receiving the request.
+   *  If you want to modify this behavior, you can call this method and set mode to DisableSimulcastStream (never send low-quality video streams) or EnableSimulcastStream (always send low-quality video streams).
+   *  If you want to restore the default behavior after making changes, you can call this method again with mode set to AutoSimulcastStream. The difference and connection between this method and enableDualStreamModeEx is as follows:
    *  When calling this method and setting mode to DisableSimulcastStream, it has the same effect as enableDualStreamModeEx (false).
    *  When calling this method and setting mode to EnableSimulcastStream, it has the same effect as enableDualStreamModeEx (true).
    *  Both methods can be called before and after joining a channel. If both methods are used, the settings in the method called later takes precedence.
    *
    * @param mode The mode in which the video stream is sent. See SimulcastStreamMode.
-   * @param streamConfig The configuration of the low-quality video stream. See SimulcastStreamConfig.
+   * @param streamConfig The configuration of the low-quality video stream. See SimulcastStreamConfig. When setting mode to DisableSimulcastStream, setting streamConfig will not take effect.
    * @param connection The connection information. See RtcConnection.
    *
    * @returns
@@ -949,4 +905,12 @@ export abstract class IRtcEngineEx extends IRtcEngine {
    *  < 0: Failure.
    */
   abstract startMediaRenderingTracingEx(connection: RtcConnection): number;
+
+  /**
+   * @ignore
+   */
+  abstract setParametersEx(
+    connection: RtcConnection,
+    parameters: string
+  ): number;
 }

package/ts/Private/IAgoraSpatialAudio.ts

@@ -62,78 +62,80 @@ export class SpatialAudioZone {
 }
 
 /**
- * This class contains some of the APIs in the ILocalSpatialAudioEngine class.
+ * This class calculates user positions through the SDK to implement the spatial audio effect.
  *
- * The ILocalSpatialAudioEngine class inherits from IBaseSpatialAudioEngine.
+ * This class inherits from IBaseSpatialAudioEngine. Before calling other APIs in this class, you need to call the initialize method to initialize this class.
  */
-export abstract class IBaseSpatialAudioEngine {
+export abstract class ILocalSpatialAudioEngine {
   /**
-   * Destroys IBaseSpatialAudioEngine.
-   *
-   * This method releases all resources under IBaseSpatialAudioEngine. When the user does not need to use the spatial audio effect, you can call this method to release resources for other operations. After calling this method, you can no longer use any of the APIs under IBaseSpatialAudioEngine. Call this method before the release method under IRtcEngine.
+   * @ignore
    */
   abstract release(): void;
 
   /**
-   * Sets the maximum number of streams that a user can receive in a specified audio reception range.
-   *
-   * If the number of receivable streams exceeds the set value, the local user receives the maxCount streams that are closest to the local user.
+   * Initializes ILocalSpatialAudioEngine.
    *
-   * @param maxCount The maximum number of streams that a user can receive within a specified audio reception range. The value of this parameter should be ≤ 16, and the default value is 10.
+   * Before calling other methods of the ILocalSpatialAudioEngine class, you need to call this method to initialize ILocalSpatialAudioEngine.
+   *  The SDK supports creating only one ILocalSpatialAudioEngine instance for an app.
    *
    * @returns
    * 0: Success.
    *  < 0: Failure.
    */
-  abstract setMaxAudioRecvCount(maxCount: number): number;
+  abstract initialize(): number;
 
   /**
-   * Sets the audio reception range of the local user.
+   * Updates the spatial position of the specified remote user.
    *
-   * After the setting is successful, the local user can only hear the remote users within the setting range or belonging to the same team. You can call this method at any time to update the audio reception range.
+   * After successfully calling this method, the SDK calculates the spatial audio parameters based on the relative position of the local and remote user. Call this method after joinChannel.
    *
-   * @param range The maximum audio reception range. The unit is meters. The value of this parameter must be greater than 0, and the default value is 20.
+   * @param uid The user ID. This parameter must be the same as the user ID passed in when the user joined the channel.
+   * @param posInfo The spatial position of the remote user. See RemoteVoicePositionInfo.
    *
    * @returns
    * 0: Success.
    *  < 0: Failure.
    */
-  abstract setAudioRecvRange(range: number): number;
+  abstract updateRemotePosition(
+    uid: number,
+    posInfo: RemoteVoicePositionInfo
+  ): number;
 
   /**
-   * Sets the length (in meters) of the game engine distance per unit.
-   *
-   * In a game engine, the unit of distance is customized, while in the Agora spatial audio algorithm, distance is measured in meters. By default, the SDK converts the game engine distance per unit to one meter. You can call this method to convert the game engine distance per unit to a specified number of meters.
-   *
-   * @param unit The number of meters that the game engine distance per unit is equal to. The value of this parameter must be greater than 0.00, and the default value is 1.00. For example, setting unit as 2.00 means the game engine distance per unit equals 2 meters. The larger the value is, the faster the sound heard by the local user attenuates when the remote user moves far away from the local user.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
+   * @ignore
    */
-  abstract setDistanceUnit(unit: number): number;
+  abstract updateRemotePositionEx(
+    uid: number,
+    posInfo: RemoteVoicePositionInfo,
+    connection: RtcConnection
+  ): number;
 
   /**
-   * Updates the spatial position of the local user.
+   * Removes the spatial position of the specified remote user.
    *
-   * Under the ILocalSpatialAudioEngine class, this method needs to be used with updateRemotePosition. The SDK calculates the relative position between the local and remote users according to this method and the parameter settings in updateRemotePosition, and then calculates the user's spatial audio effect parameters.
+   * After successfully calling this method, the local user no longer hears the specified remote user. After leaving the channel, to avoid wasting resources, you can also call this method to delete the spatial position of the specified remote user.
    *
-   * @param position The coordinates in the world coordinate system. This parameter is an array of length 3, and the three values represent the front, right, and top coordinates in turn.
-   * @param axisForward The unit vector of the x axis in the coordinate system. This parameter is an array of length 3, and the three values represent the front, right, and top coordinates in turn.
-   * @param axisRight The unit vector of the y axis in the coordinate system. This parameter is an array of length 3, and the three values represent the front, right, and top coordinates in turn.
-   * @param axisUp The unit vector of the z axis in the coordinate system. This parameter is an array of length 3, and the three values represent the front, right, and top coordinates in turn.
+   * @param uid The user ID. This parameter must be the same as the user ID passed in when the user joined the channel.
    *
    * @returns
    * 0: Success.
    *  < 0: Failure.
    */
-  abstract updateSelfPosition(
-    position: number[],
-    axisForward: number[],
-    axisRight: number[],
-    axisUp: number[]
+  abstract removeRemotePosition(uid: number): number;
+
+  /**
+   * @ignore
+   */
+  abstract removeRemotePositionEx(
+    uid: number,
+    connection: RtcConnection
   ): number;
 
+  /**
+   * @ignore
+   */
+  abstract clearRemotePositionsEx(connection: RtcConnection): number;
+
   /**
    * @ignore
    */
@@ -146,16 +148,32 @@ export abstract class IBaseSpatialAudioEngine {
   ): number;
 
   /**
-   * Updates the spatial position of the media player.
-   *
-   * After a successful update, the local user can hear the change in the spatial position of the media player.
-   *
-   * @param playerId The ID of the media player.
-   * @param positionInfo The spatial position of the media player. See RemoteVoicePositionInfo.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
+   * @ignore
+   */
+  abstract setMaxAudioRecvCount(maxCount: number): number;
+
+  /**
+   * @ignore
+   */
+  abstract setAudioRecvRange(range: number): number;
+
+  /**
+   * @ignore
+   */
+  abstract setDistanceUnit(unit: number): number;
+
+  /**
+   * @ignore
+   */
+  abstract updateSelfPosition(
+    position: number[],
+    axisForward: number[],
+    axisRight: number[],
+    axisUp: number[]
+  ): number;
+
+  /**
+   * @ignore
    */
   abstract updatePlayerPositionInfo(
     playerId: number,
@@ -168,156 +186,55 @@ export abstract class IBaseSpatialAudioEngine {
   abstract setParameters(params: string): number;
 
   /**
-   * Stops or resumes publishing the local audio stream.
-   *
-   * This method does not affect any ongoing audio recording, because it does not disable the audio capture device.
-   *  Call this method after joinChannel.
-   *  When using the spatial audio effect, if you need to set whether to stop subscribing to the audio stream of a specified user, Agora recommends calling this method instead of the muteLocalAudioStream method in IRtcEngine.
-   *  A successful call of this method triggers the onUserMuteAudio and onRemoteAudioStateChanged callbacks on the remote client.
-   *
-   * @param mute Whether to stop publishing the local audio stream: true : Stop publishing the local audio stream. false : Publish the local audio stream.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
+   * @ignore
    */
   abstract muteLocalAudioStream(mute: boolean): number;
 
   /**
-   * Stops or resumes subscribing to the audio streams of all remote users.
-   *
-   * After successfully calling this method, the local user stops or resumes subscribing to the audio streams of all remote users, including all subsequent users.
-   *  Call this method after joinChannel.
-   *  When using the spatial audio effect, if you need to set whether to stop subscribing to the audio streams of all remote users, Agora recommends calling this method instead of the muteAllRemoteAudioStreams method in IRtcEngine.
-   *  After calling this method, you need to call updateSelfPosition and updateRemotePosition to update the spatial location of the local user and the remote user; otherwise, the settings in this method do not take effect.
-   *
-   * @param mute Whether to stop subscribing to the audio streams of all remote users: true : Stop subscribing to the audio streams of all remote users. false : Subscribe to the audio streams of all remote users.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
+   * @ignore
    */
   abstract muteAllRemoteAudioStreams(mute: boolean): number;
 
   /**
-   * Sets the sound insulation area.
-   *
-   * In virtual interactive scenarios, you can use this method to set the sound insulation area and sound attenuation coefficient. When the sound source (which can be the user or the media player) and the listener belong to the inside and outside of the sound insulation area, they can experience the attenuation effect of sound similar to the real environment when it encounters a building partition.
-   *  When the sound source and the listener belong to the inside and outside of the sound insulation area, the sound attenuation effect is determined by the sound attenuation coefficient in SpatialAudioZone.
-   *  If the user or media player is in the same sound insulation area, it is not affected by SpatialAudioZone, and the sound attenuation effect is determined by the attenuation parameter in setPlayerAttenuation or setRemoteAudioAttenuation. If you do not call setPlayerAttenuation or setRemoteAudioAttenuation, the default sound attenuation coefficient of the SDK is 0.5, which simulates the attenuation of the sound in the real environment.
-   *  If the sound source and the receiver belong to two sound insulation areas, the receiver cannot hear the sound source. If this method is called multiple times, the last sound insulation area set takes effect.
-   *
-   * @param zones Sound insulation area settings. See SpatialAudioZone. On the Windows platform, it is necessary to ensure that the number of members in the zones array is equal to the value of zoneCount; otherwise, it may cause a crash.
-   * @param zoneCount The number of sound insulation areas.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
+   * @ignore
    */
-  abstract setZones(zones: SpatialAudioZone[], zoneCount: number): number;
+  abstract muteRemoteAudioStream(uid: number, mute: boolean): number;
 
   /**
-   * Sets the sound attenuation properties of the media player.
+   * Sets the sound attenuation effect for the specified user.
    *
-   * @param playerId The ID of the media player.
-   * @param attenuation The sound attenuation coefficient of the remote user or media player. The value range is [0,1]. The values are as follows:
+   * @param uid The user ID. This parameter must be the same as the user ID passed in when the user joined the channel.
+   * @param attenuation For the user's sound attenuation coefficient, the value range is [0,1]. The values are as follows:
    *  0: Broadcast mode, where the volume and timbre are not attenuated with distance, and the volume and timbre heard by local users do not change regardless of distance.
    *  (0,0.5): Weak attenuation mode, that is, the volume and timbre are only weakly attenuated during the propagation process, and the sound can travel farther than the real environment.
    *  0.5: (Default) simulates the attenuation of the volume in the real environment; the effect is equivalent to not setting the speaker_attenuation parameter.
    *  (0.5,1]: Strong attenuation mode, that is, the volume and timbre attenuate rapidly during the propagation process.
-   * @param forceSet Whether to force the sound attenuation effect of the media player: true : Force attenuation to set the attenuation of the media player. At this time, the attenuation coefficient of the sound insulation are set in the audioAttenuation in the SpatialAudioZone does not take effect for the media player. false : Do not force attenuation to set the sound attenuation effect of the media player, as shown in the following two cases.
+   * @param forceSet Whether to force the user's sound attenuation effect: true : Force attenuation to set the sound attenuation of the user. At this time, the attenuation coefficient of the sound insulation area set in the audioAttenuation of the SpatialAudioZone does not take effect for the user.
    *  If the sound source and listener are inside and outside the sound isolation area, the sound attenuation effect is determined by the audioAttenuation in SpatialAudioZone.
-   *  If the sound source and the listener are in the same sound insulation area or outside the same sound insulation area, the sound attenuation effect is determined by attenuation in this method.
+   *  If the sound source and the listener are in the same sound insulation area or outside the same sound insulation area, the sound attenuation effect is determined by attenuation in this method. false : Do not force attenuation to set the user's sound attenuation effect, as shown in the following two cases.
    *
    * @returns
    * 0: Success.
    *  < 0: Failure.
    */
-  abstract setPlayerAttenuation(
-    playerId: number,
+  abstract setRemoteAudioAttenuation(
+    uid: number,
     attenuation: number,
     forceSet: boolean
   ): number;
 
-  /**
-   * Stops or resumes subscribing to the audio stream of a specified user.
-   *
-   * Call this method after joinChannel.
-   *  When using the spatial audio effect, if you need to set whether to stop subscribing to the audio stream of a specified user, Agora recommends calling this method instead of the muteRemoteAudioStream method in IRtcEngine.
-   *
-   * @param uid The user ID. This parameter must be the same as the user ID passed in when the user joined the channel.
-   * @param mute Whether to subscribe to the specified remote user's audio stream. true : Stop subscribing to the audio stream of the specified user. false : (Default) Subscribe to the audio stream of the specified user. The SDK decides whether to subscribe according to the distance between the local user and the remote user.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
-   */
-  abstract muteRemoteAudioStream(uid: number, mute: boolean): number;
-}
-
-/**
- * This class calculates user positions through the SDK to implement the spatial audio effect.
- *
- * This class inherits from IBaseSpatialAudioEngine. Before calling other APIs in this class, you need to call the initialize method to initialize this class.
- */
-export abstract class ILocalSpatialAudioEngine extends IBaseSpatialAudioEngine {
-  /**
-   * Initializes ILocalSpatialAudioEngine.
-   *
-   * Before calling other methods of the ILocalSpatialAudioEngine class, you need to call this method to initialize ILocalSpatialAudioEngine.
-   *  The SDK supports creating only one ILocalSpatialAudioEngine instance for an app.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
-   */
-  abstract initialize(): number;
-
-  /**
-   * Updates the spatial position of the specified remote user.
-   *
-   * After successfully calling this method, the SDK calculates the spatial audio parameters based on the relative position of the local and remote user. Call this method after joinChannel.
-   *
-   * @param uid The user ID. This parameter must be the same as the user ID passed in when the user joined the channel.
-   * @param posInfo The spatial position of the remote user. See RemoteVoicePositionInfo.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
-   */
-  abstract updateRemotePosition(
-    uid: number,
-    posInfo: RemoteVoicePositionInfo
-  ): number;
-
   /**
    * @ignore
    */
-  abstract updateRemotePositionEx(
-    uid: number,
-    posInfo: RemoteVoicePositionInfo,
-    connection: RtcConnection
-  ): number;
-
-  /**
-   * Removes the spatial position of the specified remote user.
-   *
-   * After successfully calling this method, the local user no longer hears the specified remote user. After leaving the channel, to avoid wasting resources, you can also call this method to delete the spatial position of the specified remote user.
-   *
-   * @param uid The user ID. This parameter must be the same as the user ID passed in when the user joined the channel.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
-   */
-  abstract removeRemotePosition(uid: number): number;
+  abstract setZones(zones: SpatialAudioZone[], zoneCount: number): number;
 
   /**
    * @ignore
    */
-  abstract removeRemotePositionEx(
-    uid: number,
-    connection: RtcConnection
+  abstract setPlayerAttenuation(
+    playerId: number,
+    attenuation: number,
+    forceSet: boolean
   ): number;
 
   /**
@@ -330,32 +247,4 @@ export abstract class ILocalSpatialAudioEngine extends IBaseSpatialAudioEngine {
    *  < 0: Failure.
    */
   abstract clearRemotePositions(): number;
-
-  /**
-   * @ignore
-   */
-  abstract clearRemotePositionsEx(connection: RtcConnection): number;
-
-  /**
-   * Sets the sound attenuation effect for the specified user.
-   *
-   * @param uid The user ID. This parameter must be the same as the user ID passed in when the user joined the channel.
-   * @param attenuation For the user's sound attenuation coefficient, the value range is [0,1]. The values are as follows:
-   *  0: Broadcast mode, where the volume and timbre are not attenuated with distance, and the volume and timbre heard by local users do not change regardless of distance.
-   *  (0,0.5): Weak attenuation mode, that is, the volume and timbre are only weakly attenuated during the propagation process, and the sound can travel farther than the real environment.
-   *  0.5: (Default) simulates the attenuation of the volume in the real environment; the effect is equivalent to not setting the speaker_attenuation parameter.
-   *  (0.5,1]: Strong attenuation mode, that is, the volume and timbre attenuate rapidly during the propagation process.
-   * @param forceSet Whether to force the user's sound attenuation effect: true : Force attenuation to set the sound attenuation of the user. At this time, the attenuation coefficient of the sound insulation area set in the audioAttenuation of the SpatialAudioZone does not take effect for the user.
-   *  If the sound source and listener are inside and outside the sound isolation area, the sound attenuation effect is determined by the audioAttenuation in SpatialAudioZone.
-   *  If the sound source and the listener are in the same sound insulation area or outside the same sound insulation area, the sound attenuation effect is determined by attenuation in this method. false : Do not force attenuation to set the user's sound attenuation effect, as shown in the following two cases.
-   *
-   * @returns
-   * 0: Success.
-   *  < 0: Failure.
-   */
-  abstract setRemoteAudioAttenuation(
-    uid: number,
-    attenuation: number,
-    forceSet: boolean
-  ): number;
 }

package/ts/Private/IAudioDeviceManager.ts

@@ -63,12 +63,23 @@ export abstract class IAudioDeviceManager {
   abstract getPlaybackDeviceInfo(): AudioDeviceInfo;
 
   /**
-   * @ignore
+   * Sets the volume of the audio playback device.
+   *
+   * This method applies to Windows only.
+   *
+   * @param volume The volume of the audio playback device. The value range is [0,255].
+   *
+   * @returns
+   * 0: Success.
+   *  < 0: Failure.
    */
   abstract setPlaybackDeviceVolume(volume: number): number;
 
   /**
-   * @ignore
+   * Retrieves the volume of the audio playback device.
+   *
+   * @returns
+   * The volume of the audio playback device. The value range is [0,255].
    */
   abstract getPlaybackDeviceVolume(): number;
 
@@ -115,7 +126,12 @@ export abstract class IAudioDeviceManager {
   abstract setRecordingDeviceVolume(volume: number): number;
 
   /**
-   * @ignore
+   * Retrieves the volume of the audio recording device.
+   *
+   * This method applies to Windows only.
+   *
+   * @returns
+   * The volume of the audio recording device. The value range is [0,255].
    */
   abstract getRecordingDeviceVolume(): number;
 
@@ -177,8 +193,7 @@ export abstract class IAudioDeviceManager {
   /**
    * Starts the audio playback device test.
    *
-   * This method tests whether the audio playback device works properly. Once a user starts the test, the SDK plays an audio file specified by the user. If the user can hear the audio, the playback device works properly. After calling this method, the SDK triggers the onAudioVolumeIndication callback every 100 ms, reporting uid = 1 and the volume information of the playback device.
-   *  Ensure that you call this method before joining a channel.
+   * This method tests whether the audio device for local playback works properly. Once a user starts the test, the SDK plays an audio file specified by the user. If the user can hear the audio, the playback device works properly. After calling this method, the SDK triggers the onAudioVolumeIndication callback every 100 ms, reporting uid = 1 and the volume information of the playback device. The difference between this method and the startEchoTest method is that the former checks if the local audio playback device is working properly, while the latter can check the audio and video devices and network conditions. Ensure that you call this method before joining a channel. After the test is completed, call stopPlaybackDeviceTest to stop the test before joining a channel.
    *
    * @param testAudioFilePath The path of the audio file. The data format is string in UTF-8.
    *  Supported file formats: wav, mp3, m4a, and aac.
@@ -193,8 +208,7 @@ export abstract class IAudioDeviceManager {
   /**
    * Stops the audio playback device test.
    *
-   * This method stops the audio playback device test. You must call this method to stop the test after calling the startPlaybackDeviceTest method.
-   *  Ensure that you call this method before joining a channel.
+   * This method stops the audio playback device test. You must call this method to stop the test after calling the startPlaybackDeviceTest method. Ensure that you call this method before joining a channel.
    *
    * @returns
    * 0: Success.
@@ -203,24 +217,23 @@ export abstract class IAudioDeviceManager {
   abstract stopPlaybackDeviceTest(): number;
 
   /**
-   * Starts the audio capture device test.
+   * Starts the audio capturing device test.
    *
-   * This method tests whether the audio capture device works properly. After calling this method, the SDK triggers the onAudioVolumeIndication callback at the time interval set in this method, which reports uid = 0 and the volume information of the capturing device.
-   *  Ensure that you call this method before joining a channel.
+   * This method tests whether the audio capturing device works properly. After calling this method, the SDK triggers the onAudioVolumeIndication callback at the time interval set in this method, which reports uid = 0 and the volume information of the capturing device. The difference between this method and the startEchoTest method is that the former checks if the local audio capturing device is working properly, while the latter can check the audio and video devices and network conditions. Ensure that you call this method before joining a channel. After the test is completed, call stopRecordingDeviceTest to stop the test before joining a channel.
    *
-   * @param indicationInterval The time interval (ms) at which the SDK triggers the onAudioVolumeIndication callback. Agora recommends setting a value greater than 200 ms. This value must not be less than 10 ms; otherwise, you can not receive the onAudioVolumeIndication callback.
+   * @param indicationInterval The interval (ms) for triggering the onAudioVolumeIndication callback. This value should be set to greater than 10, otherwise, you will not receive the onAudioVolumeIndication callback and the SDK returns the error code -2. Agora recommends that you set this value to 100.
    *
    * @returns
    * 0: Success.
    *  < 0: Failure.
+   *  -2: Invalid parameters. Check your parameter settings.
    */
   abstract startRecordingDeviceTest(indicationInterval: number): number;
 
   /**
-   * Stops the audio capture device test.
+   * Stops the audio capturing device test.
    *
-   * This method stops the audio capture device test. You must call this method to stop the test after calling the startRecordingDeviceTest method.
-   *  Ensure that you call this method before joining a channel.
+   * This method stops the audio capturing device test. You must call this method to stop the test after calling the startRecordingDeviceTest method. Ensure that you call this method before joining a channel.
    *
    * @returns
    * 0: Success.