@tencent-rtc/mcp 1.6.7 → 1.7.0

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

@@ -123,7 +123,6 @@ Version: 13.4
 |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.
 >
 
@@ -154,7 +153,6 @@ In other words, if you do not set the  `listenerHandler`  attribute, all callbac
 |listenerHandler||
 
 > **Note**
->
 > If you specify a  `listenerHandler` , please do not manipulate the UI in the TRTCCloudListener callback function; otherwise, thread safety issues will occur.
 >
 
@@ -197,7 +195,6 @@ After calling this API, you will receive the  `onEnterRoom(result)`  callback fr
 |scene|Application scenario, which is used to specify the use case. The same TRTCAppScene should be configured for all users in the same room.|
 
 > **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.
 >
 > 2. The same  `scene`  should be configured for all users in the same room.
@@ -229,7 +226,6 @@ You can use the  `role`  field in TRTCParams during room entry to specify the us
 | 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.
@@ -250,7 +246,6 @@ You can use the  `role`  field in TRTCParams during room entry to specify the us
 | 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.
@@ -274,7 +269,6 @@ The API call result will be called back through  `onSwitchRoom(errCode, errMsg)`
 |config|Room parameter. For more information, please see TRTCSwitchRoomConfig.|
 
 > **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:
 >
 > 1. If you decide to use  `strRoomId` , then set  `roomId`  to 0. If both are specified,  `roomId`  will be used.
@@ -354,7 +348,6 @@ In your application scenario, if there are many audio/video streams being publis
 |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.
@@ -399,7 +392,6 @@ Below is the sample code:
 ```
 
 > **Note**
->
 > 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` .
@@ -483,12 +475,11 @@ You can use the TRTCPublishMode parameter in TRTCPublishTarget to specify the pu
 
 |Param|DESC|
 |---------|---------|
-|config|The On-Cloud MixTranscoding settings. This parameter is invalid in the relay-to-CDN mode. It is required if you transcode and publish the stream to a CDN or to a TRTC room. For details, see TRTCStreamMixingConfig.|
-|params|The encoding settings. This parameter is required if you transcode and publish the stream to a CDN or to a TRTC room. If you relay to a CDN without transcoding, to improve the relaying stability and playback compatibility, we also recommend you set this parameter. For details, see TRTCStreamEncoderParam.|
-|target|The publishing destination. You can relay the stream to a CDN (after transcoding or without transcoding) or transcode and publish the stream to a TRTC room. For details, see TRTCPublishTarget.|
+|config|The On-Cloud MixTranscoding settings. This parameter is invalid in the relay-to-CDN mode. It is required if you transcode and publish the stream to a CDN or to a TRTC room. .|
+|params|The encoding settings. This parameter is required if you transcode and publish the stream to a CDN or to a TRTC room. If you relay to a CDN without transcoding, to improve the relaying stability and playback compatibility, we also recommend you set this parameter. .|
+|target|The publishing destination. You can relay the stream to a CDN (after transcoding or without transcoding) or transcode and publish the stream to a TRTC room. .|
 
 > **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.
@@ -506,13 +497,12 @@ You can use this API to change the parameters of a publishing task initiated by
 
 |Param|DESC|
 |---------|---------|
-|config|The On-Cloud MixTranscoding settings. This parameter is invalid in the relay-to-CDN mode. It is required if you transcode and publish the stream to a CDN or to a TRTC room. For details, see TRTCStreamMixingConfig.|
-|params|The encoding settings. This parameter is required if you transcode and publish the stream to a CDN or to a TRTC room. If you relay to a CDN without transcoding, to improve the relaying stability and playback compatibility, we recommend you set this parameter. For details, see TRTCStreamEncoderParam.|
-|target|The publishing destination. You can relay the stream to a CDN (after transcoding or without transcoding) or transcode and publish the stream to a TRTC room. For details, see TRTCPublishTarget.|
+|config|The On-Cloud MixTranscoding settings. This parameter is invalid in the relay-to-CDN mode. It is required if you transcode and publish the stream to a CDN or to a TRTC room. .|
+|params|The encoding settings. This parameter is required if you transcode and publish the stream to a CDN or to a TRTC room. If you relay to a CDN without transcoding, to improve the relaying stability and playback compatibility, we recommend you set this parameter. .|
+|target|The publishing destination. You can relay the stream to a CDN (after transcoding or without transcoding) or transcode and publish the stream to a TRTC room. .|
 |taskId|The task ID returned to you via the onStartPublishMediaStream callback.|
 
 > **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).
@@ -531,7 +521,6 @@ You can use this API to stop a task initiated by startPublishMediaStream.
 |taskId|The task ID returned to you via the onStartPublishMediaStream callback.|
 
 > **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.
@@ -553,7 +542,6 @@ When the first camera video frame starts to be rendered, you will receive the  `
 |view|Control that carries the video image|
 
 > **Note**
->
 > 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.
@@ -616,7 +604,6 @@ Calling this API only starts pulling the video stream, and the image needs to be
 | 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.
@@ -655,7 +642,6 @@ Calling this API will cause the SDK to stop receiving the user's video stream an
 Calling this API will cause the SDK to stop receiving all remote video streams and release all decoding and rendering resources.
 
 > **Note**
->
 > If a substream image (screen sharing) is being displayed, it will also be stopped.
 >
 
@@ -671,7 +657,6 @@ This API only pauses/resumes receiving the specified user's video stream but doe
 | userId | ID of the specified remote user |
 
 > **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.
 >
 
@@ -686,7 +671,6 @@ This API only pauses/resumes receiving all users' video streams but does not rel
 |mute|Whether to pause receiving|
 
 > **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.
@@ -703,7 +687,6 @@ This setting can determine the quality of image viewed by remote users, which is
 |param|It is used to set relevant parameters for the video encoder. For more information, please see TRTCVideoEncParam.|
 
 > **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.
 >
 
@@ -715,7 +698,7 @@ This setting determines the quality control policy in a poor network environment
 
 |Param|DESC|
 |---------|---------|
-|param|It is used to set relevant parameters for network quality control. For details, please refer to TRTCNetworkQosParam.|
+|param|It is used to set relevant parameters for network quality control. For details, please .|
 
 ## setLocalRenderParams
 
@@ -753,7 +736,6 @@ In this way, other users in the room can choose to subscribe to the **HD big ima
 |smallVideoEncParam|Video parameters of small image stream|
 
 > **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.
 >
 
@@ -775,7 +757,6 @@ You can use this API to select whether the image subscribed to is the big image
 |userId|ID of the specified remote user|
 
 > **Note**
->
 > To implement this feature, the target user must have enabled the dual-channel encoding mode through enableEncSmallVideoStream; otherwise, this API will not work.
 >
 
@@ -792,7 +773,6 @@ You can use this API to screencapture the local video image or the primary strea
 |userId|User ID. A null value indicates to screencapture the local video.|
 
 > **Note**
->
 > On Windows, only video image from the TRTC_SNAPSHOT_SOURCE_TYPE_STREAM source can be screencaptured currently.
 >
 
@@ -836,7 +816,6 @@ After local audio capturing and publishing is enabled, other users in the room w
 | 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.
 >
 
@@ -876,7 +855,6 @@ When you mute the remote audio of a specified user, the SDK will stop playing ba
 |userId|ID of the specified remote user|
 
 > **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).
 >
 
@@ -891,7 +869,6 @@ When you mute the audio of all remote users, the SDK will stop playing back all
 |mute|true: mute; false: unmute|
 
 > **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).
 >
 
@@ -923,7 +900,6 @@ You can mute the audio of a remote user through  `setRemoteAudioVolume(userId, 0
 |volume|Volume. 100 is the original volume. Value range: [0,150]. Default value: 100|
 
 > **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.
 >
 
@@ -936,7 +912,6 @@ You can mute the audio of a remote user through  `setRemoteAudioVolume(userId, 0
 |volume|Volume. 100 is the original volume. Value range: [0,150]. Default value: 100|
 
 > **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.
 >
 
@@ -957,7 +932,6 @@ This API controls the volume of the sound ultimately delivered by the SDK to the
 |volume|Volume. 100 is the original volume. Value range: [0,150]. Default value: 100|
 
 > **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.
 >
 
@@ -977,7 +951,6 @@ After this feature is enabled, the SDK will return the audio volume assessment i
 |params|Volume evaluation and other related parameters, please see TRTCAudioVolumeEvaluateParams|
 
 > **Note**
->
 > To enable this feature, call this API before calling startLocalAudio.
 >
 
@@ -996,7 +969,6 @@ The startup and completion status of the recording will be notified through loca
 |param|Recording parameter. For more information, please see TRTCAudioRecordingParams|
 
 > **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 .
 >
 
@@ -1060,7 +1032,6 @@ Update self position and orientation in the world coordinate system. The SDK wil
 |position|The coordinate of self in the world coordinate system. The three values represent the forward, right and up coordinate values in turn.|
 
 > **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.
@@ -1078,7 +1049,6 @@ Update the specified remote user's position in the world coordinate system. The
 |userId|ID of the specified remote user.|
 
 > **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.
@@ -1149,7 +1119,6 @@ then the coordinates of the top-left point of the watermark will be (540 * 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.
 >
 
@@ -1182,7 +1151,6 @@ In online education scenarios, a teacher can use this API to have the SDK captur
 In live music scenarios, an anchor can use this API to have the SDK capture the music played back by his or her player so as to add background music to the room.
 
 > **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.
@@ -1227,7 +1195,6 @@ Video encoding parameters recommended for screen sharing on Android (TRTCVideoEn
 |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" />
@@ -1246,7 +1213,6 @@ Video encoding parameters recommended for screen sharing on Android (TRTCVideoEn
 **Pause screen sharing.**
 
 > **Note**
->
 > Begin from v11.5 version, paused screen capture will use the last frame to output at a frame rate of 1fps.
 >
 
@@ -1297,11 +1263,10 @@ 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. .|
 |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.
 >
 > 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.
@@ -1322,7 +1287,6 @@ You need to use sendCustomAudioData to continuously insert the captured audio da
 |enable|Whether to enable. Default value: false|
 
 > **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.
 >
 
@@ -1348,7 +1312,6 @@ For more information, please see [Custom Capturing and Rendering](https://www.te
 |frame|Audio data|
 
 > **Note**
->
 > Please call this API accurately at intervals of the frame length; otherwise, sound lag may occur due to uneven data delivery intervals.
 >
 
@@ -1364,7 +1327,6 @@ After this feature is enabled, you can mix a custom audio track into the SDK thr
 |enablePublish|Whether the mixed audio track should be played back remotely. Default value: false|
 
 > **Note**
->
 > If you specify both  `enablePublish`  and  `enablePlayout`  as  `false` , the custom audio track will be completely closed.
 >
 
@@ -1475,7 +1437,6 @@ For more information, please see [Custom Capturing and Rendering](https://www.te
 |userId|ID of the specified remote user|
 
 > **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.
 >
 
@@ -1497,7 +1458,6 @@ After this callback is set, the SDK will internally call back the audio data (in
 -  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.
 >
 
@@ -1592,7 +1552,6 @@ After you enable custom audio playback, the SDK will stop using its audio API to
 |enable|Whether to enable custom audio playback. It's disabled by default.|
 
 > **Note**
->
 > The parameter must be set before room entry to take effect.
 >
 
@@ -1618,7 +1577,6 @@ Assume that the sample rate is 48000, and sound channels mono-channel. The buffe
 |audioFrame|Audio frames|
 
 > **Note**
->
 > 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` .
@@ -1642,7 +1600,6 @@ Other users in the room can receive the message through the  `onRecvCustomCmdMsg
 |reliable|Whether reliable sending is enabled. Reliable sending can achieve a higher success rate but with a longer reception delay than unreliable sending.|
 
 > **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).
@@ -1682,7 +1639,6 @@ Other users in the room can receive the message through the  `onRecvSEIMsg`  cal
 |repeatCount|Data sending count|
 
 > **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.
@@ -1711,7 +1667,6 @@ true: the message is allowed and will be sent with subsequent video frames; fals
 |params|speed test options|
 
 > **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.
@@ -1775,7 +1730,6 @@ You can use this API to change the default storage path of the SDK's local logs,
 |path|Log storage path|
 
 > **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.
 >
 
@@ -1822,10 +1776,8 @@ After the user exits the room, the SDK will automatically close the private encr
 |enabled|Whether to enable media stream private encryption.|
 
 > **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.

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

@@ -126,7 +126,6 @@ TRTC provides three different video streams, including:
 -  Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their own substream.
 
 > **Note**
->
 > The SDK does not support enabling the smooth small image alone, which must be enabled together with the big image. It will automatically set the resolution and bitrate of the small image.
 >
 
@@ -698,9 +697,9 @@ You can test the network speed through the startSpeedTest: interface before the
 | expectedDownBandwidth | Expected downstream bandwidth (kbps, value range: 10 to 5000, no downlink bandwidth test when it is 0).<br><strong>Note</strong><br>When the parameter  <code>scene</code>  is set to TRTC_SpeedTestScene_Online_Chorus_Testing, in order to obtain more accurate information such as  <code>rtt / jitter</code> , the value range is limited to [10, 1000]. |
 | expectedUpBandwidth | Expected upstream bandwidth (kbps, value range: 10 to 5000, no uplink bandwidth test when it is 0).<br><strong>Note</strong><br>When the parameter  <code>scene</code>  is set to TRTC_SpeedTestScene_Online_Chorus_Testing, in order to obtain more accurate information such as  <code>rtt / jitter</code> , the value range is limited to [10, 1000]. |
 | scene | Speed test scene. |
-| sdkAppId | Application identification, please refer to the relevant instructions in TRTCParams. |
-| userId | User identification, please refer to the relevant instructions in TRTCParams. |
-| userSig | User signature, please refer to the relevant instructions in TRTCParams. |
+| sdkAppId | Application identification, please . |
+| userId | User identification, please . |
+| userSig | User signature, please . |
 
 ## TRTCSpeedTestResult
 
@@ -800,7 +799,7 @@ These parameters are used to specify the layout position information of each vid
 | mixUsers | Field description: specify the position, size, layer, and stream type of each video image in On-Cloud MixTranscoding<br>Recommended value: this field is an array in  <code>TRTCMixUser</code>  type, where each element represents the information of a video image. |
 | mode | Field description: layout mode<br>Recommended value: please choose a value according to your business needs. The preset mode has better applicability. |
 | streamId | Field description: ID of the live stream output to CDN<br>Recommended value: default value: null, that is, the audio/video streams in the room will be mixed into the audio/video stream of the caller of this API.<br>-  If you don't set this parameter, the SDK will execute the default logic, that is, it will mix the multiple audio/video streams in the room into the audio/video stream of the caller of this API, i.e., A + B => A.<br>-  If you set this parameter, the SDK will mix the audio/video streams in the room into the live stream you specify, i.e., A + B => C (C is the  <code>streamId</code>  you specify). |
-| videoBitrate | Field description: specify the target video bitrate (Kbps) of On-Cloud MixTranscoding<br>Recommended value: if you enter 0, TRTC will estimate a reasonable bitrate value based on  <code>videoWidth</code>  and  <code>videoHeight</code> . You can also refer to the recommended bitrate value in the video resolution enumeration definition (in the comment section). |
+| videoBitrate | Field description: specify the target video bitrate (Kbps) of On-Cloud MixTranscoding<br>Recommended value: if you enter 0, TRTC will estimate a reasonable bitrate value based on <code>videoWidth</code> and <code>videoHeight</code> . You can also . |
 | videoFramerate | Field description: specify the target video frame rate (fps) of On-Cloud MixTranscoding<br>Recommended value: default value: 15 fps. Value range: (0,30]. |
 | videoGOP | Field description: specify the target video keyframe interval (GOP) of On-Cloud MixTranscoding<br>Recommended value: default value: 2 (in seconds). Value range: [1,8]. |
 | videoHeight | Field description: specify the target resolution (height) of On-Cloud MixTranscoding<br>Recommended value: 640 px. If you only mix audio streams, please set both  <code>width</code>  and  <code>height</code>  to 0; otherwise, there will be a black background in the live stream after mixtranscoding. |
@@ -850,7 +849,7 @@ The startLocalRecording API is an enhanced version of the startAudioRecording AP
 
 **Sound effect parameter (disused).**
 
-"Sound effects" in TRTC refer to some short audio files (usually only a few seconds), such as "applause" and "laughter".
+"Sound effects" in TRTC .
 
 This parameter is used to specify the path and number of playback times of a sound effect file (short audio file) in the sound effect playback API TRTCCloud#playAudioEffect on legacy versions.
 
@@ -985,7 +984,7 @@ This enum type is used by the On-Cloud MixTranscoding parameter TRTCStreamMixing
 | videoEncodedFPS | <code>Description:</code>  The frame rate (fps) of the stream to publish.<br><code>Value:</code>  Value range: (0,30]. Default: 20. |
 | videoEncodedGOP | <code>Description:</code>  The keyframe interval (GOP) of the stream to publish.<br><code>Value:</code>  Value range: [1,5]. Default: 3 (seconds). |
 | videoEncodedHeight | <code>Description:</code>  The resolution (height) of the stream to publish.<br><code>Value:</code>  Recommended value: 640. If you mix only audio streams, to avoid displaying a black video in the transcoded stream, set both  <code>width</code>  and  <code>height</code>  to  <code>0</code> . |
-| videoEncodedKbps | <code>Description:</code>  The video bitrate (Kbps) of the stream to publish.<br><code>Value:</code>  If you set this parameter to  <code>0</code> , TRTC will work out a bitrate based on  <code>videoWidth</code>  and  <code>videoHeight</code> . For details, refer to the recommended bitrates for the constants of the resolution enum type (see comment). |
+| videoEncodedKbps | <code>Description:</code> The video bitrate (Kbps) of the stream to publish.<br><code>Value:</code> If you set this parameter to <code>0</code> , TRTC will work out a bitrate based on <code>videoWidth</code> and <code>videoHeight</code> . . |
 | videoEncodedWidth | <code>Description:</code>  The resolution (width) of the stream to publish.<br><code>Value:</code>  Recommended value: 368. If you mix only audio streams, to avoid displaying a black video in the transcoded stream, set both  <code>width</code>  and  <code>height</code>  to  <code>0</code> . |
 | videoSeiParams | <code>Description:</code>  SEI parameters. Default: null<br><code>Note:</code>  the parameter is passed in the form of a JSON string. Here is an example to use it:<br>`{`<br>`  "payLoadContent":"xxx",`<br>`  "payloadType":5,`<br>`  "payloadUuid":"1234567890abcdef1234567890abcdef",`<br>`  "interval":1000,`<br>`  "followIdr":false`<br>`}`<br>The currently supported fields and their meanings are as follows:<br>-  payloadContent: Required. The payload content of the passthrough SEI, which cannot be empty.<br>-  payloadType: Required. The type of the SEI message, with a value range of 5 or an integer within the range of [100, 254] (excluding 244, which is an internally defined timestamp SEI).<br>-  payloadUuid: Required when payloadType is 5, and ignored in other cases. The value must be a 32-digit hexadecimal number.<br>-  interval: Optional, default is 1000. The sending interval of the SEI, in milliseconds.<br>-  followIdr: Optional, default is false. When this value is true, the SEI will be ensured to be carried when sending a key frame, otherwise it is not guaranteed. |
 

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

@@ -159,7 +159,6 @@ int onProcessVideoFrame(TRTCCloudDef.TRTCVideoFrame srcFrame, TRTCCloudDef.TRTCV
 |srcFrame|Used to carry images captured by TRTC via the camera|
 
 > **Note**
->
 > Currently, only the OpenGL texture scheme is supported(PC supports TRTCVideoBufferType_Buffer format Only)
 >
 
@@ -183,7 +182,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
 >
 > 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
@@ -215,7 +213,6 @@ Other users in the room can receive the message through the  `TRTCAudioFrame.ext
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
 >
 > 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
@@ -240,7 +237,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |userId|User ID|
 
 > **Note**
->
 > The audio data returned via this callback can be read but not modified.
 >
 
@@ -260,7 +256,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
 >
 > 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
@@ -284,7 +279,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. This data returned via this callback is mixed from all audio in the SDK, including local audio after pre-processing (ANS, AEC, and AGC), special effects application, and music mixing, as well as all remote audio, but it does not include the in-ear monitoring data.
 >
 > 2. The audio data returned via this callback cannot be modified.
@@ -306,7 +300,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. Please avoid time-consuming operations in this callback function, or it will cause audio exceptions.
 >
 > 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
@@ -368,7 +361,6 @@ For more information on the error codes for room entry failure, see [Error Codes
 |result|If  `result`  is greater than 0, it indicates the time (in ms) the room entry takes; if  `result`  is less than 0, it represents the error code for room entry.|
 
 > **Note**
->
 > 1. In TRTC versions below 6.6, the  `onEnterRoom(result)`  callback is returned only if room entry succeeds, and the onError callback is returned if room entry fails.
 >
 > 2. In TRTC 6.6 and above, the  `onEnterRoom(result)`  callback is returned regardless of whether room entry succeeds or fails, and the onError callback is also returned if room entry fails.
@@ -470,7 +462,6 @@ Due to performance concerns, this callback works differently in different scenar
 |userId|User ID of the remote user|
 
 > **Note**
->
 > 1. The  `onRemoteUserEnterRoom`  callback indicates that a user entered the room, but it does not necessarily mean that the user enabled audio or video.
 >
 > 2. If you want to know whether a user enabled video, we recommend you use the onUserVideoAvailable callback.
@@ -519,7 +510,6 @@ You can then call startRemoteView to subscribe to the remote user’s video. If
 |userId|User ID of the remote user|
 
 > **Note**
->
 > The API used to display substream images is startRemoteView, not startRemoteSubStreamView, startRemoteSubStreamView is deprecated.
 >
 
@@ -538,7 +528,6 @@ If you receive the  `onUserAudioAvailable(userId, true)`  callback, it indicates
 |userId|User ID of the remote user|
 
 > **Note**
->
 > The auto-subscription mode is used by default. You can switch to the manual subscription mode by calling setDefaultStreamRecvMode, but it must be called before room entry for the switch to take effect.
 >
 
@@ -559,7 +548,6 @@ The SDK returns this event callback when it starts rendering your first video fr
 |width|Video width|
 
 > **Note**
->
 > 1. The callback of the first local video frame being rendered is triggered only after you call startLocalPreview or startScreenCapture.
 >
 > 2. The callback of the first video frame of a remote user being rendered is triggered only after you call startRemoteView or startRemoteSubStreamView.
@@ -670,7 +658,6 @@ This callback is returned every 2 seconds and notifies you of the statistics on
 |statistics|Statistics, including local statistics and the statistics of remote users. For details, please see TRTCStatistics.|
 
 > **Note**
->
 > If you want to learn about only the current network quality and do not want to spend much time analyzing the statistics returned by this callback, we recommend you use onNetworkQuality.
 >
 
@@ -682,7 +669,7 @@ The callback is triggered by startSpeedTest.
 
 |Param|DESC|
 |---------|---------|
-|result|Speed test data, including loss rates, rtt and bandwidth rates, please refer to TRTCSpeedTestResult for details.|
+|result|Speed test data, including loss rates, rtt and bandwidth rates, please .|
 
 ## onConnectionLost
 
@@ -762,7 +749,6 @@ Note that the SDK returns this callback at the specified interval regardless of
 |userVolumes|An array that represents the volume of all users who are speaking in the room. Value range: [0, 100]|
 
 > **Note**
->
 >  `userVolumes`  is an array. If  `userId`  is empty, the elements in the array represent the volume of the local user’s audio. Otherwise, they represent the volume of a remote user’s audio.
 >
 
@@ -795,7 +781,6 @@ If the sender sets  `reliable`  to  `true` , the SDK will use this callback to n
 |userId|User ID|
 
 > **Note**
->
 > The recipient receives this callback only if the sender sets  `reliable`  to  `true` .
 >
 
@@ -850,7 +835,6 @@ The SDK will then receive the execution result from the CVM and return the resul
 |errMsg|Error message|
 
 > **Note**
->
 > If you receive a callback that the command is executed successfully, it only means that your command was sent to Tencent Cloud’s backend server. If the CDN vendor does not accept your streams, the publishing will still fail.
 >
 
@@ -1026,7 +1010,6 @@ When you call stopLocalRecording to stop local recording, the SDK returns this c
 |width|Screenshot width|
 
 > **Note**
->
 > The parameters of the full-platform C++ interface and the Java interface are different. The C++ interface uses 7 parameters to describe a screenshot, while the Java interface uses only one Bitmap to describe a screenshot.
 >