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2 changes: 1 addition & 1 deletion src/RtcLocalView.tsx
Original file line number Diff line number Diff line change
Expand Up @@ -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
*/
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5 changes: 3 additions & 2 deletions src/src/Classes.ts
Original file line number Diff line number Diff line change
Expand Up @@ -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
/**
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4 changes: 2 additions & 2 deletions src/src/Enums.ts
Original file line number Diff line number Diff line change
Expand Up @@ -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,
}
4 changes: 2 additions & 2 deletions src/src/RtcEngine.native.ts
Original file line number Diff line number Diff line change
Expand Up @@ -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)
Expand All @@ -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
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32 changes: 22 additions & 10 deletions src/src/RtcEvents.ts
Original file line number Diff line number Diff line change
Expand Up @@ -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 =
Expand Down Expand Up @@ -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 =
Expand All @@ -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 =
Expand Down Expand Up @@ -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

Expand Down Expand Up @@ -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.
Expand Down