Class ITRTCCloud

Module: ITRTCCloud @ TXLiteAVSDK

SDK VERSION 6.0

Function: TRTC main API classes

Nouns[1]: primary stream - The channel for camera video is the primary stream.

Nouns[2]: substream - The channel for screen sharing or VOD play is the substream.

Nouns[3]: vodplay - TRTC for Windows allows the streaming of a local video file. This feature is known as VOD play.

Inheritance

System.Object

ITRTCCloud

Inherited Members

System.Object.Equals(System.Object)

System.Object.Equals(System.Object, System.Object)

System.Object.GetHashCode()

System.Object.GetType()

System.Object.MemberwiseClone()

System.Object.ToString()

System.Object.ReferenceEquals(System.Object, System.Object)

Namespace: trtc

Assembly: cs.temp.dll.dll

Syntax

public abstract class ITRTCCloud

Examples

Sample code for creating/using/terminating an ITRTCCloud object:

ITRTCCloud trtcCloud = ITRTCCloud.getTRTCShareInstance();
if (trtcCloud != null)
{
string version = trtcCloud->getSDKVersion();
}

Release an ITRTCCloud singleton object after the process is closed or if the object is no longer needed

ITRTCCloud.destroyTRTCShareInstance();
trtcCloud = NULL;

Methods

addCallback(ITRTCCloudCallback)

Set ITRTCCloudCallback

You can use `ITRTCCloudCallback` to receive different status notifications from the SDK. For details, please see the definitions in `TRTCCloudCallback.cs`.

Declaration

public abstract void addCallback(ITRTCCloudCallback callback)

Parameters

Type	Name	Description
ITRTCCloudCallback	callback	Event callback

callExperimentalAPI(String)

Call experimental APIs

Declaration

public abstract void callExperimentalAPI(string jsonStr)

Parameters

Type	Name	Description
System.String	jsonStr	JSON-string API name and parameter description

Remarks

This API is used to call experimental APIs.

connectOtherRoom(String)

Request a cross-room call (anchor competition)

The cross-room call feature allows two anchors from different rooms to call and compete with each other without having to exit their own rooms.

Declaration

public abstract void connectOtherRoom(string jsonParams)

Parameters

Type	Name	Description
System.String	jsonParams	JSON-string parameters, in which roomId is the room ID of the anchor to call and userId the user ID of the anchor to call

Remarks

The JsonCpp library is used to format JSON strings. JsonData jsonObj =new JsonData(); jsonObj["roomId"] = 1908; jsonObj["userId"] = "345"; string jsonData = JsonMapper.ToJson(jsonObj); mTRTCCloud.connectOtherRoom(jsonData);

Examples

    For example, after anchor A in room 001 uses `connectOtherRoom()` to successfully call anchor B in room 002, all users in room 001 will receive the `onUserEnter(B)` and `onUserVideoAvailable(B,true)` callbacks, and all users in room 002 will receive the `onUserEnter(A)` and `onUserVideoAvailable(A,true)` callbacks.
    In essence, a cross-room call is sharing the audio and video of two anchors in different rooms to each other so that the audiences in both rooms can see two anchors.
                        Room 001                     Room 002
              -------------               ------------
   Before cross-room call:| Anchor A      |             | Anchor B     |
             | Audience U, V, and W  |             | Audience X, Y, and Z |
              -------------               ------------
                Room 001                     Room 002
              -------------               ------------
   After cross-room call: | Anchors A and B    |             | Anchors B and A   |
             | Audience U, V, and W  |             | Audience X, Y, and Z |
              -------------               ------------
To ensure the compatibility of extended parameters for the cross-room call API, the JSON format is used for the parameters, which must contain at least two fields:
`roomId`: If anchor A in room 001 wants to call anchor B in room 002, he or she must set `roomId` to `002` when calling `connectOtherRoom()`.
`userId`: If anchor A in room 001 wants to call anchor B in room 002, he or she must set `userId` to the user ID of anchor B when calling `connectOtherRoom()`.
The result is returned through the `onConnectOtherRoom()` callback in `ITRTCCloudCallback`.

destroyTRTCShareInstance()

Release an ITRTCCloud singleton object

Declaration

public static void destroyTRTCShareInstance()

disconnectOtherRoom()

End a cross-room call

The result is returned through the `onDisconnectOtherRoom()` callback in `ITRTCCloudCallback`.

Declaration

public abstract void disconnectOtherRoom()

enableAudioVolumeEvaluation(UInt32)

Enable/Disable the volume reminder

After this feature is enabled, the SDK will return its evaluation of the volume of each stream via `onUserVoiceVolume()`.

The volume bar in our demo is implemented using this API.

To enable the volume reminder, please call this API before `startLocalAudio()`.

Declaration

public abstract void enableAudioVolumeEvaluation(uint interval)

Parameters

Type	Name	Description
System.UInt32	interval	Set the interval (ms) for triggering the onUserVoiceVolume callback. The minimum interval allowed is 100 ms. 300 ms is recommended. If the value is 0 or smaller, the callback will be disabled.

enableCustomAudioCapture(Boolean)

Enable/Disable custom audio capturing (not supported on Android)

After custom audio capturing is enabled, the SDK will skip audio capturing but will continue to encode and send audio data.

You need to keep feeding custom audio data to the SDK using `sendCustomAudioData()`.

Declaration

public abstract void enableCustomAudioCapture(bool enable)

Parameters

Type	Name	Description
System.Boolean	enable	Whether to enable custom audio capturing. Default value: false

enableCustomVideoCapture(Boolean)

Enable/Disable custom video capturing

After custom video capturing is enabled, the SDK will skip video capturing but will continue to encode and send video data.

You need to keep feeding custom video data to the SDK using `sendCustomVideoData()`.

Declaration

public abstract void enableCustomVideoCapture(bool enable)

Parameters

Type	Name	Description
System.Boolean	enable	Whether to enable custom video capturing. Default value: false

enableSmallVideoStream(Boolean, ref TRTCVideoEncParam)

Enable/Disable dual-channel (big and small images) encoding

Declaration

public abstract void enableSmallVideoStream(bool enable, ref TRTCVideoEncParam smallVideoParam)

Parameters

Type	Name	Description
System.Boolean	enable	Whether to enable dual-channel encoding. Default value: false
TRTCVideoEncParam	smallVideoParam	Parameters of the small image

Remarks

You can enable the dual-channel encoding mode for users in the major role (such as anchor, teacher, or host) and using PC or Mac. In this mode, a user will output one audio stream and two video streams: an **HD** stream and a **smooth** stream. This mode consumes more bandwidth and CPU computing resources.
For the audience in the room:
    Select the **HD** stream to watch if the user’s downstream network conditions are good.
    Select the **smooth** stream to watch if the user’s downstream network conditions are poor.

enterRoom(ref TRTCParams, TRTCAppScene)

1.1 Enter a room

You will receive the `onEnterRoom(result)` callback in `ITRTCCloudCallback`:

- If room entry succeeds, `result` will be a positive integer (`result` > 0), which indicates the time (ms) it takes to enter the room.

- If room entry fails, `result` will be a negative integer (`result` < 0), which represents the error code.

For more information on room entry error codes, please see [Error Codes](https://cloud.tencent.com/document/product/647/32257).

TRTCAppSceneVideoCall:

Video calls support 720p and 1080p HD video. Each room allows up to 300 concurrent users, and up to 50 of them can speak simultaneously.

Use cases: [one-to-one video call], [video conferencing with up to 300 participants], [online medical consultation], [video chat], [video interview], etc.

TRTCAppSceneAudioCall:

Audio calls support 48 kHz stereo audio. Each room allows up to 300 concurrent users, and up to 50 of them can speak simultaneously.

Use cases: [one-to-one audio call], [audio conferencing with up to 300 participants], [online Werewolf playing], [audio chat room], etc.

TRTCAppSceneLIVE:

Interactive video streaming allows smooth mic on/off without waiting. Anchor latency is lower than 300 ms and playback latency lower than 1,000 ms. Up to 100,000 users can play the anchor’s video at the same time.

Use cases: [interactive classroom], [interactive live streaming], [video dating], [remote training], [large-scale conferencing], etc.

TRTCAppSceneVoiceChatRoom:

Interactive audio streaming allows smooth mic on/off without waiting. Anchor latency is lower than 300 ms and playback latency lower than 1,000 ms. Up to 100,000 users can play the anchor’s audio at the same time.

Use cases: [audio chat room], [karaoke], [FM radio], etc.

Declaration

public abstract void enterRoom(ref TRTCParams param, TRTCAppScene scene)

Parameters

Type	Name	Description
TRTCParams	param
TRTCAppScene	scene	Application scenario. TRTC supports four scenarios currently: video call (VideoCall), live streaming (Live), audio call (AudioCall), and audio chat room (VoiceChatRoom).

Remarks

1. If scene is set to TRTCAppSceneLIVE or TRTCAppSceneVoiceChatRoom, you must select a role for the current user by specifying the role field in TRTCParams.
2. After you call enterRoom, regardless of whether room entry succeeds, you must call exitRoom before calling enterRoom again; otherwise, an unexpected error will occur.

exitRoom()

1.2 Leave a room

Calling `exitRoom()` will trigger the execution of room exit-related logic, including releasing resources such as audio/video devices and codecs.

After all the resources are released, the SDK will send you the `onExitRoom()` callback in `ITRTCCloudCallback`.

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.

Declaration

public abstract void exitRoom()

getAudioCaptureVolume()

Get the capturing volume of the SDK

Declaration

public abstract int getAudioCaptureVolume()

Returns

Type	Description
System.Int32

getAudioEffectManager()

Get the audio effect management class (supported on Android, iOS, and macOS)

Declaration

public abstract ITXAudioEffectManager getAudioEffectManager()

Returns

Type	Description
ITXAudioEffectManager	Audio effect management class

Remarks

This is the SDK’s audio effect management module. It supports the following features: In-ear monitoring: play the audio captured by the mic through headphones in real time Reverb effect: add reverb effects such as karaoke, room, hall, deep, and resonant Voice changing effect: add voice changing effects such as little girl, middle-aged man, metal, and punk Background music: play online or local music (support speed and pitch adjustment, playback with and without vocals, as well as looping) Short audio effect: add short audio effects such as applause and laughter. For files shorter than 10 seconds, set the isShortFile parameter to YES.

getAudioPlayoutVolume()

Get the playback volume of the SDK

Declaration

public abstract int getAudioPlayoutVolume()

Returns

Type	Description
System.Int32

getDeviceManager()

Get the device management module (supported on Android and iOS)

Declaration

public abstract ITXDeviceManager getDeviceManager()

Returns

Type	Description
ITXDeviceManager	Device management class

getScreenCaptureSources()

9.5 Enumerate shareable sources (preferably called before startScreenCapture)

You can use this API to get the ID, type, and name of sharable sources.

Declaration

public abstract TRTCScreenCaptureSourceInfo[] getScreenCaptureSources()

Returns

Type	Description
TRTCScreenCaptureSourceInfo[]	Only screen-type sources are returned currently.

Remarks

Information of shareable sources

getSDKVersion()

Get the SDK version

Declaration

public abstract string getSDKVersion()

Returns

Type	Description
System.String	UTF-8-encoded version number

getTRTCShareInstance()

Get an ITRTCCloud singleton object

Declaration

public static ITRTCCloud getTRTCShareInstance()

Returns

Type	Description
ITRTCCloud	Return an ITRTCCloud singleton object

GetVideoRenderData(String, ref Int32, ref Int32, ref Int32, ref Int32, Boolean)

Declaration

public abstract IntPtr GetVideoRenderData(string userId, ref int rotation, ref int width, ref int height, ref int length, bool isNeedDestroy)

Parameters

Type	Name	Description
System.String	userId
System.Int32	rotation
System.Int32	width
System.Int32	height
System.Int32	length
System.Boolean	isNeedDestroy

Returns

Type	Description
IntPtr

muteAllRemoteAudio(Boolean)

4.6 Mute/Unmute all users' audio

Declaration

public abstract void muteAllRemoteAudio(bool mute)

Parameters

Type	Name	Description
System.Boolean	mute	true: mute; false: unmute

Remarks

If mute is true, the SDK will stop receiving and playing all remote users’ audio; if mute is false, it will start receiving and playing all remote users’ audio.

muteAllRemoteVideoStreams(Boolean)

Pause/Resume receiving the videos of all remote users

Declaration

public abstract void muteAllRemoteVideoStreams(bool mute)

Parameters

Type	Name	Description
System.Boolean	mute	Whether to pause receiving video

muteLocalAudio(Boolean)

4.4 Mute/Unmute local audio

After local audio is muted, other users in the room will receive the `onUserAudioAvailable(userId, false)` callback.

After local audio is unmuted, other users in the room will receive the `onUserAudioAvailable(userId, true)` callback.

Different from `stopLocalAudio`, `muteLocalAudio(true)` does not stop the sending of audio or video data. Data packets continue to be sent, although without audio and at extremely low bitrate.

As MP4 and other video formats have high requirements on audio continuity, `stopLocalAudio` may make an MP4 recording file fail to be played smoothly.

`muteLocalAudio` has less impact on the compatibility of MP4 recording files. Therefore, you are advised to use `muteLocalAudio` in scenarios with high requirements on recording quality.

Declaration

public abstract void muteLocalAudio(bool mute)

Parameters

Type	Name	Description
System.Boolean	mute	true: mute; false (default): unmute

muteLocalVideo(Boolean)

3.3 Pause/Resume publishing local video

After a user pauses publishing local video, other users in the room will receive the `onUserVideoAvailable(userId, false)` callback.

After a user resumes publishing local video, other users in the room will receive the `onUserVideoAvailable(userId, true)` callback.

Declaration

public abstract void muteLocalVideo(bool mute)

Parameters

Type	Name	Description
System.Boolean	mute	true: pause; false (default): resume

muteRemoteAudio(String, Boolean)

4.5 Mute/Unmute a remote user's audio

Declaration

public abstract void muteRemoteAudio(string userId, bool mute)

Parameters

Type	Name	Description
System.String	userId	ID of the remote user
System.Boolean	mute	true: mute; false: unmute

Remarks

If mute is true, the SDK will stop receiving and playing the remote user’s audio; if mute is false, it will start receiving and playing the remote user’s audio.

muteRemoteVideoStream(String, Boolean)

Pause/Resume receiving the video of a remote user

This API will pause/resume receiving the video of the specified remote user, but will not release the resources used to display the video. After pause, the last frame of video before the pause will be displayed.

Declaration

public abstract void muteRemoteVideoStream(string userId, bool mute)

Parameters

Type	Name	Description
System.String	userId	User ID of the remote user
System.Boolean	mute	Whether to pause receiving video

pauseScreenCapture()

9.3 Pause screen sharing

Declaration

public abstract void pauseScreenCapture()

removeCallback(ITRTCCloudCallback)

Remove event callbacks

Declaration

public abstract void removeCallback(ITRTCCloudCallback callback)

Parameters

Type	Name	Description
ITRTCCloudCallback	callback	Event callback

resumeScreenCapture()

9.4 Resume screen sharing

Declaration

public abstract void resumeScreenCapture()

selectScreenCaptureTarget(TRTCScreenCaptureSourceInfo, Rect, TRTCScreenCaptureProperty)

9.6 Set screen sharing parameters (can be called during screen sharing)

During screen sharing, you can call this API to switch the screen to share. You don’t need to start screen sharing again.

You can:

-Share an entire screen: select a source whose `type` is `Screen` from `sourceInfoList` and set `captureRect` to `{0, 0, 0, 0}`

-Share a portion of a screen: select a source whose `type` is `Screen` from `sourceInfoList` and set `captureRect` to a non-null value, e.g., {100, 100, 300, 300}

Declaration

public abstract void selectScreenCaptureTarget(TRTCScreenCaptureSourceInfo source, Rect captureRect, TRTCScreenCaptureProperty property)

Parameters

Type	Name	Description
TRTCScreenCaptureSourceInfo	source	The source to share
Rect	captureRect	The portion of the screen to share
TRTCScreenCaptureProperty	property	Screen sharing properties, such as whether to enable mouse cursor capturing or show a bright border around the shared content. For details, please see the definition of TRTCScreenCaptureProperty.

sendCustomAudioData(TRTCAudioFrame)

Send custom audio data to the SDK (not supported on Android)

We recommend the following settings for the parameters of `TRTCAudioFrame` (other parameters can be left empty).

-audioFormat: only `LiteAVAudioFrameFormatPCM` is supported.

-data: audio frame buffer

-length: audio frame size. We recommend a frame duration of 20 ms. **Assume that the PCM format is used, the sample rate is 48000 Hz, and the audio mono. The size of an audio frame would be 48000 × 0.02s × 1 × 16 bits = 15360 bits = 1920 bytes.

-sampleRate: sample rate, which can only be `48000`

-channel: number of sound channels. Valid values: `1`: mono-channel; `2`: dual-channel. If dual channels are used, data will be interleaved.

-timestamp: Separate timestamps equally; otherwise, an audio-to-video sync error will occur and the quality of MP4 recording files will be severely compromised.

For more information, please see [Custom Capturing and Rendering](https://cloud.tencent.com/document/product/647/34066).

Declaration

public abstract void sendCustomAudioData(TRTCAudioFrame frame)

Parameters

Type	Name	Description
TRTCAudioFrame	frame	Audio frame. Currently, only mono audio, 48 kHz sample rate, and the LiteAVAudioFrameFormatPCM format are supported.

Remarks

You can leave the setting of timestamps to the SDK by setting timestamp to 0, but to avoid choppy audio, make sure that you call sendCustomAudioData at regular intervals.

sendCustomCmdMsg(Int32, Byte[], Int32, Boolean, Boolean)

Send a custom message to all users in the room

This API allows you to broadcast custom data to other users in the room through the audio/video data channel. Due to the use of this channel, it’s important that you control the message sending frequency and message size to avoid affecting the quality control logic of audio/video data.

Declaration

public abstract bool sendCustomCmdMsg(int cmdId, byte[] data, int dataSize, bool reliable, bool ordered)

Parameters

Type	Name	Description
System.Int32	cmdId	Message ID. Value range: 1-10
System.Byte[]	data	Message to send, which cannot exceed 1 KB
System.Int32	dataSize	Size of the data to send
System.Boolean	reliable	Whether to enable reliable messaging, which may cause latency as the recipient needs to retain the data for a while in case resending is required.
System.Boolean	ordered	Whether to enable ordered messaging, i.e., whether to require that data arrive in the same order as it is sent. This may cause latency as the recipient needs to retain the data for a while to sort it.

Returns

Type	Description
System.Boolean	true: the message is sent successfully; false: failed to send the message

Remarks

This API has the following limitations: You can send at most 30 messages to all users in the room (not supported on web and WeChat Mini Program). A data packet must not exceed 1 KB, or it may be dropped by an intermediate router or the server. Each client can send up to 8 KB of data per second. You must set both reliable and ordered to true or false. We strongly recommend that you use different cmdID for messages of different types. This can reduce message latency when ordered messaging is enabled.

sendCustomVideoData(TRTCVideoFrame)

Send custom video data to the SDK

We recommend the following settings for the parameters of `TRTCVideoFrame` (other parameters can be left empty).

-pixelFormat: only `LiteAVVideoPixelFormat_I420` is supported.

-bufferType: only `LiteAVVideoBufferType_Buffer` is supported.

-data: video frame buffer

-length: video frame size, whose value in I420 format is `width × height × 3/2`

-width: video width

-height: video height

-timestamp: Separate timestamps equally; otherwise, an audio-to-video sync error will occur and the quality of MP4 recording files will be severely compromised.

For more information, please see [Custom Capturing and Rendering](https://cloud.tencent.com/document/product/647/34066).

Declaration

public abstract void sendCustomVideoData(TRTCVideoFrame frame)

Parameters

Type	Name	Description
TRTCVideoFrame	frame	Video data in I420 format

Remarks

• The SDK has an internal frame rate control logic. It drops frames if the frame rate is higher than the target specified in setVideoEncoderParam and inserts frames if the frame rate is lower than the target.

- You can leave the setting of timestamps to the SDK by setting `timestamp` to `0`, but to avoid unstable frame rate, make sure that you call `sendCustomVideoData` at regular intervals.

sendSEIMsg(Byte[], Int32, Int32)

Insert small-volume custom data into video frames

Unlike `sendCustomCmdMsg`, `sendSEIMsg` inserts data directly into the header of video data. As a result, the data is retained even after the video frames are relayed to live streaming CDNs. However, the size of inserted data should be kept small, preferably several bytes.

The most common practice is using `sendSEIMsg` to insert custom timestamps into video frames, which can ensure that the messages and video images are in sync.

Declaration

public abstract bool sendSEIMsg(byte[] data, int dataSize, int repeatCount)

Parameters

Type	Name	Description
System.Byte[]	data	Data to send, which cannot exceed 1 KB
System.Int32	dataSize	Size of the data to send
System.Int32	repeatCount	Number of times to send the data

Returns

Type	Description
System.Boolean	true: the message meets the requirements and will be inserted into subsequent video frames; false: the message does not meet the requirements

Remarks

This API has the following limitations:

-The data is not sent the moment the API is called, but inserted into the next video frame.

-You can send at most 30 messages per second to all users in the room. This limit also applies to `sendCustomCmdMsg`.

-A data packet must not exceed 1 KB. Data packets too large may reduce video quality or cause video stutter. This limit also applies to `sendCustomCmdMsg`.

-Each client can send up to 8 KB of data per second. This limit also applies to `sendCustomCmdMsg`.

-If you send a message multiple times (`repeatCount` > 1), the data will be inserted into multiple subsequent frames (whose number equals `repeatCount`). This will drive up the video bitrate.

-If `repeatCount` is greater than 1, the same message may be returned via the `onRecvSEIMsg` callback multiple times, making deduplication necessary.

setAudioCaptureVolume(Int32)

Set the capturing volume of the SDK

Declaration

public abstract void setAudioCaptureVolume(int volume)

Parameters

Type	Name	Description
System.Int32	volume	Volume. Value range: 0-100

setAudioPlayoutVolume(Int32)

Set the playback volume of the SDK

Declaration

public abstract void setAudioPlayoutVolume(int volume)

Parameters

Type	Name	Description
System.Int32	volume	Volume. Value range: 0-150; default: 100

setBeautyStyle(TRTCBeautyStyle, UInt32, UInt32, UInt32)

Set the strength of the beauty, skin brightening, and rosy skin filters (not supported on Windows) The SDK has two built-in skin smoothing algorithms. One is "smooth", which features more obvious smoothing effect and is designed for showrooms. The other is "natural", which retains more facial details and is more natural.

Declaration

public abstract void setBeautyStyle(TRTCBeautyStyle style, uint beauty, uint white, uint ruddiness)

Parameters

Type	Name	Description
TRTCBeautyStyle	style	Beauty style, which may be smooth or natural. The former features more obvious skin smoothing effect and is suitable for entertainment scenarios.
System.UInt32	beauty
System.UInt32	white
System.UInt32	ruddiness

setConsoleEnabled(Boolean)

Enable/Disable console log printing

Declaration

public abstract void setConsoleEnabled(bool enabled)

Parameters

Type	Name	Description
System.Boolean	enabled	Whether to enable console log printing. It is disabled by default.

setDefaultStreamRecvMode(Boolean, Boolean)

Set the audio/video data receiving mode

It must be set before room entry to take effect.

Declaration

public abstract void setDefaultStreamRecvMode(bool autoRecvAudio, bool autoRecvVideo)

Parameters

Type	Name	Description
System.Boolean	autoRecvAudio	true (default): receive audio data automatically; false: call muteRemoteAudio to receive or stop receiving audio data
System.Boolean	autoRecvVideo	true (default): receive video data automatically; false: call startRemoteView/stopRemoteView to receive or stop receiving video data

Remarks

To ensure instant streaming, the SDK automatically receives audio/video upon successful room entry. This means you will receive the audio and video data of all remote users right after you enter a room. If you do not call startRemoteView, video data will be automatically canceled after the timeout period elapses. If your application scenario involves only audio (e.g., audio chat), you can use this API to disable the automatic receiving mode for video to reduce your costs.

setLocalVideoRenderCallback(TRTCVideoStreamType, TRTCVideoPixelFormat, TRTCVideoBufferType, ITRTCVideoRenderCallback)

Configure the custom rendering of local video

Declaration

public abstract int setLocalVideoRenderCallback(TRTCVideoStreamType streamType, TRTCVideoPixelFormat pixelFormat, TRTCVideoBufferType bufferType, ITRTCVideoRenderCallback callback)

Parameters

Type	Name	Description
TRTCVideoStreamType	streamType	Stream type
TRTCVideoPixelFormat	pixelFormat	Pixel format of the data returned
TRTCVideoBufferType	bufferType	Buffer type of the data returned
ITRTCVideoRenderCallback	callback	Custom rendering callback

Returns

Type	Description
System.Int32	0: successful; < 0: error

Remarks

After this API is called, the SDK will return the video data captured via a callback, which is then rendered by Unity’s Texture2D.

Call `setLocalVideoRenderCallback(TRTCVideoPixelFormat_Unknown, TRTCVideoBufferType_Unknown, null)` to disable the callback.

iOS, macOS, and Windows support only video frames in the pixel format of `TRTCVideoPixelFormat_BGRA32`. Unity can render data in BGRA32 format. Method: new Texture2D((int)_textureWidth, (int)_textureHeight, TextureFormat.BGRA32, false);

Android supports only video frames in the pixel format of `TRTCVideoPixelFormat_RGBA32`. Unity can render data in RGBA32 format. Method: new Texture2D((int)_textureWidth, (int)_textureHeight, TextureFormat.RGBA32, false);

setLogCallback(ITRTCLogCallback)

Set the log callback

Declaration

public abstract void setLogCallback(ITRTCLogCallback callback)

Parameters

Type	Name	Description
ITRTCLogCallback	callback	Log callback

setLogCompressEnabled(Boolean)

Enable/Disable local log compression

Compression can significantly reduce log size, but compressed logs can be read only after being decompressed by the Python script provided by Tencent Cloud.

If compression is disabled, logs will be stored in plaintext and will take up more storage space, but they can be read directly in Notepad.

Declaration

public abstract void setLogCompressEnabled(bool enabled)

Parameters

Type	Name	Description
System.Boolean	enabled	Whether to enable local log compression. It is disabled by default.

setLogDirPath(String)

Set the path to save logs

Declaration

public abstract void setLogDirPath(string path)

Parameters

Type	Name	Description
System.String	path	Path to save log files, e.g., "D:\Log", which must be converted to UTF-8

Remarks

Log files are stored in "C:/Users/[system username]/AppData/Roaming/Tencent/liteav/log" ("%appdata%/Tencent/liteav/log") by default. To change the path, you need to call this API before calling other APIs.

setLogLevel(TRTCLogLevel)

Set the log output level

Declaration

public abstract void setLogLevel(TRTCLogLevel level)

Parameters

Type	Name	Description
TRTCLogLevel	level	Log output level. For details, please see TRTCLogLevel. Default value: TRTCLogLevelNone

setMixTranscodingConfig(Nullable<TRTCTranscodingConfig>)

Set On-Cloud MixTranscoding parameters (not supported on Windows)

Declaration

public abstract void setMixTranscodingConfig(TRTCTranscodingConfig? config)

Parameters

Type	Name	Description
System.Nullable<TRTCTranscodingConfig>	config	For more information, please see the description of TRTCTranscodingConfig in TRTCTypeDef.h. If nullptr is passed in, On-Cloud MixTranscoding will be canceled.

Remarks

Notes: On-Cloud MixTranscoding will increase the delay of CDN live streaming by about 1-2 seconds. If you call this API, the streams of co-anchors will be mixed into your stream or the stream whose streamId is specified in config. If you are still in the room but do not need to mix streams any more, make sure that you pass in nullptr to cancel On-Cloud MixTranscoding. The On-Cloud MixTranscoding module starts working the moment you enable On-Cloud MixTranscoding. You may incur additional costs if you do not cancel it in a timely manner. When you leave the room, mixing will be canceled automatically.

Examples

If you enable relayed push on the "Function Configuration" page of the TRTC console, each stream in a room will have a default CDN playback address.

There may be multiple anchors in a room, each sending their own video and audio, but CDN audience needs only one live stream. Therefore, you need to mix multiple audio/video streams into one standard live stream, which requires mixtranscoding.

When you call the `setMixTranscodingConfig()` API, the SDK will send a command to the Tencent Cloud transcoding server to mix multiple audio/video streams in the room into one stream. You can use the `mixUsers` parameter to set the position of each channel of image and specify whether to mix only audio. You can also set the encoding parameters of the mixed stream, including `videoWidth`, `videoHeight`, and `videoBitrate`.

**Image 1** => decoding ====> \
                       \
**Image 2**=> decoding =>  image mixing => encoding => **mixed image**
                       /
**Image 3** => decoding ====> /
**Audio 1** => decoding ====> \
                       \
**Audio 2** => decoding => audio mixing => encoding => **mixed audio**
                       /
**Audio 3** => decoding ====> /

setNetworkQosParam(ref TRTCNetworkQosParam)

Set QoS parameters

The setting determines the SDK’s QoS policy under different network conditions, for example, whether to prioritize clarity or smoothness under poor network conditions.

Declaration

public abstract void setNetworkQosParam(ref TRTCNetworkQosParam param)

Parameters

Type	Name	Description
TRTCNetworkQosParam	param

setRemoteAudioVolume(String, Int32)

Set the playback volume of a remote user This API controls the volume of audio delivered to the system for playback. It affects the volume of local recording files, but not the volume of in-ear monitoring. If you want to set the volume to a value greater than 100, please contact technical support.

Declaration

public abstract void setRemoteAudioVolume(string userId, int volume)

Parameters

Type	Name	Description
System.String	userId	ID of the remote user
System.Int32	volume	Volume. Value range: 0-100

setRemoteVideoRenderCallback(String, TRTCVideoStreamType, TRTCVideoPixelFormat, TRTCVideoBufferType, ITRTCVideoRenderCallback)

Configure the custom rendering of remote video

This API is similar to `setLocalVideoRenderDelegate`, but it sets the callback of remote video rather than local video.

Declaration

public abstract int setRemoteVideoRenderCallback(string userId, TRTCVideoStreamType streamType, TRTCVideoPixelFormat pixelFormat, TRTCVideoBufferType bufferType, ITRTCVideoRenderCallback callback)

Parameters

Type	Name	Description
System.String	userId	User ID
TRTCVideoStreamType	streamType	Stream type
TRTCVideoPixelFormat	pixelFormat	Pixel format of the data returned
TRTCVideoBufferType	bufferType	Buffer type of the data returned
ITRTCVideoRenderCallback	callback	Custom rendering callback

Returns

Type	Description
System.Int32	0: successful; < 0: error

Remarks

After this API is called, the SDK will return decoded remote video data via a callback, which is then rendered by Unity’s Texture2D.

Call `setRemoteVideoRenderCallback(userId, TRTCVideoPixelFormat_Unknown, TRTCVideoBufferType_Unknown, nullptr)` to disable the callback.

iOS, macOS, and Windows support only video frames in the pixel format of `TRTCVideoPixelFormat_BGRA32`. Unity can render data in BGRA32 format. Method: new Texture2D((int)_textureWidth, (int)_textureHeight, TextureFormat.BGRA32, false);

Android supports only video frames in the pixel format of `TRTCVideoPixelFormat_RGBA32`. Unity can render data in RGBA32 format. Method: new Texture2D((int)_textureWidth, (int)_textureHeight, TextureFormat.RGBA32, false);

setRemoteVideoStreamType(String, TRTCVideoStreamType)

Specify whether to play a remote user’s big or small image

Declaration

public abstract void setRemoteVideoStreamType(string userId, TRTCVideoStreamType type)

Parameters

Type	Name	Description
System.String	userId	ID of the remote user
TRTCVideoStreamType	type	Type of the remote user’s video stream (big or small image) to play. Default value: TRTCVideoStreamTypeBig

Remarks

For this API to work, the remote user must first call enableEncSmallVideoStream to enable dual-channel encoding.

setSubStreamEncoderParam(ref TRTCVideoEncParam)

9.7 Set encoding parameters for screen sharing

Declaration

public abstract void setSubStreamEncoderParam(ref TRTCVideoEncParam param)

Parameters

Type	Name	Description
TRTCVideoEncParam	param	Substream encoding parameters. For details, please see the definition of TRTCVideoEncParam in TRTCCloudDef.h.

Remarks

You need to use the API to set encoding parameters for screen sharing even if you use the primary stream for screen sharing (by setting type to TRTCVideoStreamTypeBig when calling startScreenCapture).

setVideoEncoderMirror(Boolean)

Set the mirror mode of encoded video (not supported on Windows)

Declaration

public abstract void setVideoEncoderMirror(bool mirror)

Parameters

Type	Name	Description
System.Boolean	mirror	Whether to mirror video for remote users. true: mirror; false (default): do not mirror

Remarks

This API does not change the mirror mode of local video preview, but affects the video presented to remote users and recorded by the server.

setVideoEncoderParam(ref TRTCVideoEncParam)

3.9 Set video encoder parameters (not supported on Android and iOS)

Encoder parameters determine the quality of video watched by remote users and recorded in the cloud.

Declaration

public abstract void setVideoEncoderParam(ref TRTCVideoEncParam param)

Parameters

Type	Name	Description
TRTCVideoEncParam	param

setWaterMark(TRTCVideoStreamType, String, TRTCWaterMarkSrcType, UInt32, UInt32, Single, Single, Single)

Set a watermark (not supported on Android and Windows)

Declaration

public abstract void setWaterMark(TRTCVideoStreamType streamType, string srcData, TRTCWaterMarkSrcType srcType, uint nWidth, uint nHeight, float xOffset, float yOffset, float fWidthRatio)

Parameters

Type	Name	Description
TRTCVideoStreamType	streamType	Type of stream to watermark. Valid values: TRTCVideoStreamTypeBig, TRTCVideoStreamTypeSub`
System.String	srcData	Source of the watermark image. null means to remove the watermark.
TRTCWaterMarkSrcType	srcType	Data type of the watermark source
System.UInt32	nWidth	Pixel width of the watermark. This parameter is ignored if the watermark source is a file path.
System.UInt32	nHeight	Pixel height of the watermark. This parameter is ignored if the watermark source is a file path.
System.Single	xOffset	X-axis offset of the top-left corner of the watermark image. Value range: 0-1 (floating point number)
System.Single	yOffset	Y-axis offset of the top-left corner of the watermark image. Value range: 0-1 (floating point number)
System.Single	fWidthRatio	Ratio of the width of the watermark image to that of the video (the watermark image will be scaled by this ratio). Value range: 0-1 (floating point number)

startAudioRecording(ref TRTCAudioRecordingParams)

Start audio recording After this API is called, the SDK will record all audio of a call, including local audio, remote audio, and background music, into a file. This API works regardless of whether the user is in the room. When exitRoom is called, audio recording will stop automatically.

Declaration

public abstract int startAudioRecording(ref TRTCAudioRecordingParams audioRecordingParams)

Parameters

Type	Name	Description
TRTCAudioRecordingParams	audioRecordingParams	Audio recording parameters. For details, please see TRTCAudioRecordingParams.

Returns

Type	Description
System.Int32	0: recording successful; -1: recording has started; -2: failed to create the file or directory; -3: unsupported audio format

startLocalAudio(TRTCAudioQuality)

4.2 Enable local audio capturing and publishing

This API will start mic capturing and send audio data to other users in the room.

The SDK does not publish local audio automatically, so if you do not call this API, other users in the room will not hear your audio.

Declaration

public abstract void startLocalAudio(TRTCAudioQuality quality)

Parameters

Type	Name	Description
TRTCAudioQuality	quality	Audio quality. For details, please see TRTCAudioQuality.

Remarks

The TRTC SDK does not enable local mic capturing automatically.

startLocalPreview(Boolean, Object)

3.1 Enable local video preview

Declaration

public abstract void startLocalPreview(bool frontCamera, object rendObj)

Parameters

Type	Name	Description
System.Boolean	frontCamera	true: front camera; false: rear camera. This parameter takes effect only on Android and iOS. The front camera is used by default on Windows.
System.Object	rendObj	Only custom rendering is supported currently. Set this parameter to null.

Remarks

Only custom video rendering is supported currently. You need to call setLocalVideoRenderCallback first and then startLocalPreview to publish the local stream. The video frames will be returned via the onRenderFrame callback.

Rendering method: new Texture2D((int)_textureWidth, (int)_textureHeight, TextureFormat.BGRA32, false);

For more information, refer to `Demo/TRTCVideoRender.cs`.

startLocalRecording(ref TRTCLocalRecordingParams)

Start local recording This API records the audio and video data during live streaming into a file and saves it locally. Use cases:

1. If no streams are published, you can record local audio and video into a file after calling startLocalPreview.
2. If streams are published, you can record the entire live streaming session into a file and save it locally.

Declaration

public abstract void startLocalRecording(ref TRTCLocalRecordingParams localRecordingParams)

Parameters

Type	Name	Description
TRTCLocalRecordingParams	localRecordingParams	Recording parameters. For details, please see TRTCLocalRecordingParams.

startPublishCDNStream(TRTCPublishCDNParam)

2.3 Start publishing to the live streaming CDN of a non-Tencent Cloud vendor (not supported on Windows) The startPublishCDNStream() API is similar to startPublishing(), but it supports relaying to the live streaming CDN of a non-Tencent Cloud vendor.

Declaration

public abstract void startPublishCDNStream(TRTCPublishCDNParam param)

Parameters

Type	Name	Description
TRTCPublishCDNParam	param

startPublishing(String, TRTCVideoStreamType)

2.1 Start publishing to Tencent Cloud’s live streaming CDN

When calling this API, you need to specify a `StreamId` for the current user in Tencent Cloud’s CDN, which is used to splice the CDN playback address of the user.

For example, if you use the code below to set the `StreamId` of the current user's primary stream to `user_stream_001`, the CDN playback address of the user’s primary stream will be:

"http://yourdomain/live/user_stream_001.flv", where `yourdomain` is the domain name you use for playback.

You can configure your playback domain name in the [CSS console](https://console.cloud.tencent.com/live). Tencent Cloud doesn't provide a default playback domain name.

You can also specify `streamId` in `TRTCParams` when calling `enterRoom`. This method is recommended.

Declaration

public abstract void startPublishing(string streamId, TRTCVideoStreamType type)

Parameters

Type	Name	Description
System.String	streamId	Custom stream ID
TRTCVideoStreamType	type	Only TRTCVideoStreamTypeBig and TRTCVideoStreamTypeSub are supported.

Remarks

To play streams via CDNs, you need to enable relayed push on the "Function Configuration" page of the TRTC console.

• If you select "Specified stream for relayed push", you can use this API to publish a stream to Tencent Cloud’s CDN and specify a stream ID for it.
• If you select "Global auto-relayed push", you can use this API to change the default stream ID.

Examples

ITRTCCloud trtcCloud = ITRTCCloud.getTRTCShareInstance();
trtcCloud.enterRoom(params, TRTCAppScene.TRTCAppSceneLIVE);
trtcCloud.startLocalPreview(System.Object);
trtcCloud.startLocalAudio();
trtcCloud.startPublishing("user_stream_001", TRTCVideoStreamType.TRTCVideoStreamTypeBig);

startRemoteView(String, TRTCVideoStreamType, Object)

3.4 Start playing the video of a remote user

The `onUserVideoAvailable(userId, true)` callback indicates that a remote user has enabled video. After receiving this callback, call `startRemoteView(userId)` to load the user’s video. You can use a loading animation to improve user experience during the waiting period. When the first video frame of the remote user is rendered, you will receive the `onFirstVideoFrame(userId)` callback.

Declaration

public abstract void startRemoteView(string userId, TRTCVideoStreamType streamType, object rendObj)

Parameters

Type	Name	Description
System.String	userId	User ID of the remote user
TRTCVideoStreamType	streamType	Type of the remote user’s stream to play: HD big image: {@link TRTCVideoStreamTypeBig} Smooth small image: {@link TRTCVideoStreamTypeSmall}
System.Object	rendObj	Only custom rendering is supported currently. Set this parameter to null.

Remarks

Only custom video rendering is supported currently. You need to call setRemoteVideoRenderCallback first and then startRemoteView to pull the remote stream. The video frames will be returned via the onRenderFrame callback.

Rendering method: new Texture2D((int)_textureWidth, (int)_textureHeight, TextureFormat.BGRA32, false);

For more information, refer to `Demo/TRTCVideoRender.cs`.

startScreenCapture(TRTCVideoStreamType, ref TRTCVideoEncParam)

9.1 Start screen sharing

Declaration

public abstract void startScreenCapture(TRTCVideoStreamType type, ref TRTCVideoEncParam param)

Parameters

Type	Name	Description
TRTCVideoStreamType	type	Type of stream to use for screen sharing. Valid values: TRTCVideoStreamTypeBig: primary stream; TRTCVideoStreamTypeSub (default): substream
TRTCVideoEncParam	param	Encoding parameters for screen sharing images

Remarks

A user can publish at most one primary stream (TRTCVideoStreamTypeBig) and one substream (TRTCVideoStreamTypeSub) at the same time. By default, the screen is shared via the substream. If you use the primary stream, we recommend that you stop camera capturing (stopLocalPreview) to avoid conflicts.

startSpeedTest(Int32, String, String)

Start network speed testing (which should be avoided during video calls to ensure call quality)

The test result can be used to optimize the SDK's server selection policy, so you are advised to run the test before the first call, which will help the SDK select the optimal server.

In addition, if the test result is not satisfactory, you can show a UI message asking users to change to a better network.

Declaration

public abstract void startSpeedTest(int sdkAppId, string userId, string userSig)

Parameters

Type	Name	Description
System.Int32	sdkAppId	Application ID
System.String	userId	User ID
System.String	userSig	User signature

stopAllRemoteView()

Stop playing and pulling the videos of all remote users

Declaration

public abstract void stopAllRemoteView()

Remarks

The playing and pulling of the screen sharing stream, if any, will stop too.

stopAudioRecording()

Stop audio recording When exitRoom is called, audio recording will stop automatically.

Declaration

public abstract void stopAudioRecording()

stopLocalAudio()

4.3 Disable local audio capturing and publishing

After local audio capturing and publishing are disabled, other users in the room will receive the `onUserAudioAvailable(false)` callback.

Declaration

public abstract void stopLocalAudio()

stopLocalPreview()

3.2 Stop local video capturing and preview

Declaration

public abstract void stopLocalPreview()

stopLocalRecording()

Stop local recording When exitRoom is called, recording will stop automatically.

Declaration

public abstract void stopLocalRecording()

stopPublishCDNStream()

2.4 Stop publishing to the live streaming CDN of a non-Tencent Cloud vendor (not supported on Windows)

Declaration

public abstract void stopPublishCDNStream()

stopPublishing()

2.2 Stop publishing to Tencent Cloud’s live streaming CDN

Declaration

public abstract void stopPublishing()

stopRemoteView(String, TRTCVideoStreamType)

3.5 Stop playing and pulling the video of a remote user

After this API is called, the SDK will stop receiving the video of the specified user and release the resources used to display the video.

Declaration

public abstract void stopRemoteView(string userId, TRTCVideoStreamType streamType)

Parameters

Type	Name	Description
System.String	userId	User ID of the remote user
TRTCVideoStreamType	streamType	Type of the remote user’s stream to stop playing: HD big image: {@link TRTCVideoStreamTypeBig} Smooth small image: {@link TRTCVideoStreamTypeSmall}

stopScreenCapture()

9.2 Stop screen sharing

Declaration

public abstract void stopScreenCapture()

stopSpeedTest()

Stop network speed testing

Declaration

public abstract void stopSpeedTest()

switchRole(TRTCRoleType)

1.3 Switch roles. This API works only in live streaming scenarios (TRTCAppSceneLIVE and TRTCAppSceneVoiceChatRoom).

In live streaming scenarios, a user may need to switch between the “anchor” and “audience” roles.

You can set the role during room entry by specifying `role` in `TRTCParams`. You can also call `switchRole` to switch the role after room entry.

Declaration

public abstract void switchRole(TRTCRoleType role)

Parameters

Type	Name	Description
TRTCRoleType	role	Role, which is “anchor” by default: {@link TRTCRoleAnchor} Anchor, who can send audio and video. Up to 50 users are allowed to send audio and video at the same time in a room. {@link TRTCRoleAudience} Audience, who can play but cannot send audio or video. There is no upper limit on the audience size in a room.

switchRoom(TRTCSwitchRoomConfig)

Switch rooms

After calling this API, the user will leave the current room and immediately enter the room specified in `TRTCSwitchRoomConfig`. This API is easier to use than `exitRoom` + `enterRoom`.

It does not stop the user’s video capturing or preview. The result is returned through the `onSwitchRoom(errCode, errMsg)` callback in `ITRTCCloudCallback`.

Declaration

public abstract void switchRoom(TRTCSwitchRoomConfig config)

Parameters

Type	Name	Description
TRTCSwitchRoomConfig	config	Room switching parameters. For details, please see TRTCSwitchRoomConfig.