@tencent-rtc/mcp 1.6.4 → 1.6.6

package/resource/knowledge/trtc/android/TRTCCloud.md

@@ -4,7 +4,7 @@ Module:   TRTCCloud @ TXLiteAVSDK
 
 Function: TRTC's main feature API
 
-Version: 13.3
+Version: 13.4
 
 ## TRTCCloud
 
@@ -120,11 +120,10 @@ Version: 13.3
 
 |Param|DESC|
 |---------|---------|
-|context|It is only applicable to the Android platform. The SDK internally converts it into the ` ApplicationContext ` of Android to call the Android system API.|
+|context|It is only applicable to the Android platform. The SDK internally converts it into the  `ApplicationContext`  of Android to call the Android system API.|
 
 > **Note**
 >
-
 > 1. Please use destroySharedInstance to release the object pointer.
 >
 
@@ -146,9 +145,9 @@ You can use TRTCCloudListener to get various event notifications from the SDK, s
 
 **Set the queue that drives the TRTCCloudListener event callback.**
 
-If you do not specify a ` listenerHandler `, the SDK will use ` MainQueue ` as the queue for driving TRTCCloudListener event callbacks by default.
+If you do not specify a  `listenerHandler` , the SDK will use  `MainQueue`  as the queue for driving TRTCCloudListener event callbacks by default.
 
-In other words, if you do not set the ` listenerHandler ` attribute, all callback functions in TRTCCloudListener will be driven by ` MainQueue `.
+In other words, if you do not set the  `listenerHandler`  attribute, all callback functions in TRTCCloudListener will be driven by  `MainQueue` .
 
 |Param|DESC|
 |---------|---------|
@@ -156,8 +155,7 @@ In other words, if you do not set the ` listenerHandler ` attribute, all callbac
 
 > **Note**
 >
-
-> If you specify a ` listenerHandler `, please do not manipulate the UI in the TRTCCloudListener callback function; otherwise, thread safety issues will occur.
+> If you specify a  `listenerHandler` , please do not manipulate the UI in the TRTCCloudListener callback function; otherwise, thread safety issues will occur.
 >
 
 ## enterRoom
@@ -167,31 +165,31 @@ In other words, if you do not set the ` listenerHandler ` attribute, all callbac
 All TRTC users need to enter a room before they can "publish" or "subscribe to" audio/video streams. "Publishing" refers to pushing their own streams to the cloud, and "subscribing to" refers to pulling the streams of other users in the room from the cloud.
 
 When calling this API, you need to specify your application scenario (TRTCAppScene) to get the best audio/video transfer experience. We provide the following four scenarios for your choice:
-- TRTC_APP_SCENE_VIDEOCALL:
+-  TRTC_APP_SCENE_VIDEOCALL:
 
 Video call scenario. Use cases: [one-to-one video call], [video conferencing with up to 300 participants], [online medical diagnosis], [small class], [video interview], etc.
 
 In this scenario, each room supports up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
-- TRTC_APP_SCENE_AUDIOCALL:
+-  TRTC_APP_SCENE_AUDIOCALL:
 
 Audio call scenario. Use cases: [one-to-one audio call], [audio conferencing with up to 300 participants], [audio chat], [online Werewolf], etc.
 
 In this scenario, each room supports up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
-- TRTC_APP_SCENE_LIVE:
+-  TRTC_APP_SCENE_LIVE:
 
 Live streaming scenario. Use cases: [low-latency video live streaming], [interactive classroom for up to 100,000 participants], [live video competition], [video dating room], [remote training], [large-scale conferencing], etc.
 
 In this scenario, each room supports up to 100,000 concurrent online users, but you should specify the user roles: anchor (TRTCRoleAnchor) or audience (TRTCRoleAudience).
-- TRTC_APP_SCENE_VOICE_CHATROOM:
+-  TRTC_APP_SCENE_VOICE_CHATROOM:
 
 Audio chat room scenario. Use cases: [Clubhouse], [online karaoke room], [music live room], [FM radio], etc.
 
 In this scenario, each room supports up to 100,000 concurrent online users, but you should specify the user roles: anchor (TRTCRoleAnchor) or audience (TRTCRoleAudience).
 
-After calling this API, you will receive the ` onEnterRoom(result) ` callback from TRTCCloudListener:
--  If room entry succeeded, the ` result ` parameter will be a positive number (` result ` > 0), indicating the time in milliseconds (ms) between function call and room entry.
+After calling this API, you will receive the  `onEnterRoom(result)`  callback from TRTCCloudListener:
+-  If room entry succeeded, the  `result`  parameter will be a positive number ( `result`  > 0), indicating the time in milliseconds (ms) between function call and room entry.
 
--  If room entry failed, the ` result ` parameter will be a negative number (` result ` < 0), indicating the TXLiteAVError for room entry failure.
+-  If room entry failed, the  `result`  parameter will be a negative number ( `result`  < 0), indicating the TXLiteAVError for room entry failure.
 
 |Param|DESC|
 |---------|---------|
@@ -200,13 +198,10 @@ After calling this API, you will receive the ` onEnterRoom(result) ` callback fr
 
 > **Note**
 >
-
-> 1. If ` scene ` is specified as TRTC_APP_SCENE_LIVE or TRTC_APP_SCENE_VOICE_CHATROOM, you must use the ` role ` field in TRTCParams to specify the role of the current user in the room.
+> 1. If  `scene`  is specified as TRTC_APP_SCENE_LIVE or TRTC_APP_SCENE_VOICE_CHATROOM, you must use the  `role`  field in TRTCParams to specify the role of the current user in the room.
 >
-
-> 2. The same ` scene ` should be configured for all users in the same room.
+> 2. The same  `scene`  should be configured for all users in the same room.
 >
-
 > 3. Please try to ensure that enterRoom and exitRoom are used in pair; that is, please make sure that "the previous room is exited before the next room is entered"; otherwise, many issues may occur.
 >
 
@@ -216,7 +211,7 @@ After calling this API, you will receive the ` onEnterRoom(result) ` callback fr
 
 Calling this API will allow the user to leave the current audio or video room and release the camera, mic, speaker, and other device resources.
 
-After resources are released, the SDK will use the ` onExitRoom() ` callback in TRTCCloudListener to notify you.
+After resources are released, the SDK will use the  `onExitRoom()`  callback in TRTCCloudListener to notify you.
 
 If you need to call enterRoom again or switch to the SDK of another provider, we recommend you wait until you receive the onExitRoom callback, so as to avoid the problem of the camera or mic being occupied.
 
@@ -224,45 +219,41 @@ If you need to call enterRoom again or switch to the SDK of another provider, we
 
 **Switch role.**
 
-This API is used to switch the user role between ` anchor ` and ` audience `.
+This API is used to switch the user role between  `anchor`  and  `audience` .
 
 As video live rooms and audio chat rooms need to support an audience of up to 100,000 concurrent online users, the rule "only anchors can publish their audio/video streams" has been set. Therefore, when some users want to publish their streams (so that they can interact with anchors), they need to switch their role to "anchor" first.
 
-You can use the ` role ` field in TRTCParams during room entry to specify the user role in advance or use the ` switchRole ` API to switch roles after room entry.
+You can use the  `role`  field in TRTCParams during room entry to specify the user role in advance or use the  `switchRole`  API to switch roles after room entry.
 | Param | DESC |
 | --- | --- |
-| role | Role, which is <code> anchor </code> by default:<br>- TRTCRoleAnchor: anchor, who can publish their audio/video streams. Up to 50 anchors are allowed to publish streams at the same time in one room.<br>- TRTCRoleAudience: audience, who cannot publish their audio/video streams, but can only watch streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through switchRole. One room supports an audience of up to 100,000 concurrent online users. |
+| role | Role, which is  <code>anchor</code>  by default:<br>-  TRTCRoleAnchor: anchor, who can publish their audio/video streams. Up to 50 anchors are allowed to publish streams at the same time in one room.<br>-  TRTCRoleAudience: audience, who cannot publish their audio/video streams, but can only watch streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through switchRole. One room supports an audience of up to 100,000 concurrent online users. |
 
 > **Note**
 >
-
 > 1. This API is only applicable to two scenarios: live streaming (TRTC_APP_SCENE_LIVE) and audio chat room (TRTC_APP_SCENE_VOICE_CHATROOM).
 >
-
-> 2. If the ` scene ` you specify in enterRoom is TRTC_APP_SCENE_VIDEOCALL or TRTC_APP_SCENE_AUDIOCALL, please do not call this API.
+> 2. If the  `scene`  you specify in enterRoom is TRTC_APP_SCENE_VIDEOCALL or TRTC_APP_SCENE_AUDIOCALL, please do not call this API.
 >
 
 ## switchRole
 
 **Switch role(support permission credential).**
 
-This API is used to switch the user role between ` anchor ` and ` audience `.
+This API is used to switch the user role between  `anchor`  and  `audience` .
 
 As video live rooms and audio chat rooms need to support an audience of up to 100,000 concurrent online users, the rule "only anchors can publish their audio/video streams" has been set. Therefore, when some users want to publish their streams (so that they can interact with anchors), they need to switch their role to "anchor" first.
 
-You can use the ` role ` field in TRTCParams during room entry to specify the user role in advance or use the ` switchRole ` API to switch roles after room entry.
+You can use the  `role`  field in TRTCParams during room entry to specify the user role in advance or use the  `switchRole`  API to switch roles after room entry.
 | Param | DESC |
 | --- | --- |
-| privateMapKey | Permission credential used for permission control. If you want only users with the specified <code> userId </code> values to enter a room or push streams, you need to use <code> privateMapKey </code> to restrict the permission.<br>-  We recommend you use this parameter only if you have high security requirements. For more information, please see <a href="https://www.tencentcloud.com/document/product/647/35157">Enabling Advanced Permission Control</a>. |
-| role | Role, which is <code> anchor </code> by default:<br>- TRTCRoleAnchor: anchor, who can publish their audio/video streams. Up to 50 anchors are allowed to publish streams at the same time in one room.<br>- TRTCRoleAudience: audience, who cannot publish their audio/video streams, but can only watch streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through switchRole. One room supports an audience of up to 100,000 concurrent online users. |
+| privateMapKey | Permission credential used for permission control. If you want only users with the specified  <code>userId</code>  values to enter a room or push streams, you need to use  <code>privateMapKey</code>  to restrict the permission.<br>-  We recommend you use this parameter only if you have high security requirements. For more information, please see <a href="https://www.tencentcloud.com/document/product/647/35157">Enabling Advanced Permission Control</a>. |
+| role | Role, which is  <code>anchor</code>  by default:<br>-  TRTCRoleAnchor: anchor, who can publish their audio/video streams. Up to 50 anchors are allowed to publish streams at the same time in one room.<br>-  TRTCRoleAudience: audience, who cannot publish their audio/video streams, but can only watch streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through switchRole. One room supports an audience of up to 100,000 concurrent online users. |
 
 > **Note**
 >
-
 > 1. This API is only applicable to two scenarios: live streaming (TRTC_APP_SCENE_LIVE) and audio chat room (TRTC_APP_SCENE_VOICE_CHATROOM).
 >
-
-> 2. If the ` scene ` you specify in enterRoom is TRTC_APP_SCENE_VIDEOCALL or TRTC_APP_SCENE_AUDIOCALL, please do not call this API.
+> 2. If the  `scene`  you specify in enterRoom is TRTC_APP_SCENE_VIDEOCALL or TRTC_APP_SCENE_AUDIOCALL, please do not call this API.
 >
 
 ## switchRoom
@@ -270,13 +261,13 @@ You can use the ` role ` field in TRTCParams during room entry to specify the us
 **Switch room.**
 
 This API is used to quickly switch a user from one room to another.
--  If the user's role is ` audience `, calling this API is equivalent to ` exitRoom ` (current room) + ` enterRoom ` (new room).
+-  If the user's role is  `audience` , calling this API is equivalent to  `exitRoom`  (current room) +  `enterRoom`  (new room).
 
--  If the user's role is ` anchor `, the API will retain the current audio/video publishing status while switching the room; therefore, during the room switch, camera preview and sound capturing will not be interrupted.
+-  If the user's role is  `anchor` , the API will retain the current audio/video publishing status while switching the room; therefore, during the room switch, camera preview and sound capturing will not be interrupted.
 
-This API is suitable for the online education scenario where the supervising teacher can perform fast room switch across multiple rooms. In this scenario, using ` switchRoom ` can get better smoothness and use less code than ` exitRoom + enterRoom `.
+This API is suitable for the online education scenario where the supervising teacher can perform fast room switch across multiple rooms. In this scenario, using  `switchRoom`  can get better smoothness and use less code than  `exitRoom + enterRoom` .
 
-The API call result will be called back through ` onSwitchRoom(errCode, errMsg) ` in TRTCCloudListener.
+The API call result will be called back through  `onSwitchRoom(errCode, errMsg)`  in TRTCCloudListener.
 
 |Param|DESC|
 |---------|---------|
@@ -284,14 +275,11 @@ The API call result will be called back through ` onSwitchRoom(errCode, errMsg)
 
 > **Note**
 >
-
-> Due to the requirement for compatibility with legacy versions of the SDK, the ` config ` parameter contains both ` roomId ` and ` strRoomId ` parameters. You should pay special attention as detailed below when specifying these two parameters:
+> Due to the requirement for compatibility with legacy versions of the SDK, the  `config`  parameter contains both  `roomId`  and  `strRoomId`  parameters. You should pay special attention as detailed below when specifying these two parameters:
 >
-
-> 1. If you decide to use ` strRoomId `, then set ` roomId ` to 0. If both are specified, ` roomId ` will be used.
+> 1. If you decide to use  `strRoomId` , then set  `roomId`  to 0. If both are specified,  `roomId`  will be used.
 >
-
-> 2. All rooms need to use either ` strRoomId ` or ` roomId ` at the same time. They cannot be mixed; otherwise, there will be many unexpected bugs.
+> 2. All rooms need to use either  `strRoomId`  or  `roomId`  at the same time. They cannot be mixed; otherwise, there will be many unexpected bugs.
 >
 
 ## ConnectOtherRoom
@@ -306,10 +294,10 @@ In other words, you can use this API to share the audio/video streams of two anc
 
 The result of requesting cross-room call will be returned through the onConnectOtherRoom callback in TRTCCloudListener.
 
-For example, after anchor A in room "101" uses ` connectOtherRoom() ` to successfully call anchor B in room "102":
--  All users in room "101" will receive the ` onRemoteUserEnterRoom(B) ` and ` onUserVideoAvailable(B,true) ` event callbacks of anchor B; that is, all users in room "101" can subscribe to the audio/video streams of anchor B.
+For example, after anchor A in room "101" uses  `connectOtherRoom()`  to successfully call anchor B in room "102":
+-  All users in room "101" will receive the  `onRemoteUserEnterRoom(B)`  and  `onUserVideoAvailable(B,true)`  event callbacks of anchor B; that is, all users in room "101" can subscribe to the audio/video streams of anchor B.
 
--  All users in room "102" will receive the ` onRemoteUserEnterRoom(A) ` and ` onUserVideoAvailable(A,true) ` event callbacks of anchor A; that is, all users in room "102" can subscribe to the audio/video streams of anchor A.
+-  All users in room "102" will receive the  `onRemoteUserEnterRoom(A)`  and  `onUserVideoAvailable(A,true)`  event callbacks of anchor A; that is, all users in room "102" can subscribe to the audio/video streams of anchor A.
 
 For compatibility with subsequent extended fields for cross-room call, parameters in JSON format are used currently.
 
@@ -327,7 +315,7 @@ Below is the sample code:
 
 Case 2: string room ID
 
-If you use a string room ID, please be sure to replace the ` roomId ` in JSON with ` strRoomId `, such as ` {"strRoomId": "102", "userId": "userB"} `
+If you use a string room ID, please be sure to replace the  `roomId`  in JSON with  `strRoomId` , such as  `{"strRoomId": "102", "userId": "userB"}`
 
 Below is the sample code:
 ``` java
@@ -339,20 +327,20 @@ Below is the sample code:
 
 |Param|DESC|
 |---------|---------|
-|param|You need to pass in a string parameter in JSON format: ` roomId ` represents the room ID in numeric format, ` strRoomId ` represents the room ID in string format, and ` userId ` represents the user ID of the target anchor.|
+|param|You need to pass in a string parameter in JSON format:  `roomId`  represents the room ID in numeric format,  `strRoomId`  represents the room ID in string format, and  `userId`  represents the user ID of the target anchor.|
 
 ## DisconnectOtherRoom
 
 **Exit cross-room call.**
 
-The result will be returned through the ` onDisconnectOtherRoom() ` callback in TRTCCloudListener.
+The result will be returned through the  `onDisconnectOtherRoom()`  callback in TRTCCloudListener.
 
 ## setDefaultStreamRecvMode
 
 **Set subscription mode (which must be set before room entry for it to take effect).**
 
 You can switch between the "automatic subscription" and "manual subscription" modes through this API:
--  Automatic subscription: this is the default mode, where the user will immediately receive the audio/video streams in the room after room entry, so that the audio will be automatically played back, and the video will be automatically decoded (you still need to bind the rendering control through the ` startRemoteView ` API).
+-  Automatic subscription: this is the default mode, where the user will immediately receive the audio/video streams in the room after room entry, so that the audio will be automatically played back, and the video will be automatically decoded (you still need to bind the rendering control through the  `startRemoteView`  API).
 
 -  Manual subscription: after room entry, the user needs to manually call the startRemoteView API to start subscribing to and decoding the video stream and call the muteRemoteAudio (false) API to start playing back the audio stream.
 
@@ -362,15 +350,13 @@ In your application scenario, if there are many audio/video streams being publis
 
 |Param|DESC|
 |---------|---------|
-|autoRecvAudio|true: automatic subscription to audio; false: manual subscription to audio by calling ` muteRemoteAudio(false) `. Default value: true|
-|autoRecvVideo|true: automatic subscription to video; false: manual subscription to video by calling ` startRemoteView `. Default value: true|
+|autoRecvAudio|true: automatic subscription to audio; false: manual subscription to audio by calling  `muteRemoteAudio(false)` . Default value: true|
+|autoRecvVideo|true: automatic subscription to video; false: manual subscription to video by calling  `startRemoteView` . Default value: true|
 
 > **Note**
 >
-
 > 1. The configuration takes effect only if this API is called before room entry (enterRoom).
 >
-
 > 2. In the automatic subscription mode, if the user does not call startRemoteView to subscribe to the video stream after room entry, the SDK will automatically stop subscribing to the video stream in order to reduce the traffic consumption.
 >
 
@@ -378,11 +364,11 @@ In your application scenario, if there are many audio/video streams being publis
 
 **Create room subinstance (for concurrent multi-room listen/watch).**
 
-` TRTCCloud ` was originally designed to work in the singleton mode, which limited the ability to watch concurrently in multiple rooms.
+ `TRTCCloud`  was originally designed to work in the singleton mode, which limited the ability to watch concurrently in multiple rooms.
 
-By calling this API, you can create multiple ` TRTCCloud ` instances, so that you can enter multiple different rooms at the same time to listen/watch audio/video streams.
+By calling this API, you can create multiple  `TRTCCloud`  instances, so that you can enter multiple different rooms at the same time to listen/watch audio/video streams.
 
-However, it should be noted that your ability to publish audio and video streams in multiple ` TRTCCloud ` instances will be limited.
+However, it should be noted that your ability to publish audio and video streams in multiple  `TRTCCloud`  instances will be limited.
 
 This feature is mainly used in the "super small class" use case in the online education scenario to break the limit that "only up to 50 users can publish their audio/video streams simultaneously in one TRTC room".
 
@@ -414,23 +400,20 @@ Below is the sample code:
 
 > **Note**
 >
-
-> 1. The same user can enter multiple rooms with different ` roomId ` values by using the same ` userId `.
+> 1. The same user can enter multiple rooms with different  `roomId`  values by using the same  `userId` .
 >
-
-> 2. Two devices cannot use the same ` userId ` to enter the same room with a specified ` roomId `.
+> 2. Two devices cannot use the same  `userId`  to enter the same room with a specified  `roomId` .
 >
-
 > 3. You can set TRTCCloudListener separately for different instances to get their own event notifications.
 >
-
-> 4. The same user can push streams in multiple ` TRTCCloud ` instances at the same time, and can also call APIs related to local audio/video in the sub instance. But need to pay attention to:
+> 4. The same user can push streams in multiple  `TRTCCloud`  instances at the same time, and can also call APIs related to local audio/video in the sub instance. But need to pay attention to:
 >
 > -  Audio needs to be collected by the microphone or custom data at the same time in all instances, and the result of API calls related to the audio device will be based on the last time;
 > -  The result of camera-related API call will be based on the last time: startLocalPreview.
+
 **Return Desc:**
 
-` TRTCCloud ` subinstance
+ `TRTCCloud`  subinstance
 
 ## destroySubCloud
 
@@ -452,13 +435,13 @@ After disabling a certain upstream capability of the cross-room anchor, all user
 
 Note that this API can only be called by the anchor who is conducting the cross-room call, and the restrictions set by this API will be reset when the cross-room call is interrupted or the corresponding anchor leaves the room.
 
-The result of calling this API will be returned through the ` onUpdateOtherRoomForwardMode() ` callback in TRTCCloudListener.
+The result of calling this API will be returned through the  `onUpdateOtherRoomForwardMode()`  callback in TRTCCloudListener.
 
-For example, in room "101", there is anchor A and audience B, and in room "102", there is anchor C who publishes audio/video streams normally. Anchor A calls ` connectOtherRoom() ` to make a cross-room call with anchor C.
--  At this time, both anchor A and audience B will receive the ` onRemoteUserEnterRoom(C) `, ` onUserVideoAvailable(C,true) `, and ` onUserAudioAvailable(C,true) ` event callbacks, and can subscribe to the audio/video streams of anchor C.
+For example, in room "101", there is anchor A and audience B, and in room "102", there is anchor C who publishes audio/video streams normally. Anchor A calls  `connectOtherRoom()`  to make a cross-room call with anchor C.
+-  At this time, both anchor A and audience B will receive the  `onRemoteUserEnterRoom(C)` ,  `onUserVideoAvailable(C,true)` , and  `onUserAudioAvailable(C,true)`  event callbacks, and can subscribe to the audio/video streams of anchor C.
 
 Later, anchor A calls this API to disable the audio publishing capability of anchor C in the current room.
--  After that, the audio stream of anchor C will no longer be published to room "101", and both anchor A and audience B will receive the ` onUserAudioAvailable(C,false) ` event callback, and cannot subscribe to the audio stream of anchor C by calling ` muteRemoteAudio(C,false) `.
+-  After that, the audio stream of anchor C will no longer be published to room "101", and both anchor A and audience B will receive the  `onUserAudioAvailable(C,false)`  event callback, and cannot subscribe to the audio stream of anchor C by calling  `muteRemoteAudio(C,false)` .
 
 -  The video stream of anchor C will not be affected. The other audience members in Room 102 will not be affected and can subscribe to the audio stream of anchor C normally.
 
@@ -488,7 +471,7 @@ Case 2: string room ID
 
 |Param|DESC|
 |---------|---------|
-|param|You need to pass in a string parameter in JSON format: ` roomId ` represents the room ID in numeric format, ` strRoomId ` represents the room ID in string format, and ` userId ` represents the user ID of the target anchor. ` muteAudio `, ` muteVideo ` and ` muteSubStream ` are optional, representing whether to prohibit or allow the cross-room anchor to publish audio/video/sub-stream.|
+|param|You need to pass in a string parameter in JSON format:  `roomId`  represents the room ID in numeric format,  `strRoomId`  represents the room ID in string format, and  `userId`  represents the user ID of the target anchor.  `muteAudio` ,  `muteVideo`  and  `muteSubStream`  are optional, representing whether to prohibit or allow the cross-room anchor to publish audio/video/sub-stream.|
 
 ## startPublishMediaStream
 
@@ -506,16 +489,12 @@ You can use the TRTCPublishMode parameter in TRTCPublishTarget to specify the pu
 
 > **Note**
 >
-
 > 1. The SDK will send a task ID to you via the onStartPublishMediaStream callback.
 >
-
 > 2. You can start a publishing task only once and cannot initiate two tasks that use the same publishing mode and publishing cdn url. Note the task ID returned, which you need to pass to updatePublishMediaStream to modify the publishing parameters or stopPublishMediaStream to stop the task.
 >
-
-> 3. You can specify up to 10 CDN URLs in ` target `. You will be charged only once for transcoding even if you relay to multiple CDNs.
+> 3. You can specify up to 10 CDN URLs in  `target` . You will be charged only once for transcoding even if you relay to multiple CDNs.
 >
-
 > 4. To avoid causing errors, do not specify the same URLs for different publishing tasks executed at the same time. We recommend you add "sdkappid_roomid_userid_main" to URLs to distinguish them from one another and avoid application conflicts.
 >
 
@@ -534,13 +513,10 @@ You can use this API to change the parameters of a publishing task initiated by
 
 > **Note**
 >
-
 > 1. You can use this API to add or remove CDN URLs to publish to (you can publish to up to 10 CDNs at a time). To avoid causing errors, do not specify the same URLs for different tasks executed at the same time.
 >
-
 > 2. You can use this API to switch a relaying task to transcoding or vice versa. For example, in cross-room communication, you can first call startPublishMediaStream to relay to a CDN. When the anchor requests cross-room communication, call this API, passing in the task ID to switch the relaying task to a transcoding task. This can ensure that the live stream and CDN playback are not interrupted (you need to keep the encoding parameters consistent).
 >
-
 > 3. You can not switch output between "only audio", "only video" and "audio and video" for the same task.
 >
 
@@ -556,11 +532,9 @@ You can use this API to stop a task initiated by startPublishMediaStream.
 
 > **Note**
 >
-
 > 1. If the task ID is not saved to your backend, you can call startPublishMediaStream again when an anchor re-enters the room after abnormal exit. The publishing will fail, but the TRTC backend will return the task ID to you.
 >
-
-> 2. If ` taskId ` is left empty, the TRTC backend will end all tasks you started through startPublishMediaStream. You can leave it empty if you have started only one task or want to stop all publishing tasks started by you.
+> 2. If  `taskId`  is left empty, the TRTC backend will end all tasks you started through startPublishMediaStream. You can leave it empty if you have started only one task or want to stop all publishing tasks started by you.
 >
 
 ## startLocalPreview
@@ -571,7 +545,7 @@ If this API is called before enterRoom, the SDK will only enable the camera and
 
 If it is called after enterRoom, the SDK will enable the camera and automatically start pushing the video stream.
 
-When the first camera video frame starts to be rendered, you will receive the ` onCameraDidReady ` callback in TRTCCloudListener.
+When the first camera video frame starts to be rendered, you will receive the  `onCameraDidReady`  callback in TRTCCloudListener.
 
 |Param|DESC|
 |---------|---------|
@@ -580,11 +554,10 @@ When the first camera video frame starts to be rendered, you will receive the `
 
 > **Note**
 >
-
-> If you want to preview the camera image and adjust the beauty filter parameters through ` BeautyManager ` before going live, you can:
+> If you want to preview the camera image and adjust the beauty filter parameters through  `BeautyManager`  before going live, you can:
 >
-> -  Scheme 1. Call ` startLocalPreview ` before calling enterRoom.
-> -  Scheme 2. Call ` startLocalPreview ` and ` muteLocalVideo(true) ` after calling enterRoom.
+> -  Scheme 1. Call  `startLocalPreview`  before calling enterRoom.
+> -  Scheme 2. Call  `startLocalPreview`  and  `muteLocalVideo(true)`  after calling enterRoom.
 
 ## updateLocalView
 
@@ -600,15 +573,15 @@ When the first camera video frame starts to be rendered, you will receive the `
 
 This API can pause (or resume) publishing the local video image. After the pause, other users in the same room will not be able to see the local image.
 
-This API is equivalent to the two APIs of ` startLocalPreview/stopLocalPreview ` when TRTCVideoStreamTypeBig is specified, but has higher performance and response speed.
+This API is equivalent to the two APIs of  `startLocalPreview/stopLocalPreview`  when TRTCVideoStreamTypeBig is specified, but has higher performance and response speed.
 
-The ` startLocalPreview/stopLocalPreview ` APIs need to enable/disable the camera, which are hardware device-related operations, so they are very time-consuming.
+The  `startLocalPreview/stopLocalPreview`  APIs need to enable/disable the camera, which are hardware device-related operations, so they are very time-consuming.
 
-In contrast, ` muteLocalVideo ` only needs to pause or allow the data stream at the software level, so it is more efficient and more suitable for scenarios where frequent enabling/disabling are needed.
+In contrast,  `muteLocalVideo`  only needs to pause or allow the data stream at the software level, so it is more efficient and more suitable for scenarios where frequent enabling/disabling are needed.
 
-After local video publishing is paused, other members in the same room will receive the ` onUserVideoAvailable(userId, false) ` callback notification.
+After local video publishing is paused, other members in the same room will receive the  `onUserVideoAvailable(userId, false)`  callback notification.
 
-After local video publishing is resumed, other members in the same room will receive the ` onUserVideoAvailable(userId, true) ` callback notification.
+After local video publishing is resumed, other members in the same room will receive the  `onUserVideoAvailable(userId, true)`  callback notification.
 
 |Param|DESC|
 |---------|---------|
@@ -619,7 +592,7 @@ After local video publishing is resumed, other members in the same room will rec
 
 **Set placeholder image during local video pause.**
 
-When you call ` muteLocalVideo(true) ` to pause the local video image, you can set a placeholder image by calling this API. Then, other users in the room will see this image instead of a black screen.
+When you call  `muteLocalVideo(true)`  to pause the local video image, you can set a placeholder image by calling this API. Then, other users in the room will see this image instead of a black screen.
 
 |Param|DESC|
 |---------|---------|
@@ -630,31 +603,27 @@ When you call ` muteLocalVideo(true) ` to pause the local video image, you can s
 
 **Subscribe to remote user's video stream and bind video rendering control.**
 
-Calling this API allows the SDK to pull the video stream of the specified ` userId ` and render it to the rendering control specified by the ` view ` parameter. You can set the display mode of the video image through setRemoteRenderParams.
--  If you already know the ` userId ` of a user who has a video stream in the room, you can directly call ` startRemoteView ` to subscribe to the user's video image.
+Calling this API allows the SDK to pull the video stream of the specified  `userId`  and render it to the rendering control specified by the  `view`  parameter. You can set the display mode of the video image through setRemoteRenderParams.
+-  If you already know the  `userId`  of a user who has a video stream in the room, you can directly call  `startRemoteView`  to subscribe to the user's video image.
 
 -  If you don't know which users in the room are publishing video streams, you can wait for the notification from onUserVideoAvailable after enterRoom.
 
 Calling this API only starts pulling the video stream, and the image needs to be loaded and buffered at this time. After the buffering is completed, you will receive a notification from onFirstVideoFrame.
 | Param | DESC |
 | --- | --- |
-| streamType | Video stream type of the <code> userId </code> specified for watching:<br>-  HD big image: TRTC_VIDEO_STREAM_TYPE_BIG<br>-  Smooth small image: TRTC_VIDEO_STREAM_TYPE_SMALL (the remote user should enable dual-channel encoding through enableEncSmallVideoStream for this parameter to take effect)<br>-  Substream image (usually used for screen sharing): TRTC_VIDEO_STREAM_TYPE_SUB |
+| streamType | Video stream type of the  <code>userId</code>  specified for watching:<br>-  HD big image: TRTC_VIDEO_STREAM_TYPE_BIG<br>-  Smooth small image: TRTC_VIDEO_STREAM_TYPE_SMALL (the remote user should enable dual-channel encoding through enableEncSmallVideoStream for this parameter to take effect)<br>-  Substream image (usually used for screen sharing): TRTC_VIDEO_STREAM_TYPE_SUB |
 | userId | ID of the specified remote user |
 | view | Rendering control that carries the video image |
 
 > **Note**
 >
-
 > The following requires your attention:
 >
-
-> 1. The SDK supports watching the big image and substream image or small image and substream image of a ` userId ` at the same time, but does not support watching the big image and small image at the same time.
+> 1. The SDK supports watching the big image and substream image or small image and substream image of a  `userId`  at the same time, but does not support watching the big image and small image at the same time.
 >
-
-> 2. Only when the specified ` userId ` enables dual-channel encoding through enableEncSmallVideoStream can the user's small image be viewed.
+> 2. Only when the specified  `userId`  enables dual-channel encoding through enableEncSmallVideoStream can the user's small image be viewed.
 >
-
-> 3. If the small image of the specified ` userId ` does not exist, the SDK will switch to the big image of the user by default.
+> 3. If the small image of the specified  `userId`  does not exist, the SDK will switch to the big image of the user by default.
 >
 
 ## updateRemoteView
@@ -676,7 +645,7 @@ This API can be used to update the rendering control of the remote video image.
 Calling this API will cause the SDK to stop receiving the user's video stream and release the decoding and rendering resources for the stream.
 | Param | DESC |
 | --- | --- |
-| streamType | Video stream type of the <code> userId </code> specified for watching:<br>-  HD big image: TRTC_VIDEO_STREAM_TYPE_BIG<br>-  Smooth small image: TRTC_VIDEO_STREAM_TYPE_SMALL<br>-  Substream image (usually used for screen sharing): TRTC_VIDEO_STREAM_TYPE_SUB |
+| streamType | Video stream type of the  <code>userId</code>  specified for watching:<br>-  HD big image: TRTC_VIDEO_STREAM_TYPE_BIG<br>-  Smooth small image: TRTC_VIDEO_STREAM_TYPE_SMALL<br>-  Substream image (usually used for screen sharing): TRTC_VIDEO_STREAM_TYPE_SUB |
 | userId | ID of the specified remote user |
 
 ## stopAllRemoteView
@@ -687,7 +656,6 @@ Calling this API will cause the SDK to stop receiving all remote video streams a
 
 > **Note**
 >
-
 > If a substream image (screen sharing) is being displayed, it will also be stopped.
 >
 
@@ -704,7 +672,6 @@ This API only pauses/resumes receiving the specified user's video stream but doe
 
 > **Note**
 >
-
 > This API can be called before room entry (enterRoom), and the pause status will be reset after room exit (exitRoom). After calling this API to pause receiving the video stream from a specific user, simply calling the startRemoteView API will not be able to play the video from that user. You need to call muteRemoteVideoStream(false) or muteAllRemoteVideoStreams(false) to resume it.
 >
 
@@ -720,10 +687,8 @@ This API only pauses/resumes receiving all users' video streams but does not rel
 
 > **Note**
 >
-
 > This API can be called before room entry (enterRoom), and the pause status will be reset after room exit (exitRoom).
 >
-
 > After calling this interface to pause receiving video streams from all users, simply calling the startRemoteView interface will not be able to play the video from a specific user. You need to call muteRemoteVideoStream(false) or muteAllRemoteVideoStreams(false) to resume it.
 >
 
@@ -739,7 +704,6 @@ This setting can determine the quality of image viewed by remote users, which is
 
 > **Note**
 >
-
 > Begin from v11.5 version, the encoding output resolution will be aligned according to width 8 and height 2 bytes, and will be adjusted downward, eg: input resolution 540x960, actual encoding output resolution 536x960.
 >
 
@@ -790,9 +754,9 @@ In this way, other users in the room can choose to subscribe to the **HD big ima
 
 > **Note**
 >
-
 > Dual-channel encoding will consume more CPU resources and network bandwidth; therefore, this feature can be enabled on macOS, Windows, or high-spec tablets, but is not recommended for phones.
 >
+
 **Return Desc:**
 
 0: success; -1: the current big image has been set to a lower quality, and it is not necessary to enable dual-channel encoding
@@ -812,7 +776,6 @@ You can use this API to select whether the image subscribed to is the big image
 
 > **Note**
 >
-
 > To implement this feature, the target user must have enabled the dual-channel encoding mode through enableEncSmallVideoStream; otherwise, this API will not work.
 >
 
@@ -830,7 +793,6 @@ You can use this API to screencapture the local video image or the primary strea
 
 > **Note**
 >
-
 > On Windows, only video image from the TRTC_SNAPSHOT_SOURCE_TYPE_STREAM source can be screencaptured currently.
 >
 
@@ -871,11 +833,10 @@ The SDK does not enable the mic by default. When a user wants to publish the loc
 After local audio capturing and publishing is enabled, other users in the room will receive the onUserAudioAvailable(userId, true) notification.
 | Param | DESC |
 | --- | --- |
-| quality | Sound quality<br>- TRTC_AUDIO_QUALITY_SPEECH - Smooth: mono channel; audio bitrate: 18 Kbps. This is suitable for audio call scenarios, such as online meeting and audio call.<br>- TRTC_AUDIO_QUALITY_DEFAULT - Default: mono channel; audio bitrate: 50 Kbps. This is the default sound quality of the SDK and recommended if there are no special requirements.<br>- TRTC_AUDIO_QUALITY_MUSIC - HD: dual channel + full band; audio bitrate: 128 Kbps. This is suitable for scenarios where Hi-Fi music transfer is required, such as online karaoke and music live streaming. |
+| quality | Sound quality<br>-  TRTC_AUDIO_QUALITY_SPEECH - Smooth: mono channel; audio bitrate: 18 Kbps. This is suitable for audio call scenarios, such as online meeting and audio call.<br>-  TRTC_AUDIO_QUALITY_DEFAULT - Default: mono channel; audio bitrate: 50 Kbps. This is the default sound quality of the SDK and recommended if there are no special requirements.<br>-  TRTC_AUDIO_QUALITY_MUSIC - HD: dual channel + full band; audio bitrate: 128 Kbps. This is suitable for scenarios where Hi-Fi music transfer is required, such as online karaoke and music live streaming. |
 
 > **Note**
 >
-
 > This API will check the mic permission. If the current application does not have permission to use the mic, the SDK will automatically ask the user to grant the mic permission.
 >
 
@@ -893,11 +854,11 @@ After local audio publishing is paused, other users in the room will receive the
 
 After local audio publishing is resumed, other users in the room will receive the onUserAudioAvailable(userId, true) notification.
 
-Different from stopLocalAudio, ` muteLocalAudio(true) ` does not release the mic permission; instead, it continues to send mute packets with extremely low bitrate.
+Different from stopLocalAudio,  `muteLocalAudio(true)`  does not release the mic permission; instead, it continues to send mute packets with extremely low bitrate.
 
 This is very suitable for scenarios that require on-cloud recording, as video file formats such as MP4 have a high requirement for audio continuity, while an MP4 recording file cannot be played back smoothly if stopLocalAudio is used.
 
-Therefore, ` muteLocalAudio ` instead of stopLocalAudio is recommended in scenarios where the requirement for recording file quality is high.
+Therefore,  `muteLocalAudio`  instead of stopLocalAudio is recommended in scenarios where the requirement for recording file quality is high.
 
 |Param|DESC|
 |---------|---------|
@@ -916,8 +877,7 @@ When you mute the remote audio of a specified user, the SDK will stop playing ba
 
 > **Note**
 >
-
-> This API works when called either before or after room entry (enterRoom), and the mute status will be reset to ` false ` after room exit (exitRoom).
+> This API works when called either before or after room entry (enterRoom), and the mute status will be reset to  `false`  after room exit (exitRoom).
 >
 
 ## muteAllRemoteAudio
@@ -932,8 +892,7 @@ When you mute the audio of all remote users, the SDK will stop playing back all
 
 > **Note**
 >
-
-> This API works when called either before or after room entry (enterRoom), and the mute status will be reset to ` false ` after room exit (exitRoom).
+> This API works when called either before or after room entry (enterRoom), and the mute status will be reset to  `false`  after room exit (exitRoom).
 >
 
 ## setAudioRoute
@@ -956,7 +915,7 @@ If audio route is set to the speaker, the volume is relatively high, so there is
 
 **Set the audio playback volume of remote user.**
 
-You can mute the audio of a remote user through ` setRemoteAudioVolume(userId, 0) `.
+You can mute the audio of a remote user through  `setRemoteAudioVolume(userId, 0)` .
 
 |Param|DESC|
 |---------|---------|
@@ -965,7 +924,6 @@ You can mute the audio of a remote user through ` setRemoteAudioVolume(userId, 0
 
 > **Note**
 >
-
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -979,13 +937,12 @@ You can mute the audio of a remote user through ` setRemoteAudioVolume(userId, 0
 
 > **Note**
 >
-
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
 ## getAudioCaptureVolume
 
-**Get the capturing volume of local audio.****Return Desc:**
+**Get the capturing volume of local audio.Return Desc:**
 
 capturing volume
 
@@ -1001,7 +958,6 @@ This API controls the volume of the sound ultimately delivered by the SDK to the
 
 > **Note**
 >
-
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -1022,7 +978,6 @@ After this feature is enabled, the SDK will return the audio volume assessment i
 
 > **Note**
 >
-
 > To enable this feature, call this API before calling startLocalAudio.
 >
 
@@ -1042,9 +997,9 @@ The startup and completion status of the recording will be notified through loca
 
 > **Note**
 >
-
 > Since version 11.5, the results of audio recording have been changed to be notified through asynchronous callbacks instead of return values. Please refer to the relevant callbacks of TRTCCloudListener .
 >
+
 **Return Desc:**
 
 0: success; -1: audio recording has been started; -2: failed to create file or directory; -3: the audio format of the specified file extension is not supported.
@@ -1106,10 +1061,8 @@ Update self position and orientation in the world coordinate system. The SDK wil
 
 > **Note**
 >
-
 > 1. The length of array should be 3.
 >
-
 > 2. Please limit the calling frequency appropriately. It's recommended that the interval between two operations be at least 100ms.
 >
 
@@ -1126,10 +1079,8 @@ Update the specified remote user's position in the world coordinate system. The
 
 > **Note**
 >
-
 > 1. The length of array should be 3.
 >
-
 > 2. Please limit the calling frequency appropriately. It's recommended that the interval between two operations of the same remote user be at least 100ms.
 >
 
@@ -1176,7 +1127,7 @@ beauty filter management class TXBeautyManager.
 
 **Add watermark.**
 
-The watermark position is determined by the ` rect ` parameter, which is a quadruple in the format of (x, y, width, height).
+The watermark position is determined by the  `rect`  parameter, which is a quadruple in the format of (x, y, width, height).
 -  x: X coordinate of watermark, which is a floating-point number between 0 and 1.
 
 -  y: Y coordinate of watermark, which is a floating-point number between 0 and 1.
@@ -1187,27 +1138,26 @@ The watermark position is determined by the ` rect ` parameter, which is a quadr
 
 Sample parameter:
 
-If the encoding resolution of the current video is 540x960, and the ` rect ` parameter is set to (0.1, 0.1, 0.2, 0.0),
+If the encoding resolution of the current video is 540x960, and the  `rect`  parameter is set to (0.1, 0.1, 0.2, 0.0),
 
-then the coordinates of the top-left point of the watermark will be (540 * 0.1, 960 * 0.1), i.e., (54, 96), the watermark width will be ` 540 * 0.2 = 108 px `, and the watermark height will be calculated automatically by the SDK based on the watermark image's aspect ratio.
+then the coordinates of the top-left point of the watermark will be (540 * 0.1, 960 * 0.1), i.e., (54, 96), the watermark width will be  `540 * 0.2 = 108 px` , and the watermark height will be calculated automatically by the SDK based on the watermark image's aspect ratio.
 
 |Param|DESC|
 |---------|---------|
 |image|Watermark image, which must be a PNG image with transparent background|
-|rect|Unified coordinates of the watermark relative to the encoded resolution. Value range of ` x `, ` y `, ` width `, and ` height `: 0–1.|
+|rect|Unified coordinates of the watermark relative to the encoded resolution. Value range of  `x` ,  `y` ,  `width` , and  `height` : 0–1.|
 |streamType|Specify for which image to set the watermark. For more information, please see TRTCVideoStreamType.|
 
 > **Note**
 >
-
-> If you want to set watermarks for both the primary image (generally for the camera) and the substream image (generally for screen sharing), you need to call this API twice with ` streamType ` set to different values.
+> If you want to set watermarks for both the primary image (generally for the camera) and the substream image (generally for screen sharing), you need to call this API twice with  `streamType`  set to different values.
 >
 
 ## getAudioEffectManager
 
 **Get sound effect management class (TXAudioEffectManager).**
 
-` TXAudioEffectManager ` is a sound effect management API, through which you can implement the following features:
+ `TXAudioEffectManager`  is a sound effect management API, through which you can implement the following features:
 -  Background music: both online music and local music can be played back with various features such as speed adjustment, pitch adjustment, original voice, accompaniment, and loop.
 
 -  In-ear monitoring: the sound captured by the mic is played back in the headphones in real time, which is generally used for music live streaming.
@@ -1216,7 +1166,7 @@ then the coordinates of the top-left point of the watermark will be (540 * 0.1,
 
 -  Voice changing effect: young girl, middle-aged man, heavy metal, and other effects.
 
--  Short sound effect: short sound effect files such as applause and laughter are supported (for files less than 10 seconds in length, please set the ` isShortFile ` parameter to ` true `).
+-  Short sound effect: short sound effect files such as applause and laughter are supported (for files less than 10 seconds in length, please set the  `isShortFile`  parameter to  `true` ).
 **Return Desc:**
 
 sound effect management class TXAudioEffectManager.
@@ -1233,16 +1183,12 @@ In live music scenarios, an anchor can use this API to have the SDK capture the
 
 > **Note**
 >
-
 > 1. This interface only works on Android API 29 and above.
 >
-
 > 2. You need to use this interface to enable system sound capture first, and it will take effect only when you call startScreenCapture to enable screen sharing.
 >
-
-> 3. You need to add a foreground service to ensure that the system sound capture is not silenced, and set ` android:foregroundServiceType="mediaProjection" `.
+> 3. You need to add a foreground service to ensure that the system sound capture is not silenced, and set  `android:foregroundServiceType="mediaProjection"` .
 >
-
 > 4. The SDK only capture audio of applications that satisfies the capture strategy and audio usage. Currently, the audio usage captured by the SDK includes USAGE_MEDIA, USAGE_GAME.
 >
 
@@ -1277,21 +1223,17 @@ Video encoding parameters recommended for screen sharing on Android (TRTCVideoEn
 
 |Param|DESC|
 |---------|---------|
-|encParams|Encoding parameters. For more information, please see TRTCCloudDef#TRTCVideoEncParam. If ` encParams ` is set to ` null `, the SDK will automatically use the previously set encoding parameter.|
-|shareParams|For more information, please see TRTCCloudDef#TRTCScreenShareParams. You can use the ` floatingView ` parameter to pop up a floating window (you can also use Android's ` WindowManager ` parameter to configure automatic pop-up).|
+|encParams|Encoding parameters. For more information, please see TRTCCloudDef#TRTCVideoEncParam. If  `encParams`  is set to  `null` , the SDK will automatically use the previously set encoding parameter.|
+|shareParams|For more information, please see TRTCCloudDef#TRTCScreenShareParams. You can use the  `floatingView`  parameter to pop up a floating window (you can also use Android's  `WindowManager`  parameter to configure automatic pop-up).|
 
 > **Note**
 >
-
 > Starting from Android 14 version, if you do not need to use the screen sharing function, you need to remove the permissions of the screen sharing front-end service in AndroidManifest.xml in your project, as follows:
 >
-
 > <uses-permission android:name="android.permission.FOREGROUND_SERVICE_MEDIA_PROJECTION" tools:node="remove" />
 >
-
 > If you need to use the screen sharing feature, you need to fill in the Play Console Statement as required by Google. The reference document is as follows:
 >
-
 > https://support.google.com/googleplay/android-developer/answer/13392821
 >
 
@@ -1305,7 +1247,6 @@ Video encoding parameters recommended for screen sharing on Android (TRTCVideoEn
 
 > **Note**
 >
-
 > Begin from v11.5 version, paused screen capture will use the last frame to output at a frame rate of 1fps.
 >
 
@@ -1320,9 +1261,9 @@ Video encoding parameters recommended for screen sharing on Android (TRTCVideoEn
 This API can set the image quality of screen sharing (i.e., the substream) viewed by remote users, which is also the image quality of screen sharing in on-cloud recording files.
 
 Please note the differences between the following two APIs:
-- setVideoEncoderParam is used to set the video encoding parameters of the primary stream image (TRTC_VIDEO_STREAM_TYPE_BIG, generally for camera).
+-  setVideoEncoderParam is used to set the video encoding parameters of the primary stream image (TRTC_VIDEO_STREAM_TYPE_BIG, generally for camera).
 
-- setSubStreamEncoderParam is used to set the video encoding parameters of the substream image (TRTC_VIDEO_STREAM_TYPE_SUB, generally for screen sharing).
+-  setSubStreamEncoderParam is used to set the video encoding parameters of the substream image (TRTC_VIDEO_STREAM_TYPE_SUB, generally for screen sharing).
 
 |Param|DESC|
 |---------|---------|
@@ -1356,18 +1297,15 @@ For more information, please see [Custom Capturing and Rendering](https://www.te
 
 |Param|DESC|
 |---------|---------|
-|frame|Video data. If the memory-based delivery scheme is used, please set the ` data ` field; if the video memory-based delivery scheme is used, please set the ` TRTCTexture ` field. For ` bufferType `, TRTC_VIDEO_BUFFER_TYPE_TEXTURE is recommended; for ` pixelFormat `, TRTC_VIDEO_PIXEL_FORMAT_Texture_2D is recommended. Refer to TRTCVideoFrame for more supported formats.|
+|frame|Video data. If the memory-based delivery scheme is used, please set the  `data`  field; if the video memory-based delivery scheme is used, please set the  `TRTCTexture`  field. For  `bufferType` , TRTC_VIDEO_BUFFER_TYPE_TEXTURE is recommended; for  `pixelFormat` , TRTC_VIDEO_PIXEL_FORMAT_Texture_2D is recommended. Refer to TRTCVideoFrame for more supported formats.|
 |streamType|Specify video stream type (TRTC_VIDEO_STREAM_TYPE_BIG: HD big image; TRTC_VIDEO_STREAM_TYPE_SUB: substream image).|
 
 > **Note**
 >
-
-> 1. We recommend you call the generateCustomPTS API to get the ` timestamp ` value of a video frame immediately after capturing it, so as to achieve the best audio/video sync effect.
+> 1. We recommend you call the generateCustomPTS API to get the  `timestamp`  value of a video frame immediately after capturing it, so as to achieve the best audio/video sync effect.
 >
-
 > 2. The video frame rate eventually encoded by the SDK is not determined by the frequency at which you call this API, but by the FPS you set in setVideoEncoderParam.
 >
-
 > 3. Please try to keep the calling interval of this API even; otherwise, problems will be caused, such as unstable output frame rate of the encoder or out-of-sync audio/video.
 >
 
@@ -1385,7 +1323,6 @@ You need to use sendCustomAudioData to continuously insert the captured audio da
 
 > **Note**
 >
-
 > As acoustic echo cancellation (AEC) requires strict control over the audio capturing and playback time, after custom audio capturing is enabled, AEC may fail.
 >
 
@@ -1394,7 +1331,7 @@ You need to use sendCustomAudioData to continuously insert the captured audio da
 **Deliver captured audio data to SDK.**
 
 We recommend you enter the following information for the TRTCAudioFrame parameter (other fields can be left empty):
--  audioFormat: audio data format, which can only be ` TRTCAudioFrameFormatPCM `.
+-  audioFormat: audio data format, which can only be  `TRTCAudioFrameFormatPCM` .
 
 -  data: audio frame buffer. Audio frame data must be in PCM format, and it supports a frame length of 5–100 ms (20 ms is recommended). Length calculation method: **for example, if the sample rate is 48000, then the frame length for mono channel will be `48000 * 0.02s * 1 * 16 bit = 15360 bit = 1920 bytes`.**
 
@@ -1412,7 +1349,6 @@ For more information, please see [Custom Capturing and Rendering](https://www.te
 
 > **Note**
 >
-
 > Please call this API accurately at intervals of the frame length; otherwise, sound lag may occur due to uneven data delivery intervals.
 >
 
@@ -1429,8 +1365,7 @@ After this feature is enabled, you can mix a custom audio track into the SDK thr
 
 > **Note**
 >
-
-> If you specify both ` enablePublish ` and ` enablePlayout ` as ` false `, the custom audio track will be completely closed.
+> If you specify both  `enablePublish`  and  `enablePlayout`  as  `false` , the custom audio track will be completely closed.
 >
 
 ## mixExternalAudioFrame
@@ -1443,25 +1378,25 @@ You are expected to feed audio data into the SDK at an even pace, but we underst
 
 Given this, we have provided a buffer pool in the SDK, which can cache the audio data you pass in to reduce the fluctuations in intervals between API calls.
 
-The value returned by this API indicates the size (ms) of the buffer pool. For example, if ` 50 ` is returned, it indicates that the buffer pool has 50 ms of audio data. As long as you call this API again within 50 ms, the SDK can make sure that continuous audio data is mixed.
+The value returned by this API indicates the size (ms) of the buffer pool. For example, if  `50`  is returned, it indicates that the buffer pool has 50 ms of audio data. As long as you call this API again within 50 ms, the SDK can make sure that continuous audio data is mixed.
 
-If the value returned is ` 100 ` or greater, you can wait after an audio frame is played to call the API again. If the value returned is smaller than ` 100 `, then there isn’t enough data in the buffer pool, and you should feed more audio data into the SDK until the data in the buffer pool is above the safety level.
+If the value returned is  `100`  or greater, you can wait after an audio frame is played to call the API again. If the value returned is smaller than  `100` , then there isn’t enough data in the buffer pool, and you should feed more audio data into the SDK until the data in the buffer pool is above the safety level.
 
 Fill the fields in TRTCAudioFrame as follows (other fields are not required).
-- ` data `: audio frame buffer. Audio frames must be in PCM format. Each frame can be 5-100 ms (20 ms is recommended) in duration. Assume that the sample rate is 48000, and sound channels mono-channel. Then the **frame size would be 48000 x 0.02s x 1 x 16 bit = 15360 bit = 1920 bytes**.
+-   `data` : audio frame buffer. Audio frames must be in PCM format. Each frame can be 5-100 ms (20 ms is recommended) in duration. Assume that the sample rate is 48000, and sound channels mono-channel. Then the **frame size would be 48000 x 0.02s x 1 x 16 bit = 15360 bit = 1920 bytes**.
 
-- ` sampleRate `: sample rate. Valid values: 16000, 24000, 32000, 44100, 48000
+-   `sampleRate` : sample rate. Valid values: 16000, 24000, 32000, 44100, 48000
 
-- ` channel `: number of sound channels (if dual-channel is used, data is interleaved). Valid values: ` 1 ` (mono-channel); ` 2 ` (dual channel)
+-   `channel` : number of sound channels (if dual-channel is used, data is interleaved). Valid values:  `1`  (mono-channel);  `2`  (dual channel)
 
-- ` timestamp `: timestamp (ms). Set it to the timestamp when audio frames are captured, which you can obtain by calling generateCustomPTS after getting an audio frame.
+-   `timestamp` : timestamp (ms). Set it to the timestamp when audio frames are captured, which you can obtain by calling generateCustomPTS after getting an audio frame.
 
 |Param|DESC|
 |---------|---------|
 |frame|Audio data|
 **Return Desc:**
 
-If the value returned is ` 0 ` or greater, the value represents the current size of the buffer pool; if the value returned is smaller than ` 0 `, it means that an error occurred. ` -1 ` indicates that you didn’t call enableMixExternalAudioFrame to enable custom audio tracks.
+If the value returned is  `0`  or greater, the value represents the current size of the buffer pool; if the value returned is smaller than  `0` , it means that an error occurred.  `-1`  indicates that you didn’t call enableMixExternalAudioFrame to enable custom audio tracks.
 
 ## setMixExternalAudioVolume
 
@@ -1484,7 +1419,7 @@ When you call APIs such as sendCustomVideoData or sendCustomAudioData for custom
 
 2. Then, send the video or audio frame to the preprocessing module you use (such as a third-party beauty filter or sound effect component).
 
-3. When you actually call sendCustomVideoData or sendCustomAudioData for delivery, assign the PTS timestamp recorded when the frame was captured to the ` timestamp ` field in TRTCVideoFrame or TRTCAudioFrame.**Return Desc:**
+3. When you actually call sendCustomVideoData or sendCustomAudioData for delivery, assign the PTS timestamp recorded when the frame was captured to the  `timestamp`  field in TRTCVideoFrame or TRTCAudioFrame.**Return Desc:**
 
 Timestamp in ms
 
@@ -1492,12 +1427,12 @@ Timestamp in ms
 
 **Set video data callback for third-party beauty filters.**
 
-After this callback is set, the SDK will call back the captured video frames through the ` listener ` you set and use them for further processing by a third-party beauty filter component. Then, the SDK will encode and send the processed video frames.
+After this callback is set, the SDK will call back the captured video frames through the  `listener`  you set and use them for further processing by a third-party beauty filter component. Then, the SDK will encode and send the processed video frames.
 | Param | DESC |
 | --- | --- |
-| bufferType | Specify the format of the data called back. Currently, it supports:<br>- TRTC_VIDEO_BUFFER_TYPE_TEXTURE: suitable when <code> pixelFormat </code> is set to TRTC_VIDEO_PIXEL_FORMAT_Texture_2D.<br>- TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER: suitable when <code> pixelFormat </code> is set to TRTC_VIDEO_PIXEL_FORMAT_I420.<br>- TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY: suitable when <code> pixelFormat </code> is set to TRTC_VIDEO_PIXEL_FORMAT_I420. |
+| bufferType | Specify the format of the data called back. Currently, it supports:<br>-  TRTC_VIDEO_BUFFER_TYPE_TEXTURE: suitable when  <code>pixelFormat</code>  is set to TRTC_VIDEO_PIXEL_FORMAT_Texture_2D.<br>-  TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER: suitable when  <code>pixelFormat</code>  is set to TRTC_VIDEO_PIXEL_FORMAT_I420.<br>-  TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY: suitable when  <code>pixelFormat</code>  is set to TRTC_VIDEO_PIXEL_FORMAT_I420. |
 | listener | Custom preprocessing callback. For more information, please see TRTCVideoFrameListener |
-| pixelFormat | Specify the format of the pixel called back. Currently, it supports:<br>- TRTC_VIDEO_PIXEL_FORMAT_Texture_2D: video memory-based texture scheme.<br>- TRTC_VIDEO_PIXEL_FORMAT_I420: memory-based data scheme. |
+| pixelFormat | Specify the format of the pixel called back. Currently, it supports:<br>-  TRTC_VIDEO_PIXEL_FORMAT_Texture_2D: video memory-based texture scheme.<br>-  TRTC_VIDEO_PIXEL_FORMAT_I420: memory-based data scheme. |
 **Return Desc:**
 
 0: success; values smaller than 0: error(For more information, please see TXLiteAVError)
@@ -1507,16 +1442,16 @@ After this callback is set, the SDK will call back the captured video frames thr
 **Set the callback of custom rendering for local video.**
 
 After this callback is set, the SDK will skip its own rendering process and call back the captured data. Therefore, you need to complete image rendering on your own.
-- ` pixelFormat ` specifies the format of the data called back. Currently, Texture2D, I420, and RGBA formats are supported.
+-   `pixelFormat`  specifies the format of the data called back. Currently, Texture2D, I420, and RGBA formats are supported.
 
-- ` bufferType ` specifies the buffer type. ` BYTE_BUFFER ` is suitable for the JNI layer, while ` BYTE_ARRAY ` can be used in direct operations at the Java layer.
+-   `bufferType`  specifies the buffer type.  `BYTE_BUFFER`  is suitable for the JNI layer, while  `BYTE_ARRAY`  can be used in direct operations at the Java layer.
 
 For more information, please see [Custom Capturing and Rendering](https://www.tencentcloud.com/document/product/647/35158).
 | Param | DESC |
 | --- | --- |
-| bufferType | Specify the data structure of the video frame:<br>- TRTC_VIDEO_BUFFER_TYPE_TEXTURE: suitable when <code> pixelFormat </code> is set to TRTC_VIDEO_PIXEL_FORMAT_Texture_2D.<br>- TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER: suitable when <code> pixelFormat </code> is set to TRTC_VIDEO_PIXEL_FORMAT_I420 or TRTC_VIDEO_PIXEL_FORMAT_RGBA.<br>- TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY: suitable when <code> pixelFormat </code> is set to TRTC_VIDEO_PIXEL_FORMAT_I420 or TRTC_VIDEO_PIXEL_FORMAT_RGBA. |
+| bufferType | Specify the data structure of the video frame:<br>-  TRTC_VIDEO_BUFFER_TYPE_TEXTURE: suitable when  <code>pixelFormat</code>  is set to TRTC_VIDEO_PIXEL_FORMAT_Texture_2D.<br>-  TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER: suitable when  <code>pixelFormat</code>  is set to TRTC_VIDEO_PIXEL_FORMAT_I420 or TRTC_VIDEO_PIXEL_FORMAT_RGBA.<br>-  TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY: suitable when  <code>pixelFormat</code>  is set to TRTC_VIDEO_PIXEL_FORMAT_I420 or TRTC_VIDEO_PIXEL_FORMAT_RGBA. |
 | listener | Callback of custom video rendering. The callback is returned once for each video frame |
-| pixelFormat | Specify the format of the video frame, such as:<br>- TRTC_VIDEO_PIXEL_FORMAT_Texture_2D: OpenGL texture format, which is suitable for GPU processing and has a high processing efficiency.<br>- TRTC_VIDEO_PIXEL_FORMAT_I420: standard I420 format, which is suitable for CPU processing and has a poor processing efficiency.<br>- TRTC_VIDEO_PIXEL_FORMAT_RGBA: RGBA format, which is suitable for CPU processing and has a poor processing efficiency. |
+| pixelFormat | Specify the format of the video frame, such as:<br>-  TRTC_VIDEO_PIXEL_FORMAT_Texture_2D: OpenGL texture format, which is suitable for GPU processing and has a high processing efficiency.<br>-  TRTC_VIDEO_PIXEL_FORMAT_I420: standard I420 format, which is suitable for CPU processing and has a poor processing efficiency.<br>-  TRTC_VIDEO_PIXEL_FORMAT_RGBA: RGBA format, which is suitable for CPU processing and has a poor processing efficiency. |
 **Return Desc:**
 
 0: success; values smaller than 0: error(For more information, please see TXLiteAVError)
@@ -1526,9 +1461,9 @@ For more information, please see [Custom Capturing and Rendering](https://www.te
 **Set the callback of custom rendering for remote video.**
 
 After this callback is set, the SDK will skip its own rendering process and call back the captured data. Therefore, you need to complete image rendering on your own.
-- ` pixelFormat ` specifies the format of the called back data, such as NV12, I420, and 32BGRA.
+-   `pixelFormat`  specifies the format of the called back data, such as NV12, I420, and 32BGRA.
 
-- ` bufferType ` specifies the buffer type. ` PixelBuffer ` has the highest efficiency, while ` NSData ` makes the SDK perform a memory conversion internally, which will result in extra performance loss.
+-   `bufferType`  specifies the buffer type.  `PixelBuffer`  has the highest efficiency, while  `NSData`  makes the SDK perform a memory conversion internally, which will result in extra performance loss.
 
 For more information, please see [Custom Capturing and Rendering](https://www.tencentcloud.com/document/product/647/35158).
 
@@ -1541,9 +1476,9 @@ For more information, please see [Custom Capturing and Rendering](https://www.te
 
 > **Note**
 >
-
-> Before this API is called, ` startRemoteView(nil) ` needs to be called to get the video stream of the remote user (` view ` can be set to ` nil ` for this end); otherwise, there will be no data called back.
+> Before this API is called,  `startRemoteView(nil)`  needs to be called to get the video stream of the remote user ( `view`  can be set to  `nil`  for this end); otherwise, there will be no data called back.
 >
+
 **Return Desc:**
 
 0: success; values smaller than 0: error(For more information, please see TXLiteAVError)
@@ -1553,17 +1488,16 @@ For more information, please see [Custom Capturing and Rendering](https://www.te
 **Set custom audio data callback.**
 
 After this callback is set, the SDK will internally call back the audio data (in PCM format), including:
-- onCapturedAudioFrame: callback of the audio data captured by the local mic
+-  onCapturedAudioFrame: callback of the audio data captured by the local mic
 
-- onLocalProcessedAudioFrame: callback of the audio data captured by the local mic and preprocessed by the audio module
+-  onLocalProcessedAudioFrame: callback of the audio data captured by the local mic and preprocessed by the audio module
 
-- onRemoteUserAudioFrame: audio data from each remote user before audio mixing
+-  onRemoteUserAudioFrame: audio data from each remote user before audio mixing
 
-- onMixedPlayAudioFrame: callback of the audio data that will be played back by the system after audio streams are mixed
+-  onMixedPlayAudioFrame: callback of the audio data that will be played back by the system after audio streams are mixed
 
 > **Note**
 >
-
 > Setting the callback to null indicates to stop the custom audio callback, while setting it to a non-null value indicates to start the custom audio callback.
 >
 
@@ -1571,7 +1505,7 @@ After this callback is set, the SDK will internally call back the audio data (in
 
 **Set the callback format of audio frames captured by local mic.**
 
-This API is used to set the ` AudioFrame ` format called back by onCapturedAudioFrame:
+This API is used to set the  `AudioFrame`  format called back by onCapturedAudioFrame:
 -  sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
 
 -  channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
@@ -1580,11 +1514,11 @@ This API is used to set the ` AudioFrame ` format called back by onCapturedAudio
 
 If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: number of sample points = number of milliseconds * sample rate / 1000
 
-For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as ` 960 = 20 * 48000 / 1000 `.
+For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as  `960 = 20 * 48000 / 1000` .
 
-Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: ` number of bytes = number of sample points * number of channels * 2 (bit width) `
+Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows:  `number of bytes = number of sample points * number of channels * 2 (bit width)`
 
-For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is ` 3840 = 960 * 2 * 2 `
+For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is  `3840 = 960 * 2 * 2`
 
 |Param|DESC|
 |---------|---------|
@@ -1597,20 +1531,20 @@ For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame
 
 **Set the callback format of preprocessed local audio frames.**
 
-This API is used to set the ` AudioFrame ` format called back by onLocalProcessedAudioFrame:
+This API is used to set the  `AudioFrame`  format called back by onLocalProcessedAudioFrame:
 -  sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
 
 -  channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
 
 -  samplesPerCall: number of sample points, which defines the frame length of the callback data. The frame length must be an integer multiple of 10 ms.
 
-If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: ` number of sample points = number of milliseconds * sample rate / 1000 `.
+If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows:  `number of sample points = number of milliseconds * sample rate / 1000` .
 
-For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as ` 960 = 20 * 48000 / 1000 `.
+For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as  `960 = 20 * 48000 / 1000` .
 
-Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: ` number of bytes = number of sample points * number of channels * 2 (bit width) `.
+Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows:  `number of bytes = number of sample points * number of channels * 2 (bit width)` .
 
-For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is ` 3840 = 960 * 2 * 2 `.
+For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is  `3840 = 960 * 2 * 2` .
 
 |Param|DESC|
 |---------|---------|
@@ -1623,20 +1557,20 @@ For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame
 
 **Set the callback format of audio frames to be played back by system.**
 
-This API is used to set the ` AudioFrame ` format called back by onMixedPlayAudioFrame:
+This API is used to set the  `AudioFrame`  format called back by onMixedPlayAudioFrame:
 -  sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
 
 -  channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
 
 -  samplesPerCall: number of sample points, which defines the frame length of the callback data. The frame length must be an integer multiple of 10 ms.
 
-If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: ` number of sample points = number of milliseconds * sample rate / 1000 `.
+If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows:  `number of sample points = number of milliseconds * sample rate / 1000` .
 
-For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as ` 960 = 20 * 48000 / 1000 `.
+For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as  `960 = 20 * 48000 / 1000` .
 
-Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: ` number of bytes = number of sample points * number of channels * 2 (bit width) `.
+Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows:  `number of bytes = number of sample points * number of channels * 2 (bit width)` .
 
-For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is ` 3840 = 960 * 2 * 2 `.
+For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is  `3840 = 960 * 2 * 2` .
 
 |Param|DESC|
 |---------|---------|
@@ -1659,7 +1593,6 @@ After you enable custom audio playback, the SDK will stop using its audio API to
 
 > **Note**
 >
-
 > The parameter must be set before room entry to take effect.
 >
 
@@ -1670,15 +1603,15 @@ After you enable custom audio playback, the SDK will stop using its audio API to
 Before calling this API, you need to first enable custom audio playback using enableCustomAudioRendering.
 
 Fill the fields in TRTCAudioFrame as follows (other fields are not required):
-- ` sampleRate `: sample rate (required). Valid values: 16000, 24000, 32000, 44100, 48000
+-   `sampleRate` : sample rate (required). Valid values: 16000, 24000, 32000, 44100, 48000
 
-- ` channel `: number of sound channels (required). ` 1 `: mono-channel; ` 2 `: dual-channel; if dual-channel is used, data is interleaved.
+-   `channel` : number of sound channels (required).  `1` : mono-channel;  `2` : dual-channel; if dual-channel is used, data is interleaved.
 
-- ` data `: the buffer used to get audio data. You need to allocate memory for the buffer based on the duration of an audio frame.
+-   `data` : the buffer used to get audio data. You need to allocate memory for the buffer based on the duration of an audio frame.
 
 The PCM data obtained can have a frame duration of 10 ms or 20 ms. 20 ms is recommended.
 
-Assume that the sample rate is 48000, and sound channels mono-channel. The buffer size for a 20 ms audio frame would be ` 48000 x 0.02s x 1 x 16 bit = 15360 bit = 1920 bytes `.
+Assume that the sample rate is 48000, and sound channels mono-channel. The buffer size for a 20 ms audio frame would be  `48000 x 0.02s x 1 x 16 bit = 15360 bit = 1920 bytes` .
 
 |Param|DESC|
 |---------|---------|
@@ -1686,13 +1619,10 @@ Assume that the sample rate is 48000, and sound channels mono-channel. The buffe
 
 > **Note**
 >
-
-> 1. You must set ` sampleRate ` and ` channel ` in ` audioFrame `, and allocate memory for one frame of audio in advance.
+> 1. You must set  `sampleRate`  and  `channel`  in  `audioFrame` , and allocate memory for one frame of audio in advance.
 >
-
-> 2. The SDK will fill the data automatically based on ` sampleRate ` and ` channel `.
+> 2. The SDK will fill the data automatically based on  `sampleRate`  and  `channel` .
 >
-
 > 3. We recommend that you use the system’s audio playback thread to drive the calling of this API, so that it is called each time the playback of an audio frame is complete.
 >
 
@@ -1702,7 +1632,7 @@ Assume that the sample rate is 48000, and sound channels mono-channel. The buffe
 
 This API allows you to use TRTC's UDP channel to broadcast custom data to other users in the current room for signaling transfer.
 
-Other users in the room can receive the message through the ` onRecvCustomCmdMsg ` callback in TRTCCloudListener.
+Other users in the room can receive the message through the  `onRecvCustomCmdMsg`  callback in TRTCCloudListener.
 
 |Param|DESC|
 |---------|---------|
@@ -1713,24 +1643,19 @@ Other users in the room can receive the message through the ` onRecvCustomCmdMsg
 
 > **Note**
 >
-
 > 1. Up to 30 messages can be sent per second to all users in the room (this is not supported for web and mini program currently. this limit is shared with sendSEIMsg).
 >
-
 > 2. A packet can contain up to 1 KB of data; if the threshold is exceeded, the packet is very likely to be discarded by the intermediate router or server.(this limit is shared with sendSEIMsg).
 >
-
 > 3. A client can send up to 16 KB of data in total per second.
 >
-
-> 4. ` reliable ` and ` ordered ` must be set to the same value (` true ` or ` false `) and cannot be set to different values currently.
+> 4.  `reliable`  and  `ordered`  must be set to the same value ( `true`  or  `false` ) and cannot be set to different values currently.
 >
-
-> 5. We strongly recommend you set different ` cmdID ` values for messages of different types. This can reduce message delay when orderly sending is required.
+> 5. We strongly recommend you set different  `cmdID`  values for messages of different types. This can reduce message delay when orderly sending is required.
 >
-
 > 6. Currently only the anchor role is supported.
 >
+
 **Return Desc:**
 
 true: sent the message successfully; false: failed to send the message.
@@ -1749,7 +1674,7 @@ However, because the data block of the video frame header cannot be too large, w
 
 The most common use is to embed the custom timestamp into video frames through this API so as to implement a perfect alignment between the message and video image (such as between the teaching material and video signal in the education scenario).
 
-Other users in the room can receive the message through the ` onRecvSEIMsg ` callback in TRTCCloudListener.
+Other users in the room can receive the message through the  `onRecvSEIMsg`  callback in TRTCCloudListener.
 
 |Param|DESC|
 |---------|---------|
@@ -1758,27 +1683,21 @@ Other users in the room can receive the message through the ` onRecvSEIMsg ` cal
 
 > **Note**
 >
-
 > This API has the following restrictions:
 >
-
 > 1. The data will not be instantly sent after this API is called; instead, it will be inserted into the next video frame after the API call.
 >
-
 > 2. Up to 30 messages can be sent per second to all users in the room (this limit is shared with sendCustomCmdMsg).
 >
-
 > 3. Each packet can be up to 1 KB (this limit is shared with sendCustomCmdMsg). If a large amount of data is sent, the video bitrate will increase, which may reduce the video quality or even cause lagging.
 >
-
 > 4. Each client can send up to 16 KB of data in total per second (this limit is shared with sendCustomCmdMsg).
 >
-
-> 5. If multiple times of sending is required (i.e., ` repeatCount ` > 1), the data will be inserted into subsequent ` repeatCount ` video frames in a row for sending, which will increase the video bitrate.
+> 5. If multiple times of sending is required (i.e.,  `repeatCount`  > 1), the data will be inserted into subsequent  `repeatCount`  video frames in a row for sending, which will increase the video bitrate.
 >
-
-> 6. If ` repeatCount ` is greater than 1, the data will be sent for multiple times, and the same message may be received multiple times in the onRecvSEIMsg callback; therefore, deduplication is required.
+> 6. If  `repeatCount`  is greater than 1, the data will be sent for multiple times, and the same message may be received multiple times in the onRecvSEIMsg callback; therefore, deduplication is required.
 >
+
 **Return Desc:**
 
 true: the message is allowed and will be sent with subsequent video frames; false: the message is not allowed to be sent
@@ -1793,15 +1712,13 @@ true: the message is allowed and will be sent with subsequent video frames; fals
 
 > **Note**
 >
-
 > 1. The speed measurement process will incur a small amount of basic service fees, See [Purchase Guide > Base Services](https://intl.cloud.tencent.com/document/product/647/34610).
 >
-
 > 2. Please perform the Network speed test before room entry, because if performed after room entry, the test will affect the normal audio/video transfer, and its result will be inaccurate due to interference in the room.
 >
-
 > 3. Only one network speed test task is allowed to run at the same time.
 >
+
 **Return Desc:**
 
 interface call result, <0: failure
@@ -1847,11 +1764,11 @@ If compression is disabled, logs will be stored in plaintext and can be read dir
 **Set local log storage path.**
 
 You can use this API to change the default storage path of the SDK's local logs, which is as follows:
--  Windows: C:/Users/[username]/AppData/Roaming/liteav/log, i.e., under ` %appdata%/liteav/log `.
+-  Windows: C:/Users/[username]/AppData/Roaming/liteav/log, i.e., under  `%appdata%/liteav/log` .
 
--  iOS or macOS: under ` sandbox Documents/log `.
+-  iOS or macOS: under  `sandbox Documents/log` .
 
--  Android: under ` /app directory/files/log/liteav/ `.
+-  Android: under  `/app directory/files/log/liteav/` .
 
 |Param|DESC|
 |---------|---------|
@@ -1859,7 +1776,6 @@ You can use this API to change the default storage path of the SDK's local logs,
 
 > **Note**
 >
-
 > Please be sure to call this API before all other APIs and make sure that the directory you specify exists and your application has read/write permissions of the directory.
 >
 
@@ -1885,7 +1801,7 @@ This API is used to adjust the position of the dashboard in the video rendering
 
 |Param|DESC|
 |---------|---------|
-|margin|Inner margin of the dashboard. It should be noted that this is based on the percentage of ` parentView `. Value range: [0, 1]|
+|margin|Inner margin of the dashboard. It should be noted that this is based on the percentage of  `parentView` . Value range: [0, 1]|
 |userId|User ID|
 
 ## callExperimentalAPI
@@ -1907,9 +1823,9 @@ After the user exits the room, the SDK will automatically close the private encr
 
 > **Note**
 >
-
 > TRTC has built-in encryption for media streams before transmission. After private encryption of media streams is enabled, it will be re-encrypted with the key and initial vector you pass in.
 >
+
 **Return Desc:**
 
 Interface call result, 0: Method call succeeded, -1: The incoming parameter is invalid, -2: Your subscription has expired. If you want to renew it, Please go to purchase the [TRTC Professional Edition](https://console.trtc.io/subscription/buy/rtc), and [contact us](https://trtc.io/zh/contact) for review before use.