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Tencent Real-Time Communication
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    TRTCCloudDelegate

    Last updated: 2024-11-28 17:28:45
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      Copyright (c) 2021 Tencent. All rights reserved.
      
      Module: TRTCCloudDelegate @ TXLiteAVSDK
      
      Function: event callback APIs for TRTC’s video call feature
      
      TRTCCloudDelegate

      TRTCCloudDelegate

      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      A remote user published/unpublished primary stream video
      A remote user published/unpublished substream video
      A remote user published/unpublished audio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
      The audio route changed (for mobile devices only)
      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)
      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
      Local recording started
      Local media is being recorded
      Record fragment finished.
      Local recording stopped
      An anchor entered the room (disused)
      An anchor left the room (disused)
      Audio effects ended (disused)

      TRTCVideoRenderDelegate

      FuncList
      DESC
      Custom video rendering

      TRTCVideoFrameDelegate

      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

      TRTCAudioFrameDelegate

      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
      In-ear monitoring data

      TRTCLogDelegate

      FuncList
      DESC
      Printing of local log

      onError:errMsg:extInfo:

      onError:errMsg:extInfo:
      - (void)onError:
      (TXLiteAVError)errCode
      errMsg:
      (nullable NSString *)errMsg
      extInfo:
      (nullable NSDictionary*)extInfo

      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:warningMsg:extInfo:

      onWarning:warningMsg:extInfo:
      - (void)onWarning:
      (TXLiteAVWarning)warningCode
      warningMsg:
      (nullable NSString *)warningMsg
      extInfo:
      (nullable NSDictionary*)extInfo

      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:
      (NSInteger)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:
      (NSInteger)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:errMsg:

      onSwitchRole:errMsg:
      - (void)onSwitchRole:
      (TXLiteAVError)errCode
      errMsg:
      (nullable NSString *)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:errMsg:

      onSwitchRoom:errMsg:
      - (void)onSwitchRoom:
      (TXLiteAVError)errCode
      errMsg:
      (nullable NSString *)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:errCode:errMsg:

      onConnectOtherRoom:errCode:errMsg:
      - (void)onConnectOtherRoom:
      (NSString*)userId
      errCode:
      (TXLiteAVError)errCode
      errMsg:
      (nullable NSString *)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:errMsg:

      onDisconnectOtherRoom:errMsg:
      - (void)onDisconnectOtherRoom:
      (TXLiteAVError)errCode
      errMsg:
      (nullable NSString *)errMsg

      Result of ending cross-room call

      onUpdateOtherRoomForwardMode:errMsg:

      onUpdateOtherRoomForwardMode:errMsg:
      - (void)onUpdateOtherRoomForwardMode:
      (TXLiteAVError)errCode
      errMsg:
      (nullable NSString *)errMsg

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

      onRemoteUserEnterRoom:

      onRemoteUserEnterRoom:
      - (void)onRemoteUserEnterRoom:
      (NSString *)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:reason:

      onRemoteUserLeaveRoom:reason:
      - (void)onRemoteUserLeaveRoom:
      (NSString *)userId
      reason:
      (NSInteger)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:available:

      onUserVideoAvailable:available:
      - (void)onUserVideoAvailable:
      (NSString *)userId
      available:
      (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, YES) 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, NO) 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. YES : published; NO : unpublished
      userId
      User ID of the remote user

      onUserSubStreamAvailable:available:

      onUserSubStreamAvailable:available:
      - (void)onUserSubStreamAvailable:
      (NSString *)userId
      available:
      (BOOL)available

      A remote user published/unpublished substream video

      The substream is usually used for screen sharing images. If you receive the onUserSubStreamAvailable(userId, YES) 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. YES : published; NO : unpublished
      userId
      User ID of the remote user
      Note
      The API used to display substream images is startRemoteView, not startRemoteSubStreamView, startRemoteSubStreamView is deprecated.

      onUserAudioAvailable:available:

      onUserAudioAvailable:available:
      - (void)onUserAudioAvailable:
      (NSString *)userId
      available:
      (BOOL)available

      A remote user published/unpublished audio

      If you receive the onUserAudioAvailable(userId, YES) 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, NO) to play the user’s audio.
      Param
      DESC
      available
      Whether the user published (or unpublished) audio. YES : published; NO : 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:streamType:width:height:

      onFirstVideoFrame:streamType:width:height:
      - (void)onFirstVideoFrame:
      (NSString*)userId
      streamType:
      (TRTCVideoStreamType)streamType
      width:
      (int)width
      height:
      (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:
      (NSString*)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:
      (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:streamType:streamStatus:reason:extrainfo:

      onRemoteVideoStatusUpdated:streamType:streamStatus:reason:extrainfo:
      - (void)onRemoteVideoStatusUpdated:
      (NSString *)userId
      streamType:
      (TRTCVideoStreamType)streamType
      streamStatus:
      (TRTCAVStatusType)status
      reason:
      extrainfo:
      (nullable NSDictionary *)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:streamStatus:reason:extrainfo:

      onRemoteAudioStatusUpdated:streamStatus:reason:extrainfo:
      - (void)onRemoteAudioStatusUpdated:
      (NSString *)userId
      streamStatus:
      (TRTCAVStatusType)status
      reason:
      extrainfo:
      (nullable NSDictionary *)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:streamType:newWidth:newHeight:

      onUserVideoSizeChanged:streamType:newWidth:newHeight:
      - (void)onUserVideoSizeChanged:
      (NSString *)userId
      streamType:
      (TRTCVideoStreamType)streamType
      newWidth:
      (int)newWidth
      newHeight:
      (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:remoteQuality:

      onNetworkQuality:remoteQuality:
      - (void)onNetworkQuality:
      (TRTCQualityInfo*)localQuality
      remoteQuality:
      (NSArray<TRTCQualityInfo*>*)remoteQuality

      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:
      (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:
      (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.

      onAudioRouteChanged:fromRoute:

      onAudioRouteChanged:fromRoute:
      - (void)onAudioRouteChanged:
      (TRTCAudioRoute)route
      fromRoute:
      (TRTCAudioRoute)fromRoute

      The audio route changed (for mobile devices only)

      Audio route is the route (speaker or receiver) through which audio is played.
      When audio is played through the receiver, the volume is relatively low, and the sound can be heard only when the phone is put near the ear. This mode has a high level of privacy and is suitable for answering calls.
      When audio is played through the speaker, the volume is relatively high, and there is no need to put the phone near the ear. This mode enables the "hands-free" feature.
      When audio is played through the wired earphone.
      When audio is played through the bluetooth earphone.
      When audio is played through the USB sound card.
      Param
      DESC
      fromRoute
      The audio route used before the change
      route
      Audio route, i.e., the route (speaker or receiver) through which audio is played

      onUserVoiceVolume:totalVolume:

      onUserVoiceVolume:totalVolume:
      - (void)onUserVoiceVolume:
      (NSArray<TRTCVolumeInfo *> *)userVolumes
      totalVolume:
      (NSInteger)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.

      onDevice:type:stateChanged:

      onDevice:type:stateChanged:
      - (void)onDevice:
      (NSString *)deviceId
      type:
      (TRTCMediaDeviceType)deviceType
      stateChanged:
      (NSInteger)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
      state
      Device status. 0 : disconnected; 1 : connected

      onAudioDeviceCaptureVolumeChanged:muted:

      onAudioDeviceCaptureVolumeChanged:muted:
      - (void)onAudioDeviceCaptureVolumeChanged:
      (NSInteger)volume
      muted:
      (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. YES : muted; NO : 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:muted:

      onAudioDevicePlayoutVolumeChanged:muted:
      - (void)onAudioDevicePlayoutVolumeChanged:
      (NSInteger)volume
      muted:
      (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. YES : muted; NO : 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)err

      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.

      onRecvCustomCmdMsgUserId:cmdID:seq:message:

      onRecvCustomCmdMsgUserId:cmdID:seq:message:
      - (void)onRecvCustomCmdMsgUserId:
      (NSString *)userId
      cmdID:
      (NSInteger)cmdID
      seq:
      (UInt32)seq
      message:
      (NSData *)message

      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

      onMissCustomCmdMsgUserId:cmdID:errCode:missed:

      onMissCustomCmdMsgUserId:cmdID:errCode:missed:
      - (void)onMissCustomCmdMsgUserId:
      (NSString *)userId
      cmdID:
      (NSInteger)cmdID
      errCode:
      (NSInteger)errCode
      missed:
      (NSInteger)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 YES ), 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 YES , 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 YES .

      onRecvSEIMsg:message:

      onRecvSEIMsg:message:
      - (void)onRecvSEIMsg:
      (NSString *)userId
      message:
      (NSData*)message

      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:errMsg:

      onStartPublishing:errMsg:
      - (void)onStartPublishing:
      (int)err
      errMsg:
      (NSString*)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:errMsg:

      onStopPublishing:errMsg:
      - (void)onStopPublishing:
      (int)err
      errMsg:
      (NSString*)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:errMsg:

      onStartPublishCDNStream:errMsg:
      - (void)onStartPublishCDNStream:
      (int)err
      errMsg:
      (NSString *)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:errMsg:

      onStopPublishCDNStream:errMsg:
      - (void)onStopPublishCDNStream:
      (int)err
      errMsg:
      (NSString *)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:errMsg:

      onSetMixTranscodingConfig:errMsg:
      - (void)onSetMixTranscodingConfig:
      (int)err
      errMsg:
      (NSString*)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:code:message:extraInfo:

      onStartPublishMediaStream:code:message:extraInfo:
      - (void)onStartPublishMediaStream:
      (NSString*)taskId
      code:
      (int)code
      message:
      (NSString*)message
      extraInfo:
      (nullable NSDictionary *)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:code:message:extraInfo:

      onUpdatePublishMediaStream:code:message:extraInfo:
      - (void)onUpdatePublishMediaStream:
      (NSString*)taskId
      code:
      (int)code
      message:
      (NSString*)message
      extraInfo:
      (nullable NSDictionary *)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:code:message:extraInfo:

      onStopPublishMediaStream:code:message:extraInfo:
      - (void)onStopPublishMediaStream:
      (NSString*)taskId
      code:
      (int)code
      message:
      (NSString*)message
      extraInfo:
      (nullable NSDictionary *)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:status:code:msg:extraInfo:

      onCdnStreamStateChanged:status:code:msg:extraInfo:
      - (void)onCdnStreamStateChanged:
      (NSString*)cdnUrl
      status:
      (int)status
      code:
      (int)code
      msg:
      (NSString*)msg
      extraInfo:
      (nullable NSDictionary *)info

      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.

      onLocalRecordBegin:storagePath:

      onLocalRecordBegin:storagePath:
      - (void)onLocalRecordBegin:
      (NSInteger)errCode
      storagePath:
      (NSString *)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:storagePath:

      onLocalRecording:storagePath:
      - (void)onLocalRecording:
      (NSInteger)duration
      storagePath:
      (NSString *)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:
      (NSString *)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:storagePath:

      onLocalRecordComplete:storagePath:
      - (void)onLocalRecordComplete:
      (NSInteger)errCode
      storagePath:
      (NSString *)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

      onUserEnter:

      onUserEnter:
      - (void)onUserEnter:
      (NSString *)userId

      An anchor entered the room (disused)

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

      onUserExit:reason:

      onUserExit:reason:
      - (void)onUserExit:
      (NSString *)userId
      reason:
      (NSInteger)reason

      An anchor left the room (disused)

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

      onAudioEffectFinished:code:

      onAudioEffectFinished:code:
      - (void)onAudioEffectFinished:
      (int) effectId
      code:
      (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.

      onRenderVideoFrame:userId:streamType:

      onRenderVideoFrame:userId:streamType:
      - (void) onRenderVideoFrame:
      (TRTCVideoFrame * _Nonnull)frame
      userId:
      (NSString* __nullable)userId
      streamType:
      (TRTCVideoStreamType)streamType

      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:dstFrame:

      onProcessVideoFrame:dstFrame:
      - (uint32_t)onProcessVideoFrame:
      (TRTCVideoFrame * _Nonnull)srcFrame
      dstFrame:
      (TRTCVideoFrame * _Nonnull)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.
      uint32_t onProcessVideoFrame(TRTCVideoFrame * _Nonnull)srcFrame dstFrame:(TRTCVideoFrame * _Nonnull)dstFrame{
          self.frameID += 1;
          dstFrame.pixelBuffer = [[FURenderer shareRenderer] renderPixelBuffer:srcFrame.pixelBuffer
                                                                   withFrameId:self.frameID
                                                                         items:self.renderItems
                                                                     itemCount:self.renderItems.count];
          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:
      uint32_t onProcessVideoFrame(TRTCVideoFrame * _Nonnull)srcFrame dstFrame:(TRTCVideoFrame * _Nonnull)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)

      onGLContextDestory

      onGLContextDestory

      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.

      onRemoteUserAudioFrame:userId:

      onRemoteUserAudioFrame:userId:
      - (void) onRemoteUserAudioFrame:
      (TRTCAudioFrame *)frame
      userId:
      (NSString *)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.

      onVoiceEarMonitorAudioFrame:

      onVoiceEarMonitorAudioFrame:
      - (void) onVoiceEarMonitorAudioFrame:
      (TRTCAudioFrame *)frame

      In-ear monitoring data

      After you configure the callback of custom audio processing, the SDK will return to you via this callback the in-ear monitoring data (PCM format) before it is submitted to the system for playback.
      The audio returned is in PCM format and has a not-fixed frame length (time).
      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 length of 0.02s frame 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, 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.

      onLog:LogLevel:WhichModule:

      onLog:LogLevel:WhichModule:
      -(void) onLog:
      (nullable NSString*)log
      LogLevel:
      (TRTCLogLevel)level
      WhichModule:
      (nullable NSString*)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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