@tencent-rtc/mcp 1.6.7 → 1.6.8

package/resource/knowledge/trtc/c++/ITRTCCloud.md

@@ -131,7 +131,6 @@ 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. If you use  `delete ITRTCCloud*` , a compilation error will occur. Please use destroyTRTCShareInstance to release the object pointer.
 >
 > 2. On Windows, macOS, or iOS, please call the  `getTRTCShareInstance()`  API.
@@ -192,7 +191,6 @@ 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 TRTCAppSceneLIVE or TRTCAppSceneVoiceChatRoom, 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.
@@ -224,7 +222,6 @@ 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 (TRTCAppSceneLIVE) and audio chat room (TRTCAppSceneVoiceChatRoom).
 >
 > 2. If the  `scene`  you specify in enterRoom is TRTCAppSceneVideoCall or TRTCAppSceneAudioCall, please do not call this API.
@@ -245,7 +242,6 @@ 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 (TRTCAppSceneLIVE) and audio chat room (TRTCAppSceneVoiceChatRoom).
 >
 > 2. If the  `scene`  you specify in enterRoom is TRTCAppSceneVideoCall or TRTCAppSceneAudioCall, please do not call this API.
@@ -269,7 +265,6 @@ 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.
@@ -353,7 +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.
@@ -398,7 +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` .
@@ -482,12 +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.
@@ -505,13 +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).
@@ -530,7 +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.
@@ -552,7 +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.
@@ -575,7 +564,6 @@ 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.
@@ -638,7 +626,6 @@ 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.
@@ -677,7 +664,6 @@ 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.
 >
 
@@ -693,7 +679,6 @@ 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.
 >
 
@@ -708,7 +693,6 @@ 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.
@@ -725,7 +709,6 @@ 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.
 >
 
@@ -737,7 +720,7 @@ 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
 
@@ -775,7 +758,6 @@ 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.
 >
 
@@ -797,7 +779,6 @@ 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 enableSmallVideoStream; otherwise, this API will not work.
 >
 
@@ -814,7 +795,6 @@ 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 TRTCSnapshotSourceTypeStream source can be screencaptured currently.
 >
 
@@ -846,7 +826,6 @@ After local audio capturing and publishing is enabled, other users in the room w
 | quality | Sound quality<br>-  TRTCAudioQualitySpeech - Smooth: mono channel; audio bitrate: 18 Kbps. This is suitable for audio call scenarios, such as online meeting and audio call.<br>-  TRTCAudioQualityDefault - 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>-  TRTCAudioQualityMusic - 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.
 >
 
@@ -886,7 +865,6 @@ 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).
 >
 
@@ -901,7 +879,6 @@ 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).
 >
 
@@ -917,7 +894,6 @@ 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.
 >
 
@@ -930,7 +906,6 @@ 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.
 >
 
@@ -951,7 +926,6 @@ 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.
 >
 
@@ -971,7 +945,6 @@ 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.
 >
 
@@ -990,7 +963,6 @@ 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 ITRTCCloudCallback .
 >
 
@@ -1054,7 +1026,6 @@ 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.
@@ -1072,7 +1043,6 @@ 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.
@@ -1141,7 +1111,6 @@ The watermark position is determined by the  `xOffset` ,  `yOffset` , and  `fWid
 |yOffset|Top-left offset on the Y axis of watermark|
 
 > **Note**
->
 > This API only supports adding an image watermark to the primary stream
 >
 
@@ -1178,7 +1147,6 @@ In live music scenarios, an anchor can use this API to have the SDK capture the
 |deviceName|If this parameter is empty, the audio of the entire system is captured.|
 
 > **Note**
->
 > On the Windows platform, you can specify the parameter  `deviceName`  to the absolute path of an executable file (such as  `QQMuisc.exe` ) of a certain application. In this case, the SDK will only capture the sound of that application (32-bit version of the SDK is supported, 64-bit version of the SDK requires Windows version 10.0.19042 or higher).
 >
 > You can also specify  `deviceName`  as the name of a certain speaker device to capture specific speaker sound (you can use the getDevicesList interface in TXDeviceManager to obtain the speaker devices of type TXMediaDeviceTypeSpeaker).
@@ -1223,7 +1191,6 @@ This API can capture the content of the entire screen or a specified application
 |view|Parent control of the rendering control, which can be set to a null value, indicating not to display the preview of the shared screen.|
 
 > **Note**
->
 > 1. A user can publish at most one primary stream (TRTCVideoStreamTypeBig) and one substream (TRTCVideoStreamTypeSub) at the same time.
 >
 > 2. By default, screen sharing uses the substream image. If you want to use the primary stream for screen sharing, you need to stop camera capturing (through stopLocalPreview) in advance to avoid conflicts.
@@ -1242,7 +1209,6 @@ This API can capture the content of the entire screen or a specified application
 **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.
 >
 
@@ -1264,7 +1230,6 @@ Through this API, you can query the IDs, names, and thumbnails of sharable windo
 |thumbnailSize|Specify the thumbnail size of the window to be obtained. The thumbnail can be drawn on the window selection UI.|
 
 > **Note**
->
 > 1. The returned list contains the screen and the application windows. The screen is the first element in the list. If the user has multiple displays, then each display is a sharing target.
 >
 > 2. Please do not use  `delete ITRTCScreenCaptureSourceList*`  to delete the  `SourceList` ; otherwise, crashes may occur. Instead, please use the  `release`  method in ITRTCScreenCaptureSourceList to release the list.
@@ -1298,7 +1263,6 @@ The following four sharing modes are supported:
 |source|Specify sharing source|
 
 > **Note**
->
 > Setting the highlight border color and width parameters does not take effect on macOS.
 >
 
@@ -1340,7 +1304,6 @@ You can set the filtered windows before starting screen sharing or dynamically a
 |window|Window not to be shared|
 
 > **Note**
->
 > 1. This API takes effect only if the  `type`  in TRTCScreenCaptureSourceInfo is specified as TRTCScreenCaptureSourceTypeScreen; that is, the feature of excluding specified windows works only when the entire screen is shared.
 >
 > 2. The windows added to the exclusion list through this API will be automatically cleared by the SDK after room exit.
@@ -1373,7 +1336,6 @@ You can call it before or after startScreenCapture.
 |windowID|Window to be shared (which is a window handle  `HWND`  on Windows)|
 
 > **Note**
->
 > The windows added to the inclusion list by this method will be automatically cleared by the SDK after room exit.
 >
 
@@ -1439,7 +1401,6 @@ For more information, please see [Custom Capturing and Rendering](https://www.te
 |streamType|Specify video stream type (TRTCVideoStreamTypeBig: HD big image; TRTCVideoStreamTypeSub: 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.
@@ -1464,7 +1425,6 @@ 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.
 >
 
@@ -1490,7 +1450,6 @@ 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.
 >
 
@@ -1506,7 +1465,6 @@ 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.
 >
 
@@ -1629,7 +1587,6 @@ After this callback is set, the SDK will skip its own rendering process and call
 |userId|remote user id|
 
 > **Note**
->
 > In actual use, you need to call  `startRemoteView(userid, nullptr)`  to get the video stream of the remote user first (set  `view`  to  `nullptr` ); otherwise, there will be no data called back.
 >
 
@@ -1651,7 +1608,6 @@ 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.
 >
 
@@ -1746,7 +1702,6 @@ 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.
 >
 
@@ -1772,7 +1727,6 @@ 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` .
@@ -1796,7 +1750,6 @@ 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).
@@ -1836,7 +1789,6 @@ 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.
@@ -1865,7 +1817,6 @@ 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.
@@ -1929,7 +1880,6 @@ 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.
 >
 
@@ -1965,10 +1915,8 @@ 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/c++/ITXAudioEffectManager.md

@@ -114,7 +114,6 @@ Given that not all phones deliver excellent in-ear monitoring effects, we have b
 |enable| `true:`  enable;  `false` : disable|
 
 > **Note**
->
 > In-ear monitoring can be enabled only when earphones are used. Please remind anchors to use wired earphones.
 >
 
@@ -129,7 +128,6 @@ This API is used to set the volume of in-ear monitoring.
 |volume|Volume. Value range: [0, 150]; default: 100|
 
 > **Note**
->
 > The volume of the ear monitor is one of the most important factors affecting the user experience. Considering that different users have different sensitivities to volume, it is strongly recommended to provide a ear monitor volume adjustment slider on the UI interface. The recommended adjustment range is between 0 and 150. The gain effect of 100-150 is very helpful for some Android phones with low recording volume.
 >
 
@@ -140,7 +138,6 @@ This API is used to set the volume of in-ear monitoring.
 This API is used to set reverb effects for human voice. For the effects supported, please see TXVoiceReverbType.
 
 > **Note**
->
 > Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.
 >
 
@@ -151,7 +148,6 @@ This API is used to set reverb effects for human voice. For the effects supporte
 This API is used to set voice changing effects. For the effects supported, please see TXVoiceChangeType.
 
 > **Note**
->
 > Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.
 >
 
@@ -166,7 +162,6 @@ This API is used to set the volume of speech. It is often used together with the
 |volume|Volume. Value range: [0, 150]; default: 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.
 >
 
@@ -192,7 +187,6 @@ Before playing background music, please use this API to set the music callback,
 |observer|For more information, please see the APIs defined in  `ITXMusicPlayObserver` .|
 
 > **Note**
->
 > 1. If the ID does not need to be used, the observer can be set to NULL to release it completely.
 >
 
@@ -207,7 +201,6 @@ You must assign an ID to each music track so that you can start, stop, or set th
 |musicParam|Music parameter|
 
 > **Note**
->
 > 1. If you play the same music track multiple times, please use the same ID instead of a separate ID for each playback.
 >
 > 2. If you want to play different music tracks at the same time, use different IDs for them.
@@ -253,7 +246,6 @@ This API is used to set the local and remote playback volume of background music
 |volume|Volume. Value range: [0, 150]; default: 60|
 
 > **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.
 >
 
@@ -269,7 +261,6 @@ This API is used to control the remote playback volume (the volume heard by audi
 |volume|Volume. Value range: [0, 150]; default: 60|
 
 > **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.
 >
 
@@ -285,7 +276,6 @@ This API is used to control the local playback volume (the volume heard by ancho
 |volume|Volume. Value range: [0, 150]. default: 60|
 
 > **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.
 >
 
@@ -339,7 +329,6 @@ The length of the specified music file is returned. -1 indicates failure to get
 |pts|Unit: millisecond|
 
 > **Note**
->
 > Do not call this API frequently as the music file may be read and written to each time the API is called, which can be time-consuming.
 >
 > Wait till users finish dragging the progress bar before you call this API.
@@ -357,7 +346,6 @@ The length of the specified music file is returned. -1 indicates failure to get
 |scratchSpeedRate|Scratch disc speed, the default value is 1.0f, the range is: a floating point number between [-12.0 ~ 12.0], the positive/negative speed value indicates the direction is positive/negative, and the absolute value indicates the speed.|
 
 > **Note**
->
 > Precondition preloadMusic succeeds.
 >
 
@@ -382,7 +370,6 @@ You must assign an ID to each music track so that you can start, stop, or set th
 |musicParam|Music parameter|
 
 > **Note**
->
 > 1. Preload supports up to 2 preloads with different IDs at the same time, and the preload time does not exceed 10 minutes,you need to call stopPlayMusic API after use, otherwise the memory will not be released.
 >
 > 2. If the music corresponding to the ID is being played, the preloading fails, and stopPlayMusic must be called first.
@@ -408,7 +395,6 @@ You must assign an ID to each music track so that you can start, stop, or set th
 |index|Specify which track to play (the first track is played by default). Value range 0, total number of tracks).|
 
 > **Note**
->
 > The total number of tracks can be obtained through the [getMusicTrackCount interface.
 >
 

package/resource/knowledge/trtc/c++/ITXDeviceManager.md

@@ -106,7 +106,6 @@ This API can be used to achieve the following:
 |position|The spot to focus on. Pass in the coordinates of the spot you want to focus on.|
 
 > **Note**
->
 > Before using this API, you must first disable auto focus using enableCameraAutoFocus.
 >
 
@@ -137,7 +136,6 @@ If the audio route is set to the speaker, the volume is relatively high, and the
 |type|Device type. Set it to the type of device you want to get. For details, please see the definition of TXMediaDeviceType.|
 
 > **Note**
->
 > -  To ensure that the SDK can manage the lifecycle of the  `ITXDeviceCollection`  object, after using this API, please call the  `release`  method to release the resources.
 > -  Do not use  `delete`  to release the Collection object returned as deleting the ITXDeviceCollection* pointer will cause crash.
 > -  The valid values of  `type`  are  `TXMediaDeviceTypeMic` ,  `TXMediaDeviceTypeSpeaker` , and  `TXMediaDeviceTypeCamera` .
@@ -202,7 +200,6 @@ This API is used to set the microphone and speaker types. Camera following the s
 **Starting camera testing (for desktop OS).**
 
 > **Note**
->
 > You can use the setCurrentDevice API to switch between cameras during testing.
 >
 
@@ -221,7 +218,6 @@ This API is used to test whether the mic functions properly. The mic volume dete
 |interval|Interval of volume callbacks, in milliseconds.|
 
 > **Note**
->
 > When this interface is called, the sound recorded by the microphone will be played back to the speakers by default.
 >