@tencent-rtc/mcp 1.6.7 → 1.6.8

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.
 >
 

package/resource/knowledge/trtc/android/TXAudioEffectManager.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/android/TXDeviceManager.md

@@ -82,7 +82,6 @@ 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.
 >
 

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

@@ -46,7 +46,7 @@
 | ERR_AUDIO_PLUGIN_INSTALLED_BUT_NEED_RESTART | -1333 | The virtual sound card is installed successfully, but due to the restrictions of macOS, you cannot use it right after installation. Ask users to restart the app upon receiving this error code. |
 | ERR_AUDIO_ENCODE_FAIL | -1304 | Failed to encode audio frames. This may occur if the SDK could not process the custom audio data passed in. |
 | ERR_UNSUPPORTED_SAMPLERATE | -1306 | Unsupported audio sample rate. |
-| 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. |