@tencent-rtc/mcp 1.6.7 → 1.7.0

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.
 >
 

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

@@ -150,7 +150,6 @@ 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.
@@ -252,7 +251,6 @@ 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.
@@ -301,7 +299,6 @@ 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.
 >
 
@@ -320,7 +317,6 @@ 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.
 >
 
@@ -341,7 +337,6 @@ 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.
@@ -452,7 +447,6 @@ 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.
 >
 
@@ -464,7 +458,7 @@ 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
 
@@ -544,7 +538,6 @@ 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.
 >
 
@@ -578,7 +571,6 @@ When users set the mic capturing volume via the UI or a keyboard shortcut, the S
 |volume|System audio capturing volume, which users can set in the audio control panel. Value range: [0, 100]|
 
 > **Note**
->
 > You need to call enableAudioVolumeEvaluation and set the callback interval ( `interval`  > 0) to enable the callback. To disable the callback, set  `interval`  to  `0` .
 >
 
@@ -598,7 +590,6 @@ When users set the system’s playback volume via the UI or a keyboard shortcut,
 |volume|The system playback volume, which users can set in the audio control panel. Value range: 0-100|
 
 > **Note**
->
 > You need to call enableAudioVolumeEvaluation and set the callback interval ( `interval`  > 0) to enable the callback. To disable the callback, set  `interval`  to  `0` .
 >
 
@@ -671,7 +662,6 @@ 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` .
 >
 
@@ -726,7 +716,6 @@ 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.
 >
 
@@ -905,7 +894,6 @@ 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.
 >
 
@@ -1013,7 +1001,6 @@ int onProcessVideoFrame(TRTCVideoFrame *srcFrame, TRTCVideoFrame *dstFrame) {
 |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)
 >
 
@@ -1037,7 +1024,6 @@ 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.
@@ -1069,7 +1055,6 @@ 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.
@@ -1094,7 +1079,6 @@ 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.
 >
 
@@ -1114,7 +1098,6 @@ 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.
@@ -1138,7 +1121,6 @@ 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.

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

@@ -120,7 +120,6 @@ 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.
 >
 
@@ -612,9 +611,9 @@ 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 TRTCSpeedTestScene_OnlineChorusTesting, 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 TRTCSpeedTestScene_OnlineChorusTesting, 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
 
@@ -715,7 +714,7 @@ These parameters are used to specify the layout position information of each vid
 | mixUsersArraySize | Field description: number of elements in the  <code>mixUsersArray</code>  array |
 | 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. |
@@ -765,7 +764,7 @@ 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.
 
@@ -884,7 +883,7 @@ 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/c++/TXLiteAVCode.md

@@ -48,7 +48,7 @@
 | ERR_UNSUPPORTED_SAMPLERATE | -1306 | Unsupported audio sample rate. |
 | ERR_SYSTEM_LOOPBACK_START_FAILED | -1307 | System audio loopback startup encountered an error. Common error reasons and solutions are: 1. The current system version does not support the include or excluded specified process loopback function, which requires Windows system version 10.0.19042 (20H2) or higher. For x64 specified path loopback function, it also requires Windows system version 10.0.19042 (20H2) or higher. Please check the current system version. 2. Invalid input parameters (including empty parameters, non-existent input paths, non-existent process IDs, and non-existent device names). Please check whether the input parameters are legal. 3. No available speaker device, please check whether it is available. |
 | ERR_SYSTEM_LOOPBACK_THIRD_PARTY_APP_EXIT | -1324 | The third party app loopback capture encountered an error. This error occurred after the third party app loopback was successfully started. During the sound collection process, the specified process or application unexpectedly exited. Please check the status of the specified process or application and try again by re-entering the process number |
-| ERR_TRTC_ENTER_ROOM_FAILED | -3301 | Failed to enter the room. For the reason, refer to the error message for -3301 in  <code>onError</code> . |
+| ERR_TRTC_ENTER_ROOM_FAILED | -3301 | Failed to enter the room. For the reason. |
 | ERR_TRTC_REQUEST_IP_TIMEOUT | -3307 | IP and signature request timed out. Check your network connection and whether your firewall allows UDP.<br>Try visiting the IP address 162.14.22.165:8000 or 162.14.6.105:8000 and the domain default-query.trtc.tencent-cloud.com:8000. |
 | ERR_TRTC_CONNECT_SERVER_TIMEOUT | -3308 | Room entry request timed out. Check your network connection and whether VPN is used. You can also switch to 4G to run a test. |
 | ERR_TRTC_ROOM_PARAM_NULL | -3316 | Empty room entry parameters. Please check whether valid parameters were passed in to the  <code>enterRoom:appScene:</code>  API. |