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Tencent Real-Time Communication
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    TRTCCloudCallback

    Last updated: 2024-11-28 17:44:44
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      Copyright (c) 2021 Tencent. All rights reserved.
      
      Module: ITRTCCloudCallback @ TXLiteAVSDK
      
      Function: event callback APIs for TRTC’s video call feature
      
      TRTCCloudCallback

      ITRTCCloudCallback

      FuncList
      DESC
      Error event callback
      Warning event callback
      Whether room entry is successful
      Room exit
      Role switching
      Result of room switching
      Result of requesting cross-room call
      Result of ending cross-room call
      Result of changing the upstream capability of the cross-room anchor
      A user entered the room
      A user exited the room
      A remote user published/unpublished primary stream video
      A remote user published/unpublished substream video
      A remote user published/unpublished audio
      The SDK started rendering the first video frame of the local or a remote user
      The SDK started playing the first audio frame of a remote user
      The first local video frame was published
      The first local audio frame was published
      Change of remote video status
      Change of remote audio status
      Change of remote video size
      Real-time network quality statistics
      Real-time statistics on technical metrics
      Callback of network speed test
      The SDK was disconnected from the cloud
      The SDK is reconnecting to the cloud
      The SDK is reconnected to the cloud
      The camera is ready
      The mic is ready
      Volume
      The status of a local device changed (for desktop OS only)
      The capturing volume of the mic changed
      The playback volume changed
      Whether system audio capturing is enabled successfully (for desktop OS only)
      Volume during mic test
      Volume during speaker test
      Receipt of custom message
      Loss of custom message
      Receipt of SEI message
      Started publishing to Tencent Cloud CSS CDN
      Stopped publishing to Tencent Cloud CSS CDN
      Started publishing to non-Tencent Cloud’s live streaming CDN
      Stopped publishing to non-Tencent Cloud’s live streaming CDN
      Set the layout and transcoding parameters for On-Cloud MixTranscoding
      Callback for starting to publish
      Callback for modifying publishing parameters
      Callback for stopping publishing
      Callback for change of RTMP/RTMPS publishing status
      Screen sharing started
      Screen sharing was paused
      Screen sharing was resumed
      Screen sharing stopped
      The shared window was covered (for Windows only)
      Local recording started
      Local media is being recorded
      Record fragment finished.
      Local recording stopped
      Finished taking a local screenshot
      An anchor entered the room (disused)
      An anchor left the room (disused)
      Audio effects ended (disused)
      Started playing background music (disused)
      Playback progress of background music (disused)
      Background music stopped (disused)
      Result of server speed testing (disused)

      ITRTCVideoRenderCallback

      FuncList
      DESC
      Custom video rendering

      ITRTCVideoFrameCallback

      FuncList
      DESC
      An OpenGL context was created in the SDK.
      Video processing by third-party beauty filters
      The OpenGL context in the SDK was destroyed

      ITRTCAudioFrameCallback

      FuncList
      DESC
      Audio data captured by the local mic and pre-processed by the audio module
      Audio data captured by the local mic, pre-processed by the audio module, effect-processed and BGM-mixed
      Audio data of each remote user before audio mixing
      Data mixed from each channel before being submitted to the system for playback
      Data mixed from all the captured and to-be-played audio in the SDK

      ITRTCLogCallback

      FuncList
      DESC
      Printing of local log

      onError

      onError
      void onError
      (TXLiteAVError errCode
      
      const char* errMsg
      
      void* extraInfo)

      Error event callback

      Error event, which indicates that the SDK threw an irrecoverable error such as room entry failure or failure to start device
      For more information, see Error Codes.
      Param
      DESC
      errCode
      Error code
      errMsg
      Error message
      extInfo
      Extended field. Certain error codes may carry extra information for troubleshooting.

      onWarning

      onWarning
      void onWarning
      (TXLiteAVWarning warningCode
      
      const char* warningMsg
      
      void* extraInfo)

      Warning event callback

      Warning event, which indicates that the SDK threw an error requiring attention, such as video lag or high CPU usage
      For more information, see Error Codes.
      Param
      DESC
      extInfo
      Extended field. Certain warning codes may carry extra information for troubleshooting.
      warningCode
      Warning code
      warningMsg
      Warning message

      onEnterRoom

      onEnterRoom
      void onEnterRoom
      (int result)

      Whether room entry is successful

      After calling the enterRoom() API in TRTCCloud to enter a room, you will receive the onEnterRoom(result) callback from TRTCCloudDelegate .
      If room entry succeeded, result will be a positive number ( result > 0), indicating the time in milliseconds (ms) the room entry takes.
      If room entry failed, result will be a negative number (result < 0), indicating the error code for the failure.
      For more information on the error codes for room entry failure, see Error Codes.
      Param
      DESC
      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.

      onExitRoom

      onExitRoom
      void onExitRoom
      (int reason)

      Room exit

      Calling the exitRoom() API in TRTCCloud will trigger the execution of room exit-related logic, such as releasing resources of audio/video devices and codecs.
      After all resources occupied by the SDK are released, the SDK will return the onExitRoom() callback.
      
      If you need to call enterRoom() again or switch to another audio/video SDK, please wait until you receive the onExitRoom() callback.
      Otherwise, you may encounter problems such as the camera or mic being occupied.
      Param
      DESC
      reason
      Reason for room exit. 0 : the user called exitRoom to exit the room; 1 : the user was removed from the room by the server; 2 : the room was dismissed.

      onSwitchRole

      onSwitchRole
      void onSwitchRole
      (TXLiteAVError errCode
      
      const char* errMsg)

      Role switching

      You can call the switchRole() API in TRTCCloud to switch between the anchor and audience roles. This is accompanied by a line switching process.
      After the switching, the SDK will return the onSwitchRole() event callback.
      Param
      DESC
      errCode
      Error code. ERR_NULL indicates a successful switch. For more information, please see Error Codes.
      errMsg
      Error message

      onSwitchRoom

      onSwitchRoom
      void onSwitchRoom
      (TXLiteAVError errCode
      
      const char* errMsg)

      Result of room switching

      You can call the switchRoom() API in TRTCCloud to switch from one room to another.
      After the switching, the SDK will return the onSwitchRoom() event callback.
      Param
      DESC
      errCode
      Error code. ERR_NULL indicates a successful switch. For more information, please see Error Codes.
      errMsg
      Error message

      onConnectOtherRoom

      onConnectOtherRoom
      void onConnectOtherRoom
      (const char* userId
      
      TXLiteAVError errCode
      
      const char* errMsg)

      Result of requesting cross-room call

      You can call the connectOtherRoom() API in TRTCCloud to establish a video call with the anchor of another room. This is the “anchor competition” feature.
      The caller will receive the onConnectOtherRoom() callback, which can be used to determine whether the cross-room call is successful.
      If it is successful, all users in either room will receive the onUserVideoAvailable() callback from the anchor of the other room.
      Param
      DESC
      errCode
      Error code. ERR_NULL indicates that cross-room connection is established successfully. For more information, please see Error Codes.
      errMsg
      Error message
      userId
      The user ID of the anchor (in another room) to be called

      onDisconnectOtherRoom

      onDisconnectOtherRoom
      void onDisconnectOtherRoom
      (TXLiteAVError errCode
      
      const char* errMsg)

      Result of ending cross-room call

      onUpdateOtherRoomForwardMode

      onUpdateOtherRoomForwardMode
      void onUpdateOtherRoomForwardMode
      (TXLiteAVError errCode
      
      const char* errMsg)

      Result of changing the upstream capability of the cross-room anchor

      onRemoteUserEnterRoom

      onRemoteUserEnterRoom
      void onRemoteUserEnterRoom
      (const char* userId)

      A user entered the room

      Due to performance concerns, this callback works differently in different scenarios (i.e., AppScene , which you can specify by setting the second parameter when calling enterRoom ).
      Live streaming scenarios ( TRTCAppSceneLIVE or TRTCAppSceneVoiceChatRoom ): in live streaming scenarios, a user is either in the role of an anchor or audience. The callback is returned only when an anchor enters the room.
      Call scenarios ( TRTCAppSceneVideoCall or TRTCAppSceneAudioCall ): in call scenarios, the concept of roles does not apply (all users can be considered as anchors), and the callback is returned when any user enters the room.
      Param
      DESC
      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.

      onRemoteUserLeaveRoom

      onRemoteUserLeaveRoom
      void onRemoteUserLeaveRoom
      (const char* userId
      
      int reason)

      A user exited the room

      As with onRemoteUserEnterRoom , this callback works differently in different scenarios (i.e., AppScene , which you can specify by setting the second parameter when calling enterRoom ).
      Live streaming scenarios ( TRTCAppSceneLIVE or TRTCAppSceneVoiceChatRoom ): the callback is triggered only when an anchor exits the room.
      Call scenarios ( TRTCAppSceneVideoCall or TRTCAppSceneAudioCall ): in call scenarios, the concept of roles does not apply, and the callback is returned when any user exits the room.
      Param
      DESC
      reason
      Reason for room exit. 0 : the user exited the room voluntarily; 1 : the user exited the room due to timeout; 2 : the user was removed from the room; 3 : the anchor user exited the room due to switch to audience.
      userId
      User ID of the remote user

      onUserVideoAvailable

      onUserVideoAvailable
      void onUserVideoAvailable
      (const char* userId
      
      bool available)

      A remote user published/unpublished primary stream video

      The primary stream is usually used for camera images. If you receive the onUserVideoAvailable(userId, true) callback, it indicates that the user has available primary stream video.
      You can then call startRemoteView to subscribe to the remote user’s video. If the subscription is successful, you will receive the onFirstVideoFrame(userid) callback, which indicates that the first video frame of the user is rendered.
      
      If you receive the onUserVideoAvailable(userId, false) callback, it indicates that the video of the remote user is disabled, which may be because the user called muteLocalVideo or stopLocalPreview.
      Param
      DESC
      available
      Whether the user published (or unpublished) primary stream video. true : published; false : unpublished
      userId
      User ID of the remote user

      onUserSubStreamAvailable

      onUserSubStreamAvailable
      void onUserSubStreamAvailable
      (const char* userId
      
      bool available)

      A remote user published/unpublished substream video

      The substream is usually used for screen sharing images. If you receive the onUserSubStreamAvailable(userId, true) callback, it indicates that the user has available substream video.
      You can then call startRemoteView to subscribe to the remote user’s video. If the subscription is successful, you will receive the onFirstVideoFrame(userid) callback, which indicates that the first frame of the user is rendered.
      Param
      DESC
      available
      Whether the user published (or unpublished) substream video. true : published; false : unpublished
      userId
      User ID of the remote user
      Note
      The API used to display substream images is startRemoteView, not startRemoteSubStreamView, startRemoteSubStreamView is deprecated.

      onUserAudioAvailable

      onUserAudioAvailable
      void onUserAudioAvailable
      (const char* userId
      
      bool available)

      A remote user published/unpublished audio

      If you receive the onUserAudioAvailable(userId, true) callback, it indicates that the user published audio.
      In auto-subscription mode, the SDK will play the user’s audio automatically.
      In manual subscription mode, you can call muteRemoteAudio(userid, false) to play the user’s audio.
      Param
      DESC
      available
      Whether the user published (or unpublished) audio. true : published; false : unpublished
      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.

      onFirstVideoFrame

      onFirstVideoFrame
      void onFirstVideoFrame
      (const char* userId
      
      const TRTCVideoStreamType streamType
      
      const int width
      
      const int height)

      The SDK started rendering the first video frame of the local or a remote user

      The SDK returns this event callback when it starts rendering your first video frame or that of a remote user. The userId in the callback can help you determine whether the frame is yours or a remote user’s.
      If userId is empty, it indicates that the SDK has started rendering your first video frame. The precondition is that you have called startLocalPreview or startScreenCapture.
      If userId is not empty, it indicates that the SDK has started rendering the first video frame of a remote user. The precondition is that you have called startRemoteView to subscribe to the user’s video.
      Param
      DESC
      height
      Video height
      streamType
      Video stream type. The primary stream ( Main ) is usually used for camera images, and the substream ( Sub ) for screen sharing images.
      userId
      The user ID of the local or a remote user. If it is empty, it indicates that the first local video frame is available; if it is not empty, it indicates that the first video frame of a remote user is available.
      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.

      onFirstAudioFrame

      onFirstAudioFrame
      void onFirstAudioFrame
      (const char* userId)

      The SDK started playing the first audio frame of a remote user

      The SDK returns this callback when it plays the first audio frame of a remote user. The callback is not returned for the playing of the first audio frame of the local user.
      Param
      DESC
      userId
      User ID of the remote user

      onSendFirstLocalVideoFrame

      onSendFirstLocalVideoFrame
      void onSendFirstLocalVideoFrame
      (const TRTCVideoStreamType streamType)

      The first local video frame was published

      After you enter a room and call startLocalPreview or startScreenCapture to enable local video capturing (whichever happens first),
      the SDK will start video encoding and publish the local video data via its network module to the cloud.
      It returns the onSendFirstLocalVideoFrame callback after publishing the first local video frame.
      Param
      DESC
      streamType
      Video stream type. The primary stream ( Main ) is usually used for camera images, and the substream ( Sub ) for screen sharing images.

      onSendFirstLocalAudioFrame

      onSendFirstLocalAudioFrame

      The first local audio frame was published

      After you enter a room and call startLocalAudio to enable audio capturing (whichever happens first),
      the SDK will start audio encoding and publish the local audio data via its network module to the cloud.
      The SDK returns the onSendFirstLocalAudioFrame callback after sending the first local audio frame.

      onRemoteVideoStatusUpdated

      onRemoteVideoStatusUpdated
      void onRemoteVideoStatusUpdated
      (const char* userId
      
      TRTCVideoStreamType streamType
      
      TRTCAVStatusType status
      
      
      void *extrainfo)

      Change of remote video status

      You can use this callback to get the status ( Playing , Loading , or Stopped ) of the video of each remote user and display it on the UI.
      Param
      DESC
      extraInfo
      Extra information
      reason
      Reason for the change of status
      status
      Video status, which may be Playing , Loading , or Stopped
      streamType
      Video stream type. The primary stream ( Main ) is usually used for camera images, and the substream ( Sub ) for screen sharing images.
      userId
      User ID

      onRemoteAudioStatusUpdated

      onRemoteAudioStatusUpdated
      void onRemoteAudioStatusUpdated
      (const char* userId
      
      TRTCAVStatusType status
      
      
      void *extrainfo)

      Change of remote audio status

      You can use this callback to get the status ( Playing , Loading , or Stopped ) of the audio of each remote user and display it on the UI.
      Param
      DESC
      extraInfo
      Extra information
      reason
      Reason for the change of status
      status
      Audio status, which may be Playing , Loading , or Stopped
      userId
      User ID

      onUserVideoSizeChanged

      onUserVideoSizeChanged
      void onUserVideoSizeChanged
      (const char* userId
      
      TRTCVideoStreamType streamType
      
      int newWidth
      
      int newHeight)

      Change of remote video size

      If you receive the onUserVideoSizeChanged(userId, streamtype, newWidth, newHeight) callback, it indicates that the user changed the video size. It may be triggered by setVideoEncoderParam or setSubStreamEncoderParam .
      Param
      DESC
      newHeight
      Video height
      newWidth
      Video width
      streamType
      Video stream type. The primary stream ( Main ) is usually used for camera images, and the substream ( Sub ) for screen sharing images.
      userId
      User ID

      onNetworkQuality

      onNetworkQuality
      void onNetworkQuality
      (TRTCQualityInfo localQuality
      
      TRTCQualityInfo* remoteQuality
      
      uint32_t remoteQualityCount)

      Real-time network quality statistics

      This callback is returned every 2 seconds and notifies you of the upstream and downstream network quality detected by the SDK.
      The SDK uses a built-in proprietary algorithm to assess the current latency, bandwidth, and stability of the network and returns a result.
      If the result is 1 (excellent), it means that the current network conditions are excellent; if it is 6 (down), it means that the current network conditions are too bad to support TRTC calls.
      Param
      DESC
      localQuality
      Upstream network quality
      remoteQuality
      Downstream network quality, it refers to the data quality finally measured on the local side after the data flow passes through a complete transmission link of "remote ->cloud ->local". Therefore, the downlink network quality here represents the joint impact of the remote uplink and the local downlink.
      Note
      The uplink quality of remote users cannot be determined independently through this interface.

      onStatistics

      onStatistics
      void onStatistics
      (const TRTCStatistics& statistics)

      Real-time statistics on technical metrics

      This callback is returned every 2 seconds and notifies you of the statistics on technical metrics related to video, audio, and network. The metrics are listed in TRTCStatistics:
      Video statistics: video resolution ( resolution ), frame rate ( FPS ), bitrate ( bitrate ), etc.
      Audio statistics: audio sample rate ( samplerate ), number of audio channels ( channel ), bitrate ( bitrate ), etc.
      Network statistics: the round trip time ( rtt ) between the SDK and the cloud (SDK -> Cloud -> SDK), package loss rate ( loss ), upstream traffic ( sentBytes ), downstream traffic ( receivedBytes ), etc.
      Param
      DESC
      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.

      onSpeedTestResult

      onSpeedTestResult
      void onSpeedTestResult
      (const TRTCSpeedTestResult& result)

      Callback of network speed test

      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.

      onConnectionLost

      onConnectionLost

      The SDK was disconnected from the cloud

      The SDK returns this callback when it is disconnected from the cloud, which may be caused by network unavailability or change of network, for example, when the user walks into an elevator.
      After returning this callback, the SDK will attempt to reconnect to the cloud, and will return the onTryToReconnect callback. When it is reconnected, it will return the onConnectionRecovery callback.
      In other words, the SDK proceeds from one event to the next in the following order:
      

      onTryToReconnect

      onTryToReconnect

      The SDK is reconnecting to the cloud

      When the SDK is disconnected from the cloud, it returns the onConnectionLost callback. It then attempts to reconnect and returns this callback (onTryToReconnect). After it is reconnected, it returns the onConnectionRecovery callback.

      onConnectionRecovery

      onConnectionRecovery

      The SDK is reconnected to the cloud

      When the SDK is disconnected from the cloud, it returns the onConnectionLost callback. It then attempts to reconnect and returns the onTryToReconnect callback. After it is reconnected, it returns this callback (onConnectionRecovery).

      onCameraDidReady

      onCameraDidReady

      The camera is ready

      After you call startLocalPreivew, the SDK will try to start the camera and return this callback if the camera is started.
      If it fails to start the camera, it’s probably because the application does not have access to the camera or the camera is being used.
      You can capture the onError callback to learn about the exception and let users know via UI messages.

      onMicDidReady

      onMicDidReady

      The mic is ready

      After you call startLocalAudio, the SDK will try to start the mic and return this callback if the mic is started.
      If it fails to start the mic, it’s probably because the application does not have access to the mic or the mic is being used.
      You can capture the onError callback to learn about the exception and let users know via UI messages.

      onUserVoiceVolume

      onUserVoiceVolume
      void onUserVoiceVolume
      (TRTCVolumeInfo* userVolumes
      
      uint32_t userVolumesCount
      
      uint32_t totalVolume)

      Volume

      The SDK can assess the volume of each channel and return this callback on a regular basis. You can display, for example, a waveform or volume bar on the UI based on the statistics returned.
      You need to first call enableAudioVolumeEvaluation to enable the feature and set the interval for the callback.
      Note that the SDK returns this callback at the specified interval regardless of whether someone is speaking in the room.
      Param
      DESC
      totalVolume
      The total volume of all remote users. Value range: 0-100
      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.

      onDeviceChange

      onDeviceChange
      void onDeviceChange
      (const char* deviceId
      
      TRTCDeviceType type
      
      TRTCDeviceState state)

      The status of a local device changed (for desktop OS only)

      The SDK returns this callback when a local device (camera, mic, or speaker) is connected or disconnected.
      Param
      DESC
      deviceId
      Device ID
      deviceType
      Device type. 0 : microphone; 1 : speak; 2 : camera
      state
      Device status. 0 : connected; 1 : disconnected; 2 : started

      onAudioDeviceCaptureVolumeChanged

      onAudioDeviceCaptureVolumeChanged
      void onAudioDeviceCaptureVolumeChanged
      (uint32_t volume
      
      bool muted)

      The capturing volume of the mic changed

      On desktop OS such as macOS and Windows, users can set the capturing volume of the mic in the audio control panel.
      The higher volume a user sets, the higher the volume of raw audio captured by the mic.
      On some keyboards and laptops, users can also mute the mic by pressing a key (whose icon is a crossed out mic).
      When users set the mic capturing volume via the UI or a keyboard shortcut, the SDK will return this callback.
      Param
      DESC
      muted
      Whether the mic is muted. true : muted; false : unmuted
      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 .

      onAudioDevicePlayoutVolumeChanged

      onAudioDevicePlayoutVolumeChanged
      void onAudioDevicePlayoutVolumeChanged
      (uint32_t volume
      
      bool muted)

      The playback volume changed

      On desktop OS such as macOS and Windows, users can set the system’s playback volume in the audio control panel.
      On some keyboards and laptops, users can also mute the speaker by pressing a key (whose icon is a crossed out speaker).
      
      When users set the system’s playback volume via the UI or a keyboard shortcut, the SDK will return this callback.
      Param
      DESC
      muted
      Whether the speaker is muted. true : muted; false : unmuted
      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 .

      onSystemAudioLoopbackError

      onSystemAudioLoopbackError
      void onSystemAudioLoopbackError
      (TXLiteAVError errCode)

      Whether system audio capturing is enabled successfully (for desktop OS only)

      On macOS, you can call startSystemAudioLoopback to install an audio driver and have the SDK capture the audio played back by the system.
      On Windows systems, you can use startSystemAudioLoopback to have the SDK capture the audio played back by the system.
      In use cases such as video teaching and music live streaming, the teacher can use this feature to let the SDK capture the sound of the video played by his or her computer, so that students in the room can hear the sound too.
      The SDK returns this callback after trying to enable system audio capturing. To determine whether it is actually enabled, pay attention to the error parameter in the callback.
      Param
      DESC
      err
      If it is ERR_NULL , system audio capturing is enabled successfully. Otherwise, it is not.

      onTestMicVolume

      onTestMicVolume
      void onTestMicVolume
      (uint32_t volume)

      Volume during mic test

      When you call startMicDeviceTest to test the mic, the SDK will keep returning this callback. The volume parameter represents the volume of the audio captured by the mic.
      If the value of the volume parameter fluctuates, the mic works properly. If it is 0 throughout the test, it indicates that there is a problem with the mic, and users should be prompted to switch to a different mic.
      Param
      DESC
      volume
      Captured mic volume. Value range: 0-100

      onTestSpeakerVolume

      onTestSpeakerVolume
      void onTestSpeakerVolume
      (uint32_t volume)

      Volume during speaker test

      When you call startSpeakerDeviceTest to test the speaker, the SDK will keep returning this callback.
      The volume parameter in the callback represents the volume of audio sent by the SDK to the speaker for playback. If its value fluctuates but users cannot hear any sound, the speaker is not working properly.
      Param
      DESC
      volume
      The volume of audio sent by the SDK to the speaker for playback. Value range: 0-100

      onRecvCustomCmdMsg

      onRecvCustomCmdMsg
      void onRecvCustomCmdMsg
      (const char* userId
      
      int32_t cmdID
      
      uint32_t seq
      
      const uint8_t* message
      
      uint32_t messageSize)

      Receipt of custom message

      When a user in a room uses sendCustomCmdMsg to send a custom message, other users in the room can receive the message through the onRecvCustomCmdMsg callback.
      Param
      DESC
      cmdID
      Command ID
      message
      Message data
      seq
      Message serial number
      userId
      User ID

      onMissCustomCmdMsg

      onMissCustomCmdMsg
      void onMissCustomCmdMsg
      (const char* userId
      
      int32_t cmdID
      
      int32_t errCode
      
      int32_t missed)

      Loss of custom message

      When you use sendCustomCmdMsg to send a custom UDP message, even if you enable reliable transfer (by setting reliable to true ), there is still a chance of message loss. Reliable transfer only helps maintain a low probability of message loss, which meets the reliability requirements in most cases.
      If the sender sets reliable to true , the SDK will use this callback to notify the recipient of the number of custom messages lost during a specified time period (usually 5s) in the past.
      Param
      DESC
      cmdID
      Command ID
      errCode
      Error code
      missed
      Number of lost messages
      userId
      User ID
      Note
      The recipient receives this callback only if the sender sets reliable to true .

      onRecvSEIMsg

      onRecvSEIMsg
      void onRecvSEIMsg
      (const char* userId
      
      const uint8_t* message
      
      uint32_t messageSize)

      Receipt of SEI message

      If a user in the room uses sendSEIMsg to send an SEI message via video frames, other users in the room can receive the message through the onRecvSEIMsg callback.
      Param
      DESC
      message
      Data
      userId
      User ID

      onStartPublishing

      onStartPublishing
      void onStartPublishing
      (int err
      
      const char *errMsg)

      Started publishing to Tencent Cloud CSS CDN

      When you call startPublishing to publish streams to Tencent Cloud CSS CDN, the SDK will sync the command to the CVM immediately.
      The SDK will then receive the execution result from the CVM and return the result to you via this callback.
      Param
      DESC
      err
      0 : successful; other values: failed
      errMsg
      Error message

      onStopPublishing

      onStopPublishing
      void onStopPublishing
      (int err
      
      const char *errMsg)

      Stopped publishing to Tencent Cloud CSS CDN

      When you call stopPublishing to stop publishing streams to Tencent Cloud CSS CDN, the SDK will sync the command to the CVM immediately.
      The SDK will then receive the execution result from the CVM and return the result to you via this callback.
      Param
      DESC
      err
      0 : successful; other values: failed
      errMsg
      Error message

      onStartPublishCDNStream

      onStartPublishCDNStream
      void onStartPublishCDNStream
      (int errCode
      
      const char* errMsg)

      Started publishing to non-Tencent Cloud’s live streaming CDN

      When you call startPublishCDNStream to start publishing streams to a non-Tencent Cloud’s live streaming CDN, the SDK will sync the command to the CVM immediately.
      The SDK will then receive the execution result from the CVM and return the result to you via this callback.
      Param
      DESC
      err
      0 : successful; other values: failed
      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.

      onStopPublishCDNStream

      onStopPublishCDNStream
      void onStopPublishCDNStream
      (int errCode
      
      const char* errMsg)

      Stopped publishing to non-Tencent Cloud’s live streaming CDN

      When you call stopPublishCDNStream to stop publishing to a non-Tencent Cloud’s live streaming CDN, the SDK will sync the command to the CVM immediately.
      The SDK will then receive the execution result from the CVM and return the result to you via this callback.
      Param
      DESC
      err
      0 : successful; other values: failed
      errMsg
      Error message

      onSetMixTranscodingConfig

      onSetMixTranscodingConfig
      void onSetMixTranscodingConfig
      (int err
      
      const char* errMsg)

      Set the layout and transcoding parameters for On-Cloud MixTranscoding

      When you call setMixTranscodingConfig to modify the layout and transcoding parameters for On-Cloud MixTranscoding, the SDK will sync the command to the CVM immediately.
      The SDK will then receive the execution result from the CVM and return the result to you via this callback.
      Param
      DESC
      err
      0 : successful; other values: failed
      errMsg
      Error message

      onStartPublishMediaStream

      onStartPublishMediaStream
      void onStartPublishMediaStream
      (const char* taskId
      
      int code
      
      const char* message
      
      void* extraInfo)

      Callback for starting to publish

      When you call startPublishMediaStream to publish a stream to the TRTC backend, the SDK will immediately update the command to the cloud server.
      The SDK will then receive the publishing result from the cloud server and will send the result to you via this callback.
      Param
      DESC
      code
      : 0 : Successful; other values: Failed.
      extraInfo
      : Additional information. For some error codes, there may be additional information to help you troubleshoot the issues.
      message
      : The callback information.
      taskId
      : If a request is successful, a task ID will be returned via the callback. You need to provide this task ID when you call updatePublishMediaStream to modify publishing parameters or stopPublishMediaStream to stop publishing.

      onUpdatePublishMediaStream

      onUpdatePublishMediaStream
      void onUpdatePublishMediaStream
      (const char* taskId
      
      int code
      
      const char* message
      
      void* extraInfo)

      Callback for modifying publishing parameters

      When you call updatePublishMediaStream to modify publishing parameters, the SDK will immediately update the command to the cloud server.
      The SDK will then receive the modification result from the cloud server and will send the result to you via this callback.
      Param
      DESC
      code
      : 0 : Successful; other values: Failed.
      extraInfo
      : Additional information. For some error codes, there may be additional information to help you troubleshoot the issues.
      message
      : The callback information.
      taskId
      : The task ID you pass in when calling updatePublishMediaStream, which is used to identify a request.

      onStopPublishMediaStream

      onStopPublishMediaStream
      void onStopPublishMediaStream
      (const char* taskId
      
      int code
      
      const char* message
      
      void* extraInfo)

      Callback for stopping publishing

      When you call stopPublishMediaStream to stop publishing, the SDK will immediately update the command to the cloud server.
      The SDK will then receive the modification result from the cloud server and will send the result to you via this callback.
      Param
      DESC
      code
      : 0 : Successful; other values: Failed.
      extraInfo
      : Additional information. For some error codes, there may be additional information to help you troubleshoot the issues.
      message
      : The callback information.
      taskId
      : The task ID you pass in when calling stopPublishMediaStream, which is used to identify a request.

      onCdnStreamStateChanged

      onCdnStreamStateChanged
      void onCdnStreamStateChanged
      (const char* cdnUrl
      
      int status
      
      int code
      
      const char* msg
      
      void* extraInfo)

      Callback for change of RTMP/RTMPS publishing status

      When you call startPublishMediaStream to publish a stream to the TRTC backend, the SDK will immediately update the command to the cloud server.
      If you set the publishing destination (TRTCPublishTarget) to the URL of Tencent Cloud or a third-party CDN, you will be notified of the RTMP/RTMPS publishing status via this callback.
      Param
      DESC
      cdnUrl
      : The URL you specify in TRTCPublishTarget when you call startPublishMediaStream.
      code
      : The publishing result. 0 : Successful; other values: Failed.
      extraInfo
      : Additional information. For some error codes, there may be additional information to help you troubleshoot the issues.
      message
      : The publishing information.
      status
      : The publishing status.
      0: The publishing has not started yet or has ended. This value will be returned after you call stopPublishMediaStream.
      1: The TRTC server is connecting to the CDN server. If the first attempt fails, the TRTC backend will retry multiple times and will return this value via the callback (every five seconds). After publishing succeeds, the value 2 will be returned. If a server error occurs or publishing is still unsuccessful after 60 seconds, the value 4 will be returned.
      2: The TRTC server is publishing to the CDN. This value will be returned if the publishing succeeds.
      3: The TRTC server is disconnected from the CDN server and is reconnecting. If a CDN error occurs or publishing is interrupted, the TRTC backend will try to reconnect and resume publishing and will return this value via the callback (every five seconds). After publishing resumes, the value 2 will be returned. If a server error occurs or the attempt to resume publishing is still unsuccessful after 60 seconds, the value 4 will be returned.
      4: The TRTC server is disconnected from the CDN server and failed to reconnect within the timeout period. In this case, the publishing is deemed to have failed. You can call updatePublishMediaStream to try again.
      5: The TRTC server is disconnecting from the CDN server. After you call stopPublishMediaStream, the SDK will return this value first and then the value 0 .

      onScreenCaptureStarted

      onScreenCaptureStarted

      Screen sharing started

      The SDK returns this callback when you call startScreenCapture and other APIs to start screen sharing.

      onScreenCapturePaused

      onScreenCapturePaused
      void onScreenCapturePaused
      (int reason)

      Screen sharing was paused

      The SDK returns this callback when you call pauseScreenCapture to pause screen sharing.
      Param
      DESC
      reason
      Reason.
      0 : the user paused screen sharing.
      1 : screen sharing was paused because the shared window became invisible(Mac). screen sharing was paused because setting parameters(Windows).
      2 : screen sharing was paused because the shared window became minimum(only for Windows).
      3 : screen sharing was paused because the shared window became invisible(only for Windows).

      onScreenCaptureResumed

      onScreenCaptureResumed
      void onScreenCaptureResumed
      (int reason)

      Screen sharing was resumed

      The SDK returns this callback when you call resumeScreenCapture to resume screen sharing.
      Param
      DESC
      reason
      Reason.
      0 : the user resumed screen sharing.
      1 : screen sharing was resumed automatically after the shared window became visible again(Mac). screen sharing was resumed automatically after setting parameters(Windows).
      2 : screen sharing was resumed automatically after the shared window became minimize recovery(only for Windows).
      3 : screen sharing was resumed automatically after the shared window became visible again(only for Windows).

      onScreenCaptureStoped

      onScreenCaptureStoped
      void onScreenCaptureStoped
      (int reason)

      Screen sharing stopped

      The SDK returns this callback when you call stopScreenCapture to stop screen sharing.
      Param
      DESC
      reason
      Reason. 0 : the user stopped screen sharing; 1 : screen sharing stopped because the shared window was closed.

      onScreenCaptureCovered

      onScreenCaptureCovered

      The shared window was covered (for Windows only)

      The SDK returns this callback when the shared window is covered and cannot be captured. Upon receiving this callback, you can prompt users via the UI to move and expose the window.

      onLocalRecordBegin

      onLocalRecordBegin
      void onLocalRecordBegin
      (int errCode
      
      const char* storagePath)

      Local recording started

      When you call startLocalRecording to start local recording, the SDK returns this callback to notify you whether recording is started successfully.
      Param
      DESC
      errCode
      status.
      0: successful.
      -1: failed.
      -2: unsupported format.
      -6: recording has been started. Stop recording first.
      -7: recording file already exists and needs to be deleted.
      -8: recording directory does not have the write permission. Please check the directory permission.
      storagePath
      Storage path of recording file

      onLocalRecording

      onLocalRecording
      void onLocalRecording
      (long duration
      
      const char* storagePath)

      Local media is being recorded

      The SDK returns this callback regularly after local recording is started successfully via the calling of startLocalRecording.
      You can capture this callback to stay up to date with the status of the recording task.
      You can set the callback interval when calling startLocalRecording.
      Param
      DESC
      duration
      Cumulative duration of recording, in milliseconds
      storagePath
      Storage path of recording file

      onLocalRecordFragment

      onLocalRecordFragment
      void onLocalRecordFragment
      (const char* storagePath)

      Record fragment finished.

      When fragment recording is enabled, this callback will be invoked when each fragment file is finished.
      Param
      DESC
      storagePath
      Storage path of the fragment.

      onLocalRecordComplete

      onLocalRecordComplete
      void onLocalRecordComplete
      (int errCode
      
      const char* storagePath)

      Local recording stopped

      When you call stopLocalRecording to stop local recording, the SDK returns this callback to notify you of the recording result.
      Param
      DESC
      errCode
      status
      0: successful.
      -1: failed.
      -2: Switching resolution or horizontal and vertical screen causes the recording to stop.
      -3: recording duration is too short or no video or audio data is received. Check the recording duration or whether audio or video capture is enabled.
      storagePath
      Storage path of recording file

      onSnapshotComplete

      onSnapshotComplete
      void onSnapshotComplete
      (const char* userId
      
      
      char* data
      
      uint32_t length
      
      uint32_t width
      
      uint32_t height
      
      TRTCVideoPixelFormat format)

      Finished taking a local screenshot

      Param
      DESC
      bmp
      Screenshot result. If it is null , the screenshot failed to be taken.
      data
      Screenshot data. If it is nullptr , it indicates that the SDK failed to take the screenshot.
      format
      Screenshot data format. Only TRTCVideoPixelFormat_BGRA32 is supported now.
      height
      Screenshot height
      length
      Screenshot data length. In BGRA32 format, length = width * height * 4.
      type
      Video stream type
      userId
      User ID. If it is empty, the screenshot is a local image.
      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.

      onUserEnter

      onUserEnter
      void onUserEnter
      (const char* userId)

      An anchor entered the room (disused)

      @deprecated This callback is not recommended in the new version. Please use onRemoteUserEnterRoom instead.

      onUserExit

      onUserExit
      void onUserExit
      (const char* userId
      
      int reason)

      An anchor left the room (disused)

      @deprecated This callback is not recommended in the new version. Please use onRemoteUserLeaveRoom instead.

      onAudioEffectFinished

      onAudioEffectFinished
      void onAudioEffectFinished
      (int effectId
      
      int code)

      Audio effects ended (disused)

      @deprecated This callback is not recommended in the new version. Please use ITXAudioEffectManager instead.
      Audio effects and background music can be started using the same API (startPlayMusic) now instead of separate ones.

      onPlayBGMBegin

      onPlayBGMBegin
      void onPlayBGMBegin
      (TXLiteAVError errCode)

      Started playing background music (disused)

      @deprecated This callback is not recommended in the new version. Please use ITXMusicPlayObserver instead.
      Audio effects and background music can be started using the same API (startPlayMusic) now instead of separate ones.

      onPlayBGMProgress

      onPlayBGMProgress
      void onPlayBGMProgress
      (uint32_t progressMS
      
      uint32_t durationMS)

      Playback progress of background music (disused)

      @deprecated This callback is not recommended in the new version. Please use ITXMusicPlayObserver instead.
      Audio effects and background music can be started using the same API (startPlayMusic) now instead of separate ones.

      onPlayBGMComplete

      onPlayBGMComplete
      void onPlayBGMComplete
      (TXLiteAVError errCode)

      Background music stopped (disused)

      @deprecated This callback is not recommended in the new version. Please use ITXMusicPlayObserver instead.
      Audio effects and background music can be started using the same API (startPlayMusic) now instead of separate ones.

      onSpeedTest

      onSpeedTest
      void onSpeedTest
      (const TRTCSpeedTestResult& currentResult
      
      uint32_t finishedCount
      
      uint32_t totalCount)

      Result of server speed testing (disused)

      @deprecated This callback is not recommended in the new version. Please use onSpeedTestResult: instead.

      onRenderVideoFrame

      onRenderVideoFrame
      void onRenderVideoFrame
      (const char* userId
      
      TRTCVideoStreamType streamType
      
      TRTCVideoFrame* frame)

      Custom video rendering

      If you have configured the callback of custom rendering for local or remote video, the SDK will return to you via this callback video frames that are otherwise sent to the rendering control, so that you can customize rendering.
      Param
      DESC
      frame
      Video frames to be rendered
      streamType
      Stream type. The primary stream ( Main ) is usually used for camera images, and the substream ( Sub ) for screen sharing images.
      userId
      userId of the video source. This parameter can be ignored if the callback is for local video ( setLocalVideoRenderDelegate ).

      onGLContextCreated

      onGLContextCreated

      An OpenGL context was created in the SDK.

      onProcessVideoFrame

      onProcessVideoFrame
      int onProcessVideoFrame
      (TRTCVideoFrame *srcFrame
      
      TRTCVideoFrame *dstFrame)

      Video processing by third-party beauty filters

      If you use a third-party beauty filter component, you need to configure this callback in TRTCCloud to have the SDK return to you video frames that are otherwise pre-processed by TRTC.
      You can then send the video frames to the third-party beauty filter component for processing. As the data returned can be read and modified, the result of processing can be synced to TRTC for subsequent encoding and publishing.
      
      Case 1: the beauty filter component generates new textures
      If the beauty filter component you use generates a frame of new texture (for the processed image) during image processing, please set dstFrame.textureId to the ID of the new texture in the callback function.
      int onProcessVideoFrame(TRTCVideoFrame * srcFrame, TRTCVideoFrame *dstFrame) {
          dstFrame->textureId = mFURenderer.onDrawFrameSingleInput(srcFrame->textureId);
          return 0;
      }
      Case 2: you need to provide target textures to the beauty filter component
      If the third-party beauty filter component you use does not generate new textures and you need to manually set an input texture and an output texture for the component, you can consider the following scheme:
      int onProcessVideoFrame(TRTCVideoFrame *srcFrame, TRTCVideoFrame *dstFrame) {
          thirdparty_process(srcFrame->textureId, srcFrame->width, srcFrame->height, dstFrame->textureId);
          return 0;
      }
      Param
      DESC
      dstFrame
      Used to receive video images processed by third-party beauty filters
      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)

      onGLContextDestroy

      onGLContextDestroy

      The OpenGL context in the SDK was destroyed

      onCapturedAudioFrame

      onCapturedAudioFrame
      void onCapturedAudioFrame
      (TRTCAudioFrame *frame)

      Audio data captured by the local mic and pre-processed by the audio module

      After you configure the callback of custom audio processing, the SDK will return via this callback the data captured and pre-processed (ANS, AEC, and AGC) in PCM format.
      The audio returned is in PCM format and has a fixed frame length (time) of 0.02s.
      The formula to convert a frame length in seconds to one in bytes is sample rate * frame length in seconds * number of sound channels * audio bit depth.
      Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be 48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920 bytes.
      Param
      DESC
      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.
      3. The audio data is returned via this callback after ANS, AEC and AGC, but it does not include pre-processing effects like background music, audio effects, or reverb, and therefore has a short delay.

      onLocalProcessedAudioFrame

      onLocalProcessedAudioFrame
      void onLocalProcessedAudioFrame
      (TRTCAudioFrame *frame)

      Audio data captured by the local mic, pre-processed by the audio module, effect-processed and BGM-mixed

      After you configure the callback of custom audio processing, the SDK will return via this callback the data captured, pre-processed (ANS, AEC, and AGC), effect-processed and BGM-mixed in PCM format, before it is submitted to the network module for encoding.
      The audio data returned via this callback is in PCM format and has a fixed frame length (time) of 0.02s.
      The formula to convert a frame length in seconds to one in bytes is sample rate * frame length in seconds * number of sound channels * audio bit depth.
      Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be 48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920 bytes.
      
      Instructions:
      You could write data to the TRTCAudioFrame.extraData filed, in order to achieve the purpose of transmitting signaling.
      Because the data block of the audio frame header cannot be too large, we recommend you limit the size of the signaling data to only a few bytes when using this API. If extra data more than 100 bytes, it won't be sent.
      Other users in the room can receive the message through the TRTCAudioFrame.extraData in onRemoteUserAudioFrame callback in TRTCAudioFrameDelegate.
      Param
      DESC
      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.
      3. Audio data is returned via this callback after ANS, AEC, AGC, effect-processing and BGM-mixing, and therefore the delay is longer than that with onCapturedAudioFrame.

      onPlayAudioFrame

      onPlayAudioFrame
      void onPlayAudioFrame
      (TRTCAudioFrame *frame
      
      const char* userId)

      Audio data of each remote user before audio mixing

      After you configure the callback of custom audio processing, the SDK will return via this callback the raw audio data (PCM format) of each remote user before mixing.
      The audio data returned via this callback is in PCM format and has a fixed frame length (time) of 0.02s.
      The formula to convert a frame length in seconds to one in bytes is sample rate * frame length in seconds * number of sound channels * audio bit depth.
      Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be 48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920 bytes.
      Param
      DESC
      frame
      Audio frames in PCM format
      userId
      User ID
      Note
      The audio data returned via this callback can be read but not modified.

      onMixedPlayAudioFrame

      onMixedPlayAudioFrame
      void onMixedPlayAudioFrame
      (TRTCAudioFrame *frame)

      Data mixed from each channel before being submitted to the system for playback

      After you configure the callback of custom audio processing, the SDK will return to you via this callback the data (PCM format) mixed from each channel before it is submitted to the system for playback.
      The audio data returned via this callback is in PCM format and has a fixed frame length (time) of 0.02s.
      The formula to convert a frame length in seconds to one in bytes is sample rate * frame length in seconds * number of sound channels * audio bit depth.
      Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be 48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920 bytes.
      Param
      DESC
      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.
      3. The audio data returned via this callback is the audio data mixed from each channel before it is played. It does not include the in-ear monitoring data.

      onMixedAllAudioFrame

      onMixedAllAudioFrame
      void onMixedAllAudioFrame
      (TRTCAudioFrame *frame)

      Data mixed from all the captured and to-be-played audio in the SDK

      After you configure the callback of custom audio processing, the SDK will return via this callback the data (PCM format) mixed from all captured and to-be-played audio in the SDK, so that you can customize recording.
      The audio data returned via this callback is in PCM format and has a fixed frame length (time) of 0.02s.
      The formula to convert a frame length in seconds to one in bytes is sample rate * frame length in seconds * number of sound channels * audio bit depth.
      Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be 48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920 bytes.
      Param
      DESC
      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.

      onLog

      onLog
      void onLog
      (const char* log
      
      TRTCLogLevel level
      
      const char* module)

      Printing of local log

      If you want to capture the local log printing event, you can configure the log callback to have the SDK return to you via this callback all logs that are to be printed.
      Param
      DESC
      level
      Log level. For more information, please see TRTC_LOG_LEVEL .
      log
      Log content
      module
      Reserved field, which is not defined at the moment and has a fixed value of TXLiteAVSDK .
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