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Learn more 
Only  $9.9! Get 50,000 minutes with our Starter Plan, perfect for your MVP project.
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RTC Engine
Overview
  • Web
    • Demo Quick Run
    • SDK Quick Start
    • Basic Features
      • Screen Sharing
      • Live Streaming
      • Media Device
      • Audio Volume
      • Set Encoding Profile
      • Detect Network Quality
      • Detect Capabilities
    • Advance Features
      • Enable AI Denoiser
      • Enable Audio Mixer
      • Enable Watermark
      • SEI Message
      • Custom Capturing and Rendering
      • Web Beautification Effects
    • Best Practices
      • Optimize Multi-Person Video Calls
      • Handle Autoplay Restriction
      • Handle Firewall Restriction
    • API List
    • Released Notes
    • Supported Platforms
    • Web FAQs
  • Android
    • Integration
      • 1.API Examples
      • 2.Importing the SDK
      • 3.Entering a Room
      • 4.Subscribing to Audio/Video Streams
      • 5.Publish Audio/Video Streams
      • 6.Exiting a Room
      • 7.Sensing Network Quality
      • 8.Enabling Screen Sharing
      • 9.Setting Video Quality
      • 10.Rotating Videos
    • Testing Newwork Quality
    • Custom Capturing and Rendering
    • Custom Audio Capturing and Playback
    • Client APIs
      • Overview
      • TRTCCloud
      • TRTCStatistics
      • TRTCCloudListener
      • TXAudioEffectManager
      • TXBeautyManager
      • TXDeviceManager
      • Type Definition
      • Deprecated Interface
      • Error Codes
    • Release Notes
  • iOS
    • Integration
      • 1.API Examples
      • 2.Importing the SDK
      • 3.Entering a Room
      • 4.Subscribing to Audio/Video Streams
      • 5.Publish Audio/Video Streams
      • 6.Exiting a Room
      • 7.Sensing Network Quality
      • 8.Enabling Screen Sharing
      • 9.Setting Video Quality
      • 10.Rotating Videos
    • Testing Network Quality
    • Custom Capturing and Rendering
    • Custom Audio Capturing and Playback
    • Client APIs
      • Overview
      • TRTCCloud
      • TRTCCloudDelegate
      • TRTCStatistics
      • TXAudioEffectManager
      • TXBeautyManager
      • TXDeviceManager
      • Type Definition
      • Deprecated Interface
      • ErrorCode
    • Release Notes
  • macOS
    • Integration
      • 1.API Examples
      • 2.Importing the SDK
      • 3.Entering a Room
      • 4.Subscribing to Audio/Video Streams
      • 5.Publish Audio/Video Streams
      • 6.Exiting a Room
      • 7.Sensing Network Quality
      • 8.Enabling Screen Sharing
      • 9.Sharing Computer Audio
      • 10.Setting Video Quality
      • 11.Rotating Videos
    • Testing Hardware Devices
    • Testing Network Quality
    • Custom Capturing and Rendering
    • Custom Audio Capturing and Playback
    • Client APIs
      • Overview
      • TRTCCloud
      • TRTCCloudDelegate
      • TRTCStatistics
      • TXAudioEffectManager
      • TXBeautyManager
      • TXDeviceManager
      • Type Definition
      • Deprecated Interface
      • ErrorCode
      • Release Notes
    • Release Notes
  • Windows C++
    • Integration
      • 1.API Examples
      • 2.Importing the SDK
      • 3.Entering a Room
      • 4.Subscribing to Audio/Video Streams
      • 5.Publish Audio/Video Streams
      • 6.Exiting a Room
      • 7.Sensing Network Quality
      • 8.Enabling Screen Sharing
      • 9.Setting Video Quality
      • 10.Rotating Videos
    • Testing Hardware Devices
    • Testing Network Quality
    • Custom Capturing and Rendering
    • Custom Audio Capturing and Playback
    • Client APIs
      • Overview
      • ITRTCCloud
      • ITRTCStatistics
      • TRTCCloudCallback
      • ITXAudioEffectManager
      • ITXDeviceManager
      • Type Definition
      • Deprecated Interface
      • Error Codes
    • Release Notes
  • Electron
    • Integration
      • 1.API Examples
      • 2.Importing the SDK
      • 3.Entering a Room
      • 4.Subscribing to Audio/Video Streams
      • 5.Publish Audio/Video Streams
      • 6.Exiting a Room
      • 7.Sensing Network Quality
      • 8.Enabling Screen Sharing
      • 9.Sharing Computer Audio
      • 10.Setting Video Quality
      • 11.Rotating Videos
    • Client APIs
      • Overview
      • Error Codes
  • Flutter
    • Integration
      • 1.API Examples
      • 2.Importing the SDK
      • 3.Entering a Room
      • 4.Subscribing to Audio/Video Streams
      • 5.Publish Audio/Video Streams
      • 6.Exiting a Room
      • 7.Sensing Network Quality
      • 8.Enabling Screen Sharing
      • 9.Sharing Computer Audio
      • 10.Setting Video Quality
      • 11.Rotating Videos
    • Client APIs
      • Overview
      • Error Codes
  • Unity
    • Integration
      • 1.API Examples
      • 2Importing the SDK
    • Client APIs
      • Overview
      • Error Codes
  • Qt
    • Integration
      • 1.Importing the SDK
  • Overview
    • Overview
  • Pricing
    • RTC-Engine Packages
    • Billing of On-Cloud Recording
    • Billing of MixTranscoding and Relay to CDN
    • Billing Explanation for Subscription Package Duration
    • Billing of Monitoring Dashboard
    • Free Minutes
    • Pay-As-You-Go
  • Concepts
  • Features
  • Performance Statistics
  • FAQs
    • FAQs for Beginners
    • Migration Guide
      • Twilio Video to Tencent RTC
      • Billing
      • Features
      • UserSig
      • Firewall Restrictions
      • How to Downsize Installation Package
      • TRTCCalling for Web
      • Audio and Video Quality
      • Others
RTC Engine

TRTCCloudDelegate

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 macOS 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
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 macOS only)

On macOS, you can call startSystemAudioLoopback to install an audio driver and 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 ).

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 .