diff --git a/src/RtcLocalView.tsx b/src/RtcLocalView.tsx
index d1eaa962d..5b4c69949 100644
--- a/src/RtcLocalView.tsx
+++ b/src/RtcLocalView.tsx
@@ -8,7 +8,7 @@ import {RtcSurfaceView, RtcSurfaceViewProps, RtcTextureView, RtcTextureViewProps
  *
  * **Note**
  *
- * SurfaceView is supported on Android only. Use UIView on iOS.
+ * SurfaceView is supported on Android only. Use [UIView](https://developer.apple.com/documentation/uikit/uiview) on iOS.
  *
  * @noInheritDoc
  */
diff --git a/src/src/Classes.ts b/src/src/Classes.ts
index 59b8e56a1..0b55e9b73 100644
--- a/src/src/Classes.ts
+++ b/src/src/Classes.ts
@@ -80,9 +80,10 @@ export class VideoEncoderConfiguration {
      *
      * **Note**
      * <ul>
-     *     <li>The value of the dimension does not indicate the orientation mode of the output ratio. For how to set the video orientation, see [`VideoOutputOrientationMode`]{@link VideoOutputOrientationMode}.</li>
-     *     <li>Whether 720p+ can be supported depends on the device. If the device cannot support 720p, the frame rate will be lower than the one listed in the table.</li>
+     *    <li> The value of the dimension does not indicate the orientation mode of the output ratio. For how to set the video orientation, see [<code>VideoOutputOrientationMode</code>]{@link VideoOutputOrientationMode}.</li>
+     *    <li> Whether 720p+ can be supported depends on the device. If the device cannot support 720p, the frame rate will be lower than the one listed in the table.</li>
      * </ul>
+     *
      */
     dimensions?: VideoDimensions
     /**
diff --git a/src/src/Enums.ts b/src/src/Enums.ts
index 771786e42..2a3ee3905 100644
--- a/src/src/Enums.ts
+++ b/src/src/Enums.ts
@@ -2331,11 +2331,11 @@ export enum StreamSubscribeState {
 }
 
 /**
- * TODO(doc)
+ * Events during the RTMP streaming.
  */
 export enum RtmpStreamingEvent {
     /**
-     * TODO(doc)
+     * 1: An error occurs when you add a background image or a watermark image to the RTMP stream.
      */
     FailedLoadImage = 1,
 }
diff --git a/src/src/RtcEngine.native.ts b/src/src/RtcEngine.native.ts
index 602ddc23c..0873a141c 100644
--- a/src/src/RtcEngine.native.ts
+++ b/src/src/RtcEngine.native.ts
@@ -99,7 +99,7 @@ export default class RtcEngine implements RtcEngineInterface {
      * call `create` to create an [`RtcEngine`]{@link RtcEngine} instance with the new App ID.
      * @returns
      * - The `RtcEngine` instance, if the method call succeeds.
-     * - < 0, if the method call fails.
+     * - The error code, if the method call fails.
      */
     static async create(appId: string): Promise<RtcEngine> {
         return RtcEngine.createWithAreaCode(appId, AreaCode.GLOB)
@@ -125,7 +125,7 @@ export default class RtcEngine implements RtcEngineInterface {
      *
      * @returns
      * - The `RtcEngine` instance, if the method call succeeds.
-     * - < 0, if the method call fails.
+     * - The error code, if the method call fails.
      */
     static async createWithAreaCode(appId: string, areaCode: AreaCode): Promise<RtcEngine> {
         if (engine) return engine
diff --git a/src/src/RtcEvents.ts b/src/src/RtcEvents.ts
index 596b7ea4b..3cfc4c93b 100644
--- a/src/src/RtcEvents.ts
+++ b/src/src/RtcEvents.ts
@@ -107,9 +107,20 @@ export type ClientRoleCallback =
     (oldRole: ClientRole, newRole: ClientRole) => void
 export type UidWithElapsedCallback =
 /**
- * @param uid ID of the user or host who joins the channel.
- * @param elapsed Time delay (ms) from the local user calling [`joinChannel`]{@link RtcEngine.joinChannel} or [`setClientRole`]{@link RtcEngine.setClientRole}
+ * @param uid This parameter has the following definitions in different events:
+ * - [`UserJoined`]{@link RtcEngineEvents.UserJoined}: ID of the user or host who joins the channel.
+ * - [`FirstRemoteAudioFrame`]{@link RtcEngineEvents.FirstRemoteAudioFrame}: User ID of the remote user.
+ * - [`FirstRemoteAudioDecoded`]{@link RtcEngineEvents.FirstRemoteAudioDecoded}: User ID of the remote user sending the audio stream.
+ * - [`JoinChannelSuccess`]{@link RtcChannelEvents.JoinChannelSuccess}: User ID.
+ * - [`RejoinChannelSuccess`]{@link RtcChannelEvents.RejoinChannelSuccess}: User ID.
+ * @param elapsed This parameter has the following definitions in different events:
+ * - [`UserJoined`]{@link RtcEngineEvents.UserJoined}: Time delay (ms) from the local user calling [`joinChannel`]{@link RtcEngine.joinChannel} or [`setClientRole`]{@link RtcEngine.setClientRole}
  * until this callback is triggered.
+ * - [`FirstRemoteAudioFrame`]{@link RtcEngineEvents.FirstRemoteAudioFrame}: Time elapsed (ms) from the local user calling [`joinChannel`]{@link RtcEngine.joinChannel} until this callback is triggered.
+ * - [`FirstRemoteAudioDecoded`]{@link RtcEngineEvents.FirstRemoteAudioDecoded}: Time elapsed (ms) from the local user calling [`joinChannel`]{@link RtcEngine.joinChannel} until the SDK triggers this callback.
+ * - [`JoinChannelSuccess`]{@link RtcChannelEvents.JoinChannelSuccess}: Time elapsed (ms) from the local user calling [`joinChannel`]{@link RtcChannel.joinChannel} until this callback is triggered.
+ * - [`RejoinChannelSuccess`]{@link RtcChannelEvents.RejoinChannelSuccess}: Time elapsed (ms) from the local user starting to reconnect until this callback is triggered.
+ *
  */
     (uid: number, elapsed: number) => void
 export type UserOfflineCallback =
@@ -369,10 +380,10 @@ export type UrlCallback =
     (url: string) => void
 export type TransportStatsCallback =
 /**
- * @param uid User ID of the remote user sending the audio packet.
- * @param delay Network time delay (ms) from the remote user sending the audio packet to the local user.
- * @param lost Packet loss rate (%) of the audio packet sent from the remote user.
- * @param rxKBitRate Received bitrate (Kbps) of the audio packet sent from the remote user.
+ * @param uid User ID of the remote user sending the audio packet/video packet.
+ * @param delay Network time delay (ms) from the remote user sending the audio packet/video packet to the local user.
+ * @param lost Packet loss rate (%) of the audio packet/video packet sent from the remote user.
+ * @param rxKBitRate Received bitrate (Kbps) of the audio packet/video packet sent from the remote user.
  */
     (uid: number, delay: number, lost: number, rxKBitRate: number) => void
 export type UidWithEnabledCallback =
@@ -388,8 +399,8 @@ export type UidWithEnabledCallback =
 export type EnabledCallback =
 /**
  * @param enabled Whether the microphone is enabled/disabled:
- *  - `true`：Enabled.
- *  - `false`：Disabled.
+ *  - `true`: Enabled.
+ *  - `false`: Disabled.
  */
     (enabled: boolean) => void
 export type AudioQualityCallback =
@@ -435,7 +446,8 @@ export type StreamSubscribeStateCallback =
     (channel: string, oldState: StreamSubscribeState, newState: StreamSubscribeState, elapseSinceLastState: number) => void
 export type RtmpStreamingEventCallback =
 /**
- * TODO(doc)
+ * @param url The RTMP streaming URL.
+ * @param eventCode The event code. See [`RtmpStreamingEvent`]{@link RtmpStreamingEvent}.
  */
     (url: string, eventCode: RtmpStreamingEvent) => void
 
@@ -1027,7 +1039,7 @@ export interface RtcEngineEvents {
      *
      * Use [`Decoding`]{@link VideoRemoteState.Decoding} in [`RemoteAudioStateChanged`]{@link RemoteAudioStateChanged} instead.
      *
-     * This callback is triggered in either of the following scenarios：
+     * This callback is triggered in either of the following scenarios:
      * - The remote user joins the channel and sends the audio stream.
      * - The remote user stops sending the audio stream and re-sends it after 15 seconds. Possible reasons include:
      *  - The remote user leaves channel.