TRTCCloud

Create TRTCCloud main instance (singleton mode)

Terminate TRTCCloud main instance (singleton mode)

Note: All sub-instance created by main instance will also be terminated.

Create TRTCCloud sub-instance

Note: Only main instance can create sub-instance. Sub-instance cannot create sub sub-instance.

Get TRTCConfig object

TRTCConfig object can be used to open debug mode

Terminate current TRTCCloud instance

Enter Room

After calling this API, you will receive the onEnterRoom(result) event notification:

• If room entry succeeded, the result parameter will be a positive number (result > 0), indicating the time in milliseconds (ms) between function call and room entry.
• If room entry failed, the result parameter will be a negative number (result < 0), indicating the error code for room entry failure.

Parameter "scene" can be one of the following value:

• TRTCAppScene.TRTCAppSceneVideoCall:
  Video call scenario. Use cases: [one-to-one video call], [video conferencing with up to 300 participants], [online medical diagnosis], [small class], [video interview], etc. In this scenario, each room supports up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
  
• TRTCAppScene.TRTCAppSceneAudioCall:
  Audio call scenario. Use cases: [one-to-one audio call], [audio conferencing with up to 300 participants], [audio chat], [online Werewolf], etc. In this scenario, each room supports up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
  
• TRTCAppScene.TRTCAppSceneLIVE:
  Live streaming scenario. Use cases: [low-latency video live streaming], [interactive classroom for up to 100,000 participants], [live video competition], [video dating room], [remote training], [large-scale conferencing], etc. In this scenario, each room supports up to 100,000 concurrent online users, but you should specify the user roles: anchor (TRTCRoleAnchor) or audience (TRTCRoleAudience).
  
• TRTCAppScene.TRTCAppSceneVoiceChatRoom:
  Audio chat room scenario. Use cases: [Clubhouse], [online karaoke room], [music live room], [FM radio], etc. In this scenario, each room supports up to 100,000 concurrent online users, but you should specify the user roles: anchor (TRTCRoleAnchor) or audience (TRTCRoleAudience).
  

Notice:

1. If scene is specified as TRTCAppScene.TRTCAppSceneLIVE or TRTCAppScene.TRTCAppSceneVoiceChatRoom, you must use the role field in TRTCParams to specify the role of the current user in the room.
2. The same scene should be configured for all users in the same room.
3. Please try to ensure that TRTCCloud.enterRoom and TRTCCloud.exitRoom are used in pair; that is, please make sure that "the previous room is exited before the next room is entered"; otherwise, many issues may occur.

Exit room

Calling this API will allow the user to leave the current audio or video room and release the camera, mic, speaker, and other device resources. After resources are released, the SDK will emit the onExitRoom() event to notify you. If you need to call enterRoom again or switch to the SDK of another provider, we recommend you wait until you receive the onExitRoom() event, so as to avoid the problem of the camera or microphine being occupied.

Switch room

This API is used to quickly switch a user from one room to another.

• If the user's role is "audience", calling this API is equivalent to exitRoom (current room) + enterRoom (new room).
• If the user's role is "anchor", the API will retain the current audio/video publishing status while switching the room; therefore, during the room switch, camera preview and sound capturing will not be interrupted. This API is suitable for the online education scenario where the supervising teacher can perform fast room switch across multiple rooms. In this scenario, using switchRoom can get better smoothness and use less code than exitRoom + enterRoom. After calling this API, an onSwitchRoom(errCode, errMsg) event will be emitted.

Switch role

This API is used to switch the user role between "anchor" and "audience". As video live rooms and audio chat rooms need to support an audience of up to 100,000 concurrent online users, the rule "only anchors can publish their audio/video streams" has been set. Therefore, when some users want to publish their streams (so that they can interact with anchors), they need to switch their role to "anchor" first. You can use the role field in TRTCParams during room entry to specify the user role in advance or use the switchRole API to switch roles after room entry.

Notice:

1. This API is only applicable to two scenarios: live streaming (TRTCAppSceneLIVE) and audio chat room (TRTCAppSceneVoiceChatRoom).
2. If the scene you specify in enterRoom is TRTCAppSceneVideoCall or TRTCAppSceneAudioCall, call this API will not work.

Request cross-room call

By default, only users in the same room can make audio/video calls with each other, and the audio/video streams in different rooms are isolated from each other. However, you can publish the audio/video streams of an anchor in another room to the current room by calling this API. At the same time, this API will also publish the local audio/video streams to the target anchor's room. In other words, you can use this API to share the audio/video streams of two anchors in two different rooms, so that the audience in each room can watch the streams of these two anchors. This feature can be used to implement anchor competition. The result of requesting cross-room call will be returned through the onConnectOtherRoom() event. For example, after anchor A in room "101" uses connectOtherRoom() to successfully call anchor B in room "102":

• All users in room "101" will receive the onRemoteUserEnterRoom(B) and onUserVideoAvailable(B,1) event of anchor B; that is, all users in room "101" can subscribe to the audio/video streams of anchor B.
• All users in room "102" will receive the onRemoteUserEnterRoom(A) and onUserVideoAvailable(A,1) event of anchor A; that is, all users in room "102" can subscribe to the audio/video streams of anchor A.

For short, with connectOtherRoom, tow anchor in two different room can connect together, and all users in the room can see both anchor.

                                  Room 101                          Room 102
                            ---------------------               ---------------------
 Before cross-room call:   | Anchor:     A       |             | Anchor:     B       |
                           | Users :   U, V, W   |             | Users:   X, Y, Z    |
                            ---------------------               ---------------------

                                  Room 101                           Room 102
                            ---------------------               ---------------------
 After cross-room call:    | Anchors: A and B    |             | Anchors: B and A    |
                           | Users  : U, V, W    |             | Users  : X, Y, Z    |
                            ---------------------               ---------------------

For compatibility with subsequent extended fields for cross-room call, parameters in JSON format are used currently. If anchor A in room "101" wants to co-anchor with anchor B in room "102", then anchor A needs to pass in {"roomId": 102, "userId": "userB"} when calling this API.

After calling this API, the result will be returned through the onConnectOtherRoom event notification.

Exit cross-room call

The result will be returned through the onDisconnectOtherRoom event notification.

Set subscription mode

You can switch between the "automatic subscription" and "manual subscription" modes through this API:

• Automatic subscription: this is the default mode, where the user will immediately receive the audio/video streams in the room after room entry, so that the audio will be automatically played back, and the video will be automatically decoded (you still need to bind the rendering control through the startRemoteView API).
• Manual subscription: after room entry, the user needs to manually call the {@startRemoteView} API to start subscribing to and decoding the video stream and call the {@muteRemoteAudio} (false) API to start playing back the audio stream. In most scenarios, users will subscribe to the audio/video streams of all anchors in the room after room entry. Therefore, TRTC adopts the automatic subscription mode by default in order to achieve the best "instant streaming experience". In your application scenario, if there are many audio/video streams being published at the same time in each room, and each user only wants to subscribe to 1–2 streams of them, we recommend you use the "manual subscription" mode to reduce the traffic costs.

Notice: Must be set before room entry for it to take effect. YES: automatic subscription to audio; NO: manual subscription to audio by calling muteRemoteAudio(false). Default value: YES

Start publishing media stream

This API sends a command to the TRTC server, requesting it to push/encode the current user's audio and video streams to live streaming CDN or push back to the TRTC room. You can specify the specific publishing mode through TRTCPublishMode in the TRTCPublishTarget configuration.

Note:

1. The SDK will give you the task identifier (i.e., taskId) of the background task through the callback onStartPublishMediaStream.
2. The same task (with the same TRTCPublishMode and TRTCPublishCdnUrl) can only be started once. If you need to update or stop this task later, you need to record and use the returned taskId and operate through updatePublishMediaStream or stopPublishMediaStream.
3. The target supports configuring multiple CDN URLs at the same time (up to 10 at the same time). If your same forwarding/encoding task needs to be published to multiple CDN paths, you only need to configure multiple CDN URLs in the target. Even if there are multiple forwarding addresses for the same encoding task, the corresponding encoding billing is still charged only once.
4. Be careful not to push multiple tasks to the same URL address at the same time to avoid abnormal streaming status. A recommended solution is to use "sdkappid_roomid_userid_main" as a distinguishing identifier in the URL. This naming method is easy to recognize and will not conflict with your multiple applications.

Update Publishing Media Stream

This API sends a command to the TRTC server to update the media stream initiated by startPublishMediaStream.

Note:

1. You can use this API to update the published CDN URL (support adding and deleting, up to 10 at the same time). However, be careful not to push to the same URL address from multiple tasks at the same time to avoid abnormal streaming status.
2. You can update and adjust the forwarding/transcoding task through taskId. For example, in the PK business, you can initiate forwarding through startPublishMediaStream, and then update the forwarding to a transcoding task through taskId and this API when the anchor initiates PK. At this time, CDN playback will be continuous and there will be no stream disconnection (you need to keep the media stream encoding output parameter params consistent).
3. The same task does not support switching between audio-only, audio and video, and video-only.

Stop Publishing Media Stream

This API sends a command to the TRTC server to stop the media stream initiated by startPublishMediaStream.

Note:

1. If your business application does not save the taskId after your anchor exits abnormally and re-enters, if you need to obtain the taskId again, you can call startPublishMediaStream to start the task again. At this time, the TRTC backend will return a task start failure and bring you the taskId of the last start.
2. If taskId is an empty string, it will stop all media streams initiated by startPublishMediaStream by this user. If you have only started one media stream or want to stop all media streams initiated by you, it is recommended to use this method.

Start publishing audio/video streams to Tencent Cloud CSS CDN

This API sends a command to the TRTC server, requesting it to relay the current user's audio/video streams to CSS CDN. You can set the StreamId of the live stream through the streamId parameter, so as to specify the playback address of the user's audio/video streams on CSS CDN. For example, if you specify the current user's live stream ID as user_stream_001 through this API, then the corresponding CDN playback address is: "http://yourdomain/live/user_stream_001.flv", where yourdomain is your playback domain name with an ICP filing.

You can configure your playback domain name in the CSS console. Tencent Cloud does not provide a default playback domain name.

You can also specify the streamId when setting the TRTCParams parameter of enterRoom, which is the recommended approach.

Notice: You need to enable the "Enable Relayed Push" option on the "Function Configuration" page in the TRTC console in advance.

• If you select "Specified stream for relayed push", you can use this API to push the corresponding audio/video stream to Tencent Cloud CDN and specify the entered stream ID.
• If you select "Global auto-relayed push", you can use this API to adjust the default stream ID.

Stop publishing audio/video streams to Tencent Cloud CSS CDN

Start publishing audio/video streams to non-Tencent Cloud CDN

This API is similar to the startPublishing API. The difference is that startPublishing can only publish audio/video streams to Tencent Cloud CDN, while this API can relay streams to live streaming CDN services of other cloud providers.

Notice:

• Using the startPublishing API to publish audio/video streams to Tencent Cloud CSS CDN does not incur additional fees.
• Using the startPublishCDNStream API to publish audio/video streams to non-Tencent Cloud CDN incurs additional relaying bandwidth fees.

Stop publishing audio/video streams to non-Tencent Cloud CDN

Set the layout and transcoding parameters of On-Cloud MixTranscoding

In a live room, there may be multiple anchors publishing their audio/video streams at the same time, but for audience on CSS CDN, they only need to watch one video stream in HTTP-FLV or HLS format. When you call this API, the SDK will send a command to the TRTC mixtranscoding server to combine multiple audio/video streams in the room into one stream. You can use the TRTCTranscodingConfig parameter to set the layout of each channel of image. You can also set the encoding parameters of the mixed audio/video streams. For more information, please see On-Cloud MixTranscoding.

    **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 ====> //

Enable the preview image of local camera

If this API is called before enterRoom, the SDK will only enable the camera and wait until enterRoom is called before starting push. If it is called after enterRoom, the SDK will enable the camera and automatically start pushing the video stream.

When the first camera video frame starts to be rendered, you will receive the onFirstVideoFrame event.

This API will start system default camera, if you want to change camera devce, you can use setCurrentCameraDevice() to choose other camera device.

Stop camera preview

Update the HTML element used to hold local camera video

This interface can be used in cases where you want to change the place where you preview your local camera on the HTML page.

Notice: You should call this interface after calling startLocalPreview(), or else it will not work.

• If you has called startLocalPreview(view) with an valid HTML parameter， you can call this interface to change the HTML element holding the captured video from your camera
• If you has called startLocalPreview(null) with null, you can call this interface to start preview the captured video from your camera

Set camera acquisition parameter

Notice: Only Windows operating system supported.

Pause/Resume publishing local video stream

This API can pause (or resume) publishing the local video image. After the pause, other users in the same room will not be able to see the local image. This API is equivalent to the two APIs of startLocalPreview/stopLocalPreview when TRTCVideoStreamTypeBig is specified, but has higher performance and response speed. The startLocalPreview/stopLocalPreview APIs need to enable/disable the camera, which are hardware device-related operations, so they are very time-consuming. In contrast, muteLocalVideo only needs to pause or allow the data stream at the software level, so it is more efficient and more suitable for scenarios where frequent enabling/disabling are needed. After local video publishing is paused, other members in the same room will receive the onUserVideoAvailable(userId, 0) event notification. After local video publishing is resumed, other members in the same room will receive the onUserVideoAvailable(userId, 1) event notification.

Set placeholder image during local video pause

When you call muteLocalVideo(true) to pause the local video image, you can set a placeholder image by calling this API. Then, other users in the room will see this image instead of a black screen.

Subscribe to a remote user's video stream and bind it to a video rendering control

If you call this API, the SDK will pull the video stream of the specified userId and render it to the rendering control specified by the view parameter. You can set the display mode of the video image using setRemoteRenderParams.

• If you already know the userId of the user who is publishing video in the room, you can call startRemoteView to subscribe to the user's video.
• If you don't know who is publishing video in the room, you can wait for the onUserVideoAvailable event after a remote user calls enterRoom.

If you receive the onUserVideoAvailable(userId, 1) event from the SDK, it indicates that the remote user has enabled video. After receiving this event, 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 this user is displayed, you will receive the onFirstVideoFrame(userId) event.

Notes:

1. The SDK supports playing a user’s big/small image and substream image at the same time, but does not support playing the big and small images at the same time.
2. The small image of a user can be played only if the user has called enableSmallVideoStream to enable the dual-stream mode.
3. If the small image of the specified user does not exist, the SDK will switch to the big image.

Stop subscribing to remote user's video stream and release rendering control

Calling this API will cause the SDK to stop receiving the user's video stream and release the decoding and rendering resources for the stream.

Update the HTML element used to hold remote user video

This interface can be used in cases where you want to change the place where you preview remote user camera or screen-sharing video on the HTML page.

Notice: This interface should be called after calling startRemoteView(), or else it will not work.

Stop subscribing to all remote users' video streams and release all rendering resources

Calling this API will cause the SDK to stop receiving all remote video streams and release all decoding and rendering resources. Notice: If a substream image (screen sharing) is being displayed, it will also be stopped.

Pause/Resume subscribing to remote user's video stream

This API only pauses/resumes receiving the specified user's video stream but does not release displaying resources; therefore, the video image will freeze at the last frame before it is called. Notice: This API can be called before room entry (enterRoom), and the pause status will be reset after room exit (exitRoom).

Pause/Resume subscribing to all remote users' video streams

This API only pauses/resumes receiving all users' video streams but does not release displaying resources; therefore, the video image will freeze at the last frame before it is called. Notice: This API can be called before room entry (enterRoom), and the pause status will be reset after room exit (exitRoom).

Set the encoding parameters of video encoder

This setting can determine the quality of image viewed by remote users, which is also the image quality of on-cloud recording files.

Set network quality control parameters

This setting determines the quality control policy in a poor network environment, such as "image quality preferred" or "smoothness preferred".

Set the rendering parameters of local video image

The parameters that can be set include video image rotation angle, fill mode, and mirror mode.

Set the rendering mode of the local image (deprecated)

Set the rendering mode of remote video image

Set the rendering mode of a remote image (deprecated)

Set the rotation of the local image (deprecated)

Set the rotation of a remote image (deprecated)

Set the direction of image output by video encoder

This setting does not affect the preview direction of the local video image, but affects the direction of the image viewed by other users in the room (and on-cloud recording files). When a phone or tablet is rotated upside down, as the capturing direction of the camera does not change, the video image viewed by other users in the room will become upside-down. In this case, you can call this API to rotate the image encoded by the SDK 180 degrees, so that other users in the room can view the image in the normal direction. If you want to achieve the aforementioned user-friendly interactive experience, we recommend you directly call setGSensorMode to implement smarter direction adaptation, with no need to call this API manually.

Turn on/off the mirror mode for the local camera preview (deprecated)

Set the mirror mode of image output by encoder

This setting does not affect the mirror mode of the local video image, but affects the mirror mode of the image viewed by other users in the room (and on-cloud recording files).

Enable dual-channel encoding mode with big and small images

In this mode, the current user's encoder will output two channels of video streams, i.e., HD big image and Smooth small image, at the same time (only one channel of audio stream will be output though). In this way, other users in the room can choose to subscribe to the HD big image or Smooth small image according to their own network conditions or screen size.

Notice: Dual-channel encoding will consume more CPU resources and network bandwidth; therefore, this feature can be enabled on macOS, Windows, or high-spec tablets, but is not recommended for phones.

Switch the big/small image of specified remote user

After an anchor in a room enables dual-channel encoding, the video image that other users in the room subscribe to through startRemoteView will be HD big image by default. You can use this API to select whether the image subscribed to is the big image or small image. The API can take effect before or after startRemoteView is called.

Notice: To implement this feature, the target user must have enabled the dual-channel encoding mode through enableEncSmallVideoStream; otherwise, this API will not work.

Video snapshot

You can use this API to take snapshot of the local video image or the primary stream image and substream (screen sharing) image of a remote user. After calling this API, an onSnapshotComplete event will be emitted.

Set video references for playback (deprecated)

For low-end devices, we recommend you set the parameter to TRTCVideoStreamTypeSmall (small image). This API will not work if a remote user hasn’t enabled the dual-stream mode.

Start local media recording

This API records the audio/video content during live streaming into a local file.

Stop local media recording

If a recording task has not been stopped through this API before room exit, it will be automatically stopped after room exit.

Enable local audio capturing and publishing

The SDK does not enable the microphone by default. When a user wants to publish the local audio, the user needs to call this API to enable microphone capturing and encode and publish the audio to the current room. After local audio capturing and publishing is enabled, other users in the room will receive the onUserAudioAvailable(userId, 1) notification.

Stop local audio capturing and publishing

After local audio capturing and publishing is stopped, other users in the room will receive the onUserAudioAvailable(userId, false) event notification.

Pause/Resume publishing local audio stream

After local audio publishing is paused, other users in the room will receive the onUserAudioAvailable(userId, false) notification. After local audio publishing is resumed, other users in the room will receive the onUserAudioAvailable(userId, true) notification. Different from stopLocalAudio, muteLocalAudio(true) does not release the mic permission; instead, it continues to send mute packets with extremely low bitrate. This is very suitable for scenarios that require on-cloud recording, as video file formats such as MP4 have a high requirement for audio continuity, while an MP4 recording file cannot be played back smoothly if stopLocalAudio is used. Therefore, muteLocalAudio instead of stopLocalAudio is recommended in scenarios where the requirement for recording file quality is high.

Pause/Resume playing back remote audio stream

When you mute the remote audio of a specified user, the SDK will stop playing back the user's audio and pulling the user's audio data.

Notice: This API works when called either before or after room entry (enterRoom), and the mute status will be reset to false after room exit (exitRoom).

Pause/Resume playing back all remote users' audio streams

When you mute the audio of all remote users, the SDK will stop playing back all their audio streams and pulling all their audio data.

Notice: This API works when called either before or after room entry (enterRoom), and the mute status will be reset to false after room exit (exitRoom).

Set the audio playback volume of remote user

You can mute the audio of a remote user through setRemoteAudioVolume(userId, 0).

Notice: If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.

Set the capturing volume of local audio

Notice: If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.

Get the capturing volume of local audio

Set the playback volume of remote audio

This API controls the volume of the sound ultimately delivered by the SDK to the system for playback. It affects the volume of the recorded local audio file but not the volume of in-ear monitoring.

Notice: If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.

Get the playback volume of remote audio

Enable volume reminder

After this feature is enabled, the SDK will return the remote audio volume in the onUserVoiceVolume event notification.

Start audio recording

After this API is called, the SDK will record all audios of a call, including the local audio, remote audios, background music, and audio effects, into a file. This API works both before and after room entry. Recording will stop automatically after room exit, even if stopAudioRecording is not called.

Notes:

• The path must contain the filename and extension. The extension determines the format of the recording file. Supported formats include PCM, WAV, and AAC. For example, if you set the path to mypath/record/audio.aac, the SDK will generate an audio recording file in AAC format. Please specify a valid path with read/write permissions; otherwise, the system will fail to generate audio recording files.
• In versions earlier than 9.3, params (required), which indicates the path to save recording files, must be a string. In v9.3 and later versions, the parameter can be a string or TRTCAudioRecordingParams.

Stop audio recording

If a recording task has not been stopped through this API before room exit, it will be automatically stopped after room exit.

Set up an intelligent concurrent playback strategy for remote audio streams

For room with many speakers.

Set audio quality

Notice:

1. The higher sound quality will bring better listening experience, but need more bandwidth. It will be more likely to get stuck in weak network scenario.

Set microphone volume (deprecated)

Get the list of cameras

Set the camera to be used

Get the camera currently in use

Get the list of microphones

Get the microphone currently in use

Set the mic to use

This API is used to set the mic to use. If you do not call this API, the mic whose index is 0 that returned from getMicDevicesList will be used.

Get the current mic volume

This API is used to get the capturing volume of the mic. Note: This API returns the audio volume of the hardware.

Set the current mic volume

This API is used to set the capturing volume of the mic. Note: This API sets the system capturing volume. If the user adjusts the system capturing volume manually, the volume set by the API will be overwritten.

Set the mute status of the microphone currently in use

Get the mute status of the microphone currently in use

Get the list of speakers

Get the speaker currently in use

Set the speaker currently in use

Get the current speaker volume

This API is used to get the playback volume of the speaker.

Set the current speaker volume

This API is used to set the playback volume of the speaker.

Note: This API sets the system playback volume. If the user adjusts the system playback volume manually, the volume set by the API will be overwritten.

Set the mute status of the speaker currently in use

Get the mute status of the speaker currently in use

Set the audio device used by SDK to follow the system default device

Support microphone and speaker, camera not supported.

Set special effects such as beauty, brightening, and rosy skin filters

The SDK is integrated with two skin smoothing algorithms of different styles:

• "Smooth" style, which uses a more radical algorithm for more obvious effect and is suitable for show live streaming.
• "Natural" style, which retains more facial details for more natural effect and is suitable for most live streaming use cases.

Note: The computer must be equipped with a graphics card, otherwise the function will not take effect.

Add watermark

The watermark position is determined by the xOffset, yOffset, and fWidthRatio parameters.

• xOffset: X coordinate of watermark, which is a floating-point number between 0 and 1.
• yOffset: Y coordinate of watermark, which is a floating-point number between 0 and 1.
• fWidthRatio: watermark dimensions ratio, which is a floating-point number between 0 and 1.

Start displaying the substream image of remote user (deprecated)

• startRemoteView used to display big stream image (TRTCVideoStreamTypeBig, commonly used for camera).
• This API is used to display substream image (TRTCVideoStreamTypeSub, commonly used for screen sharing).

Stop displaying the substream image of remote user (deprecated)

Substream (TRTCVideoStreamTypeSub, commonly used for screen sharing).

Set the fill mode of substream image (deprecated)

• setRemoteViewFillMode used to set big image fill-mode (TRTCVideoStreamTypeBig, commonly used for camera).
• This API is used to set substream fill-mode (TRTCVideoStreamTypeSub, commonly used for screen sharing).

Set the clockwise rotation angle of substream image (deprecated)

• TRTCCloud#setRemoteViewRotation used to set the clockwise rotation angle of big stream (TRTCVideoStreamTypeBig, commonly used for camera)
• This API is used to set the clockwise rotation angle of substream image (TRTCVideoStreamTypeSub, commonly used for screen sharing)

Enumerate shareable screens and windows

When you integrate the screen sharing feature of a desktop system, you generally need to display a UI for selecting the sharing target, so that users can use the UI to choose whether to share the entire screen or a certain window. Through this API, you can query the IDs, names, and thumbnails of sharable windows on the current system. We provide a default UI implementation in the demo for your reference. Notice: The returned list contains the screen and the application windows. The screen is the first element in the list. If the user has multiple displays, then each display is a sharing target.

Select the screen or window to share

After you get the sharable screens and windows through getScreenCaptureSources, you can call this API to select the target screen or window you want to share. During the screen sharing process, you can also call this API at any time to switch the sharing target.

The following four sharing modes are supported:

• Sharing the entire screen: for source whose type is Screen in sourceInfoList, set captureRect to { 0, 0, 0, 0 }.
• Sharing a specified area: for source whose type is Screen in sourceInfoList, set captureRect to a non-nullptr value, e.g., { 100, 100, 300, 300 }.
• Sharing an entire window: for source whose type is Window in sourceInfoList, set captureRect to { 0, 0, 0, 0 }.
• Sharing a specified window area: for source whose type is Window in sourceInfoList, set captureRect to a non-nullptr value, e.g., { 100, 100, 300, 300 }.

Notice: Due to the capability of Operating System API, When you switch sharing windows created from the same application for the second time and on, the selected window may not be popped on the top all other windows. You need to bring it top manually.

Start desktop screen sharing

This API can capture the screen content of the entire macOS system or a specified application and share it with other users in the same room.

Notice:

1. A user can publish at most one primary stream (TRTCVideoStreamTypeBig) and one substream (TRTCVideoStreamTypeSub) at the same time.
2. By default, screen sharing uses the substream image. If you want to use the primary stream for screen sharing, you need to stop camera capturing (through stopLocalPreview) in advance to avoid conflicts.
3. Only one user can use the substream for screen sharing in the same room at any time; that is, only one user is allowed to enable the substream in the same room at any time.
4. When there is already a user in the room using the substream for screen sharing, calling this API will receive the onError(ERR_SERVER_CENTER_ANOTHER_USER_PUSH_SUB_VIDEO) event notification.

Pause screen sharing

Resume screen sharing

Stop screen sharing

Set the video encoding parameters of screen sharing (i.e., substream)

This API can set the image quality of screen sharing (i.e., the substream) viewed by remote users, which is also the image quality of screen sharing in on-cloud recording files. Please note the differences between the following two APIs:

• setVideoEncoderParam is used to set the video encoding parameters of the primary stream image (TRTCVideoStreamTypeBig, generally for camera).
• setSubStreamEncoderParam is used to set the video encoding parameters of the substream image (TRTCVideoStreamTypeSub, generally for screen sharing).

Notice: Even if you use the primary stream to transfer screen sharing data (set type=TRTCVideoStreamTypeBig when calling startScreenCapture), you still need to call the setSubStreamEncoderParam API instead of the setVideoEncoderParam API to set the screen sharing encoding parameters.

Set the audio mixing volume of screen sharing

The greater the value, the larger the ratio of the screen sharing volume to the mic volume. We recommend you not set a high value for this parameter as a high volume will cover the mic sound.

Add specified windows to the exclusion list of screen sharing

The excluded windows will not be shared. This feature is generally used to add a certain application's window to the exclusion list to avoid privacy issues. You can set the filtered windows before starting screen sharing or dynamically add the filtered windows during screen sharing.

Notice:

1. This API takes effect only if the type in TRTCScreenCaptureSourceInfo is specified as TRTCScreenCaptureSourceTypeScreen; that is, the feature of excluding specified windows works only when the entire screen is shared.
2. The windows added to the exclusion list through this API will be automatically cleared by the SDK after room exit.
3. On macOS, please pass in the window ID (CGWindowID), which can be obtained through the sourceId member in TRTCScreenCaptureSourceInfo.

Remove specified windows from the exclusion list of screen sharing

Remove all windows from the exclusion list of screen sharing

Add specified windows to the inclusion list of screen sharing

This API takes effect only if the type in TRTCScreenCaptureSourceInfo is specified as TRTCScreenCaptureSourceTypeWindow; that is, the feature of additionally including specified windows works only when a window is shared.
You can call it before or after startScreenCapture.

Notice: The windows added to the inclusion list by this method will be automatically cleared by the SDK after room exit.

Remove specified windows from the inclusion list of screen sharing

This API takes effect only if the type in TRTCScreenCaptureSourceInfo is specified as TRTCScreenCaptureSourceTypeWindow.
That is, the feature of additionally including specified windows works only when a window is shared.

Remove all windows from the inclusion list of screen sharing

This API takes effect only if the type in TRTCScreenCaptureSourceInfo is specified as TRTCScreenCaptureSourceTypeWindow.
That is, the feature of additionally including specified windows works only when a window is shared.

Enable custom audio capturing mode

After this mode is enabled, the SDK will not run the original audio capturing process (i.e., stopping mic data capturing) and will retain only the audio encoding and sending capabilities. You need to use sendCustomAudioData to continuously insert the captured audio data into the SDK.

Notice：

• Custom audio capture and the default microphone audio capture in the SDK are mutually exclusive. Therefore, before calling enableCustomAudioCapture(true) to enable custom audio capture, you need to call the stopLocalAudio interface to turn off the default microphone audio upload, otherwise it will not take effect. After calling enableCustomAudioCapture(false) to turn off custom capture, you need to call the stopLocalAudio interface to enable the SDK's default microphone audio capture.
• As acoustic echo cancellation (AEC) requires strict control over the audio capturing and playback time, after custom audio capturing is enabled, AEC may fail.

Deliver captured audio data to SDK

We recommend you enter the following information for the TRTCAudioFrame parameter (other fields can be left empty):

• audioFormat: audio data format, which can only be TRTCAudioFrameFormatPCM .
• data: audio frame buffer. Audio frame data must be in PCM format, and it supports a frame length of 5–100 ms (20 ms is recommended). Length calculation method: for example, if the sample rate is 48000, then the frame length for mono channel will be 48000 * 0.02s * 1 * 16 bit = 15360 bit = 1920 bytes.
• sampleRate: sample rate. Valid values: 16000, 24000, 32000, 44100, 48000.
• channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel.
• timestamp (ms): Set it to the timestamp when audio frames are captured, which you can obtain by calling generateCustomPTS after getting a audio frame.

Notice: Please call this API accurately at intervals of the frame length; otherwise, sound lag may occur due to uneven data delivery intervals.

Enable/Disable custom audio track

After this feature is enabled, you can mix a custom audio track into the SDK through this API. With two boolean parameters, you can control whether to play back this track remotely or locally.

Notice: If you specify both enablePublish and enablePlayout as false , the custom audio track will be completely closed.

Mix custom audio track into SDK

Before you use this API to mix custom PCM audio into the SDK, you need to first enable custom audio tracks through enableMixExternalAudioFrame. You are expected to feed audio data into the SDK at an even pace, but we understand that it can be challenging to call an API at absolutely regular intervals. Given this, we have provided a buffer pool in the SDK, which can cache the audio data you pass in to reduce the fluctuations in intervals between API calls. The value returned by this API indicates the size (ms) of the buffer pool. For example, if 50 is returned, it indicates that the buffer pool has 50 ms of audio data. As long as you call this API again within 50 ms, the SDK can make sure that continuous audio data is mixed. If the value returned is 100 or greater, you can wait after an audio frame is played to call the API again. If the value returned is smaller than 100 , then there isn’t enough data in the buffer pool, and you should feed more audio data into the SDK until the data in the buffer pool is above the safety level.

Fill the fields in TRTCAudioFrame as follows (other fields are not required):

• data : audio frame buffer. Audio frames must be in PCM format. Each frame can be 5-100 ms (20 ms is recommended) in duration. Assume that the sample rate is 48000, and sound channels mono-channel. Then the frame size would be 48000 x 0.02s x 1 x 16 bit = 15360 bit = 1920 bytes.
• sampleRate : sample rate. Valid values: 16000, 24000, 32000, 44100, 48000
• channel : number of sound channels (if dual-channel is used, data is interleaved). Valid values: 1 (mono-channel); 2 (dual channel)
• timestamp : timestamp (ms). Set it to the timestamp when audio frames are captured, which you can obtain by calling generateCustomPTS after getting an audio frame.

Notice：

• To mix in custom audio tracks, there needs to be an upstream audio stream driver. Supported upstream audio streams include:
  • The default microphone audio stream captured by the SDK, which can be enabled by calling the startLocalAudio interface.
• Please call this interface accurately at the interval of each frame duration. Uneven data delivery intervals can easily cause sound stuttering.

Set the publish volume and playback volume of mixed custom audio track

Generate custom capturing timestamp

This API is only suitable for the custom capturing mode and is used to solve the problem of out-of-sync audio/video caused by the inconsistency between the capturing time and delivery time of audio/video frames.

When you call APIs such as sendCustomAudioData for custom video or audio capturing, please use this API as instructed below:

1. First, when a video or audio frame is captured, call this API to get the corresponding PTS timestamp.
2. Then, send the video or audio frame to the preprocessing module you use (such as a third-party beauty filter or sound effect component).
3. When you actually call sendCustomAudioData for delivery, assign the PTS timestamp recorded when the frame was captured to the timestamp field in TRTCVideoFrame or TRTCAudioFrame.

Set custom audio data callback

After this callback is set, the SDK will internally call back the audio data (in PCM format), including: {onCapturedAudioFrame}：callback of the audio data captured by the local mic

• {onLocalProcessedAudioFrame}：callback of the audio data captured by the local mic and preprocessed by the audio module
• {onPlayAudioFrame}：audio data from each remote user before audio mixing
• {onMixedPlayAudioFrame}：callback of the audio data that will be played back by the system after audio streams are mixed
• {onMixedAllAudioFrame}：Data mixed from all the captured and to-be-played audio in the SDK

Note: Setting the callback to null indicates to stop the custom audio callback, while setting it to a non-null value indicates to start the custom audio callback.

Use UDP channel to send custom message to all users in room

This API allows you to use TRTC's UDP channel to broadcast custom data to other users in the current room for signaling transfer. The UDP channel in TRTC was originally designed to transfer audio/video data. This API works by disguising the signaling data you want to send as audio/video data packets and sending them together with the audio/video data to be sent. Other users in the room can receive the message through the onRecvCustomCmdMsg event.

Notice:

1. Up to 30 messages can be sent per second to all users in the room (this is not supported for web and mini program currently).
2. A packet can contain up to 1 KB of data; if the threshold is exceeded, the packet is very likely to be discarded by the intermediate router or server.
3. A client can send up to 8 KB of data in total per second.
4. reliable and ordered must be set to the same value (true or false) and cannot be set to different values currently.
5. We strongly recommend you set different cmdID values for messages of different types. This can reduce message delay when orderly sending is required.

Use SEI channel to send custom message to all users in room

This API allows you to use TRTC's SEI channel to broadcast custom data to other users in the current room for signaling transfer. The header of a video frame has a header data block called SEI. This API works by embedding the custom signaling data you want to send in the SEI block and sending it together with the video frame. Therefore, the SEI channel has a better compatibility than sendCustomCmdMsg as the signaling data can be transferred to the CSS CDN along with the video frame. However, because the data block of the video 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. The most common use is to embed the custom timestamp into video frames through this API so as to implement a perfect alignment between the message and video image (such as between the teaching material and video signal in the education scenario). Other users in the room can receive the message through the onRecvSEIMsg event.

Notice: This API has the following restrictions:

• 1. The data will not be instantly sent after this API is called; instead, it will be inserted into the next video frame after the API call.

2. Up to 30 messages can be sent per second to all users in the room (this limit is shared with sendCustomCmdMsg).
3. Each packet can be up to 1 KB (this limit is shared with sendCustomCmdMsg). If a large amount of data is sent, the video bitrate will increase, which may reduce the video quality or even cause lagging.
4. Each client can send up to 8 KB of data in total per second (this limit is shared with sendCustomCmdMsg).
5. If multiple times of sending is required (i.e., repeatCount > 1), the data will be inserted into subsequent repeatCount video frames in a row for sending, which will increase the video bitrate.
6. If repeatCount is greater than 1, the data will be sent for multiple times, and the same message may be received multiple times in the onRecvSEIMsg event; therefore, deduplication is required.

Play background music (deprecated)

Stop background music (deprecated)

Pause background music (deprecated)

Resume background music (deprecated)

Get the total length of background music in ms (deprecated)

Set background music playback progress (deprecated)

Set background music volume (deprecated)

When background music is started. This API is used to set background music volume for both local and remote user.

Set the local playback volume of background music (deprecated)

When background music is started. This API can be used to set the playing volume of background music for local user.

Set the remote playback volume of background music (deprecated)

When background music is started. This API can be used to set the playing volume of background music for remote user.

Enable system audio capturing

This API captures audio data from the sound card of the anchor’s computer and mixes it into the current audio stream of the SDK. This ensures that other users in the room hear the audio played back by the anchor’s computer. In online education scenarios, a teacher can use this API to have the SDK capture the audio of instructional videos and broadcast it to students in the room. In live music scenarios, an anchor can use this API to have the SDK capture the music played back by his or her player so as to add background music to the room.

Stop system audio capturing

Set the volume of system audio capturing

Setting the background music callback

Before playing background music, please use this API to set the music callback, which can inform you of the playback progress.

Starting background music

Stopping background music

Pausing background music

Resuming background music

Getting the total length (ms) of background music

Setting the playback progress (ms) of background music

Notice: Do not call this API frequently as the music file may be read and written to each time the API is called, which can be time-consuming. Wait till users finish dragging the progress bar before you call this API. The progress bar controller on the UI tends to update the progress at a high frequency as users drag the progress bar. This will result in poor user experience unless you limit the frequency.

Setting the local and remote playback volume of background music

This API is used to set the local and remote playback volume of background music.

• Local volume: the volume of music heard by anchors
• Remote volume: the volume of music heard by audience

Notice: If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.

Setting the local playback volume of a specific music track

This API is used to control the local playback volume (the volume heard by anchors) of a specific music track.

Setting the remote playback volume of a specific music track

This API is used to control the remote playback volume (the volume heard by audience) of a specific music track.

Notice: If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.

Turn on the ear monitor.

After the anchor opens the ear monitor, he can hear his own voice collected by the microphone in the headset. The special effect is applicable to the application scene of the anchor singing.

It should be noted that due to the high hardware delay of the Bluetooth headset, this special effect cannot be turned on when the anchor wears the Bluetooth headset. Please try to prompt the anchor to wear the wired headset on the user interface. At the same time, it should be noted that not all mobile phones can achieve excellent ear return effect after turning on this special effect, and we have blocked the special effect for some mobile phones with poor ear return effect.

Setting ear monitor volume

Through this interface, you can set the volume level of the sound in the ear monitor effect.

If you feel that the volume is still too low after setting the volume to 100, you can increase the volume to a maximum of 150, but please note that setting the volume above 100 may risk causing distortion, so please proceed with caution.

Setting voice volume

This interface allows you to set the volume level of the voice, which is typically used in conjunction with the setAllMusicVolume interface for adjusting the music volume. It helps to tune the respective volume proportions of voice and music before mixing.

If you find that the volume is still too low after setting the volume to 100, you can increase the volume to a maximum of 150. However, please note that setting the volume above 100 may risk causing distortion, so please proceed with caution.

Setting voice pitch

This interface allows you to set the pitch of the voice, achieving the purpose of changing the tone without altering the speed.

Setting voice change effects

This API is used to set voice change effects for human voice.

Note: The set effect will automatically expire after exiting the room. If you need the corresponding effect the next time you enter the room, you need to call this interface again to set it.

Setting voice reverb effects

This API is used to set reverb effects for human voice

Note: Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.

Play audio effect (deprecated)

Each audio effect has a unique ID. You can use the ID to start, pause or stop the effect.

If you want to play several effect in the same time, please set different ID for each effect. If you use same ID for different effect, the last one will take effect.

Set audio effect volume (deprecated)

Notice: This API will override the volume setting by setAllAudioEffectsVolume.

Stop audio effect (deprecated)

Stop all audio effects (deprecated)

Set all audio effects volume (deprecated)

Notice: This API will override the volume setting by setAudioEffectVolume.

Pause audio effect (deprecated)

Resume audio effect (deprecated)

Start network speed test (used before room entry)

As TRTC involves real-time audio/video transfer services very sensitive to the transfer latency, it has high requirements for network stability. For most users, if their network environments are below TRTC's minimum requirements, direct room entry will cause a very poor user experience. The recommended approach is to perform the network speed test before the user enters the room, so that a reminder can be displayed on the UI to prompt the user to switch to a better network (such as from Wi-Fi to 4G) first before room entry if the user's network is poor.

Notice:

1. The speed measurement process will incur a small amount of basic service fees, See Purchase Guide > Base Services.
2. Please perform the Network speed test before room entry, because if performed after room entry, the test will affect the normal audio/video transfer, and its result will be inaccurate due to interference in the room.
3. Only one network speed test task is allowed to run at the same time.

Stop network speed test

Start camera testing

After calling this API, an onFirstVideoFrame event will be emitted.

Notice: You can use the setCurrentCameraDevice API to switch between cameras during testing.

Stop camera testing

Start microphone testing

After calling this API, an onTestMicVolume event will be emitted.

This API is used to test whether the microphone functions properly. The microphone volume detected (value range: 0-100) is returned via onTestMicVolume event notification.

Stop microphone testing

Start speaker testing

After calling this API, an onTestSpeakerVolume event will be emitted.

This API is used to test whether the audio playback device functions properly by playing a specified audio file. If users can hear audio during testing, the device functions properly.

Stop speaker testing

Get SDK version information

Set log output level

Enable/Disable console log printing

Enable/Disable local log compression

If compression is enabled, the log size will significantly reduce, but 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 can be read directly in Notepad, but will take up more storage capacity.

Set local log storage path

You can use this API to change the default storage path of the SDK's local logs, which is as follows:

• Windows: C:/Users/[username]/AppData/Roaming/liteav/log, i.e., under %appdata%/liteav/log.
• macOS: under sandbox Documents/log.

Notice: Please be sure to call this API before all other APIs and make sure that the directory you specify exists and your application has read/write permissions of the directory.

Set log callback

Notice: If you set a log callback, the SDK's log information will be called back to you, and you need to handle it in your code. If callback is null, the log callback is canceled and the SDK will no longer call back the log.

Call experimental APIs

Notice: This API is used to enable some experimental feature

Set render mode(deprecated)

Set custom plugin options

You should call this API before calling addPlugin to set plugin options, otherwise the plugin will not be started and run.

Add plugin

Remove plugin

Set plugin event callback which will be called when plugin event occurs

This API can only be called once globally.

Get local media mixing manager

Note:

1. Only Windows Operating system supported currently.
   
2. The local media mixing manager only supports instances created by getTRTCShareInstance with isIPCMode set to true. Otherwise, it returns null.