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TRTCCloud

Last updated: 2022-10-28 10:12:56
    TRTCCloud

    TRTCCloud

    functions
    desc
    Request cross-room call
    Exit cross-room call
    Set dashboard margin
    Call experimental APIs
    Create room subinstance (for concurrent multi-room listen/watch)
    Terminate TRTCCloud instance (singleton mode)
    Terminate room subinstance
    Enable 3D spatial effect
    Enable volume reminder
    Enable custom audio capturing mode
    Enabling custom audio playback
    Enable/Disable custom video capturing mode
    Enable dual-channel encoding mode with big and small images
    Enable/Disable custom audio track
    enterRoom
    Enter room
    exitRoom
    Exit room
    Generate custom capturing timestamp
    Get the capturing volume of local audio
    Get sound effect management class (TXAudioEffectManager)
    Get the playback volume of remote audio
    Get beauty filter management class (TXBeautyManager)
    Getting playable audio data
    Get device management class (TXDeviceManager)
    Get SDK version information
    Mix custom audio track into SDK
    Pause/Resume playing back all remote users' audio streams
    Pause/Resume subscribing to all remote users' video streams
    Pause/Resume publishing local audio stream
    Pause/Resume publishing local video stream
    Pause/Resume playing back remote audio stream
    Pause/Resume subscribing to remote user's video stream
    Pause screen sharing
    Resume screen sharing
    Deliver captured audio data to SDK
    Use UDP channel to send custom message to all users in room
    Deliver captured video frames to SDK
    Use SEI channel to send custom message to all users in room
    Set the maximum 3D spatial attenuation range for userId's audio stream
    Set the capturing volume of local audio
    Set custom audio data callback
    Set the playback volume of remote audio
    Set audio route
    Set the callback format of original audio frames captured by local mic
    Enable/Disable console log printing
    Set subscription mode (which must be set before room entry for it to take effect)
    Set the adaptation mode of G-sensor
    Set the queue that drives the TRTCCloudDelegate event callback
    Set TRTC event callback
    Set the callback format of preprocessed local audio frames
    Set the rendering parameters of local video image
    Set video data callback for third-party beauty filters
    Set the callback of custom rendering for local video
    Enable/Disable local log compression
    Set local log storage path
    Set log output level
    Set log callback
    Set the publish volume and playback volume of mixed custom audio track
    Set the layout and transcoding parameters of On-Cloud MixTranscoding
    Set the callback format of audio frames to be played back by system
    Set network quality control parameters
    Set the parallel strategy of remote audio streams
    Set the audio playback volume of remote user
    Set the rendering mode of remote video image
    Set the callback of custom rendering for remote video
    Switch the big/small image of specified remote user
    Set the video encoding parameters of screen sharing (i.e., substream) (for desktop and mobile systems)
    Set the mirror mode of image output by encoder
    Set the encoding parameters of video encoder
    Set the direction of image output by video encoder
    Set placeholder image during local video pause
    Add watermark
    Create TRTCCloud instance (singleton mode)
    Display dashboard
    Screencapture video
    Start audio recording
    Enable local audio capturing and publishing
    Enable the preview image of local camera (mobile)
    Start local media recording
    Start publishing audio/video streams to non-Tencent Cloud CDN
    Publish a stream
    Start publishing audio/video streams to Tencent Cloud CSS CDN
    Subscribe to remote user's video stream and bind video rendering control
    Start screen sharing
    Start network speed test (used before room entry)
    Enable system audio capturing (for android system only)
    Stop subscribing to all remote users' video streams and release all rendering resources
    Stop audio recording
    Stop local audio capturing and publishing
    Stop camera preview
    Stop local media recording
    Stop publishing audio/video streams to non-Tencent Cloud CDN
    Stop publishing
    Stop publishing audio/video streams to Tencent Cloud CSS CDN
    Stop subscribing to remote user's video stream and release rendering control
    Stop screen sharing
    Stop network speed test
    Stop system audio capturing (for desktop systems and android system)
    Switch role(support permission credential)
    Switch room
    Update the preview image of local camera
    Modify publishing parameters
    Update the specified remote user's position for 3D spatial effect
    Update remote user's video rendering control
    Update self position and orientation for 3D spatial effect

    sharedInstance

    sharedInstance
    TRTCCloud sharedInstance
    (Context context)

    Create TRTCCloud instance (singleton mode)

    param
    desc
    context
    It is only applicable to the Android platform. The SDK internally converts it into the ApplicationContext of Android to call the Android system API.
    Note
    1. If you use delete ITRTCCloud* , a compilation error will occur. Please use destroyTRTCCloud to release the object pointer.
    2. On Windows, macOS, or iOS, please call the getTRTCShareInstance() API.
    3. On Android, please call the getTRTCShareInstance(void *context) API.

    destroySharedInstance

    destroySharedInstance

    Terminate TRTCCloud instance (singleton mode)

    setListener

    setListener
    void setListener
    (TRTCCloudListener listener)

    Set TRTC event callback

    You can use TRTCCloudListener to get various event notifications from the SDK, such as error codes, warning codes, and audio/video status parameters.
    param
    desc
    listener
    callback instance.

    setListenerHandler

    setListenerHandler
    void setListenerHandler
    (Handler listenerHandler)

    Set the queue that drives the TRTCCloudDelegate event callback

    If you do not specify a listenerHandler , the SDK will use MainQueue as the queue for driving TRTCCloudListener event callbacks by default.
    In other words, if you do not set the listenerHandler attribute, all callback functions in TRTCCloudListener will be driven by MainQueue .
    param
    desc
    listenerHandler
    
    Note
    If you specify a listenerHandler , please do not manipulate the UI in the TRTCCloudListener callback function; otherwise, thread safety issues will occur.

    enterRoom

    enterRoom
    void enterRoom
    (TRTCCloudDef.TRTCParams param
    
    int scene)

    Enter room

    All TRTC users need to enter a room before they can "publish" or "subscribe to" audio/video streams. "Publishing" refers to pushing their own streams to the cloud, and "subscribing to" refers to pulling the streams of other users in the room from the cloud.
    
    When calling this API, you need to specify your application scenario (TRTCAppScene) to get the best audio/video transfer experience. We provide the following four scenarios for your choice:
    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.
    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.
    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).
    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).
    
    After calling this API, you will receive the onEnterRoom(result) callback from TRTCCloudListener:
    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 TXLiteAVError for room entry failure.
    param
    desc
    param
    Room entry parameter, which is used to specify the user's identity, role, authentication credentials, and other information. For more information, please see TRTCParams.
    scene
    Application scenario, which is used to specify the use case. The same TRTCAppScene should be configured for all users in the same room.
    Note
    1. If scene is specified as TRTCAppSceneLIVE or 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 enterRoom and 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.

    exitRoom

    exitRoom

    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 use the onExitRoom() callback in TRTCCloudDelegate 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() callback, so as to avoid the problem of the camera or mic being occupied.

    switchRole

    switchRole
    void switchRole
    (int role)

    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.
    param
    desc
    role
    Role, which is anchor by default:
    TRTCRoleAnchor: anchor, who can publish their audio/video streams. Up to 50 anchors are allowed to publish streams at the same time in one room.
    TRTCRoleAudience: audience, who cannot publish their audio/video streams, but can only watch streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through switchRole. One room supports an audience of up to 100,000 concurrent online users.
    Note
    1. This API is only applicable to two scenarios: live streaming (TRTC_APP_SCENE_LIVE) and audio chat room (TRTC_APP_SCENE_VOICE_CHATROOM).
    2. If the scene you specify in enterRoom is TRTC_APP_SCENE_VIDEOCALL or TRTC_APP_SCENE_AUDIOCALL, please do not call this API.

    switchRole

    switchRole
    void switchRole
    (int role
    
    final String privateMapKey)

    Switch role(support permission credential)

    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.
    param
    desc
    privateMapKey
    Permission credential used for permission control. If you want only users with the specified userId values to enter a room or push streams, you need to use privateMapKey to restrict the permission.
    We recommend you use this parameter only if you have high security requirements. For more information, please see Enabling Advanced Permission Control.
    role
    Role, which is anchor by default:
    TRTCRoleAnchor: anchor, who can publish their audio/video streams. Up to 50 anchors are allowed to publish streams at the same time in one room.
    TRTCRoleAudience: audience, who cannot publish their audio/video streams, but can only watch streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through switchRole. One room supports an audience of up to 100,000 concurrent online users.
    Note
    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, please do not call this API.

    switchRoom

    switchRoom
    void switchRoom
    (final TRTCCloudDef.TRTCSwitchRoomConfig config)

    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 .
    The API call result will be called back through onSwitchRoom(errCode, errMsg) in TRTCCloudDelegate.
    param
    desc
    config
    Room parameter. For more information, please see TRTCSwitchRoomConfig.
    Note
    Due to the requirement for compatibility with legacy versions of the SDK, the config parameter contains both roomId and strRoomId parameters. You should pay special attention as detailed below when specifying these two parameters:
    1. If you decide to use strRoomId , then set roomId to 0. If both are specified, roomId will be used.
    2. All rooms need to use either strRoomId or roomId at the same time. They cannot be mixed; otherwise, there will be many unexpected bugs.

    ConnectOtherRoom

    ConnectOtherRoom
    void ConnectOtherRoom
    (String param)

    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 callback in TRTCCloudDelegate.
    
    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,true) event callbacks 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,true) event callbacks of anchor A; that is, all users in room "102" can subscribe to the audio/video streams of anchor A.
    
    
    
    For compatibility with subsequent extended fields for cross-room call, parameters in JSON format are used currently.
    
    Case 1: numeric room ID
    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.
    Below is the sample code:
    JSONObject jsonObj = new JSONObject();
    jsonObj.put("roomId", 102);
    jsonObj.put("userId", "userB");
    trtc.ConnectOtherRoom(jsonObj.toString());
    
    Case 2: string room ID
    If you use a string room ID, please be sure to replace the roomId in JSON with strRoomId , such as {"strRoomId": "102", "userId": "userB"}
    Below is the sample code:
    JSONObject jsonObj = new JSONObject();
    jsonObj.put("strRoomId", "102");
    jsonObj.put("userId", "userB");
    trtc.ConnectOtherRoom(jsonObj.toString());
    param
    desc
    param
    You need to pass in a string parameter in JSON format: roomId represents the room ID in numeric format, strRoomId represents the room ID in string format, and userId represents the user ID of the target anchor.

    DisconnectOtherRoom

    DisconnectOtherRoom

    Exit cross-room call

    The result will be returned through the onDisconnectOtherRoom() callback in TRTCCloudDelegate.

    setDefaultStreamRecvMode

    setDefaultStreamRecvMode
    void setDefaultStreamRecvMode
    (boolean autoRecvAudio
    
    boolean autoRecvVideo)

    Set subscription mode (which must be set before room entry for it to take effect)

    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.
    param
    desc
    autoRecvAudio
    true: automatic subscription to audio; false: manual subscription to audio by calling muteRemoteAudio(false) . Default value: true
    autoRecvVideo
    true: automatic subscription to video; false: manual subscription to video by calling startRemoteView . Default value: true
    Note
    1. The configuration takes effect only if this API is called before room entry (enterRoom).
    2. In the automatic subscription mode, if the user does not call startRemoteView to subscribe to the video stream after room entry, the SDK will automatically stop subscribing to the video stream in order to reduce the traffic consumption.

    createSubCloud

    createSubCloud

    Create room subinstance (for concurrent multi-room listen/watch)

    TRTCCloud was originally designed to work in the singleton mode, which limited the ability to watch concurrently in multiple rooms.
    By calling this API, you can create multiple TRTCCloud instances, so that you can enter multiple different rooms at the same time to listen/watch audio/video streams.
    
    However, it should be noted that because there are still only one camera and one mic available, you can exist as an "anchor" in only one TRTCCloud instance at any time; that is, you can only publish your audio/video streams in one TRTCCloud instance at any time.
    
    This feature is mainly used in the "super small class" use case in the online education scenario to break the limit that "only up to 50 users can publish their audio/video streams simultaneously in one TRTC room".
    
    Below is the sample code:
    TRTCCloud mainCloud = TRTCCloud.sharedInstance(mContext);
    mainCloud.enterRoom(params1, TRTCCloudDef.TRTC_APP_SCENE_LIVE);
    //...
    //Switch the role from "anchor" to "audience" in your own room
    mainCloud.switchRole(TRTCCloudDef.TRTCRoleAudience);
    mainCloud.muteLocalVideo(true);
    mainCloud.muteLocalAudio(true);
    //...
    //Use subcloud to enter another room and switch the role from "audience" to "anchor"
    TRTCCloud subCloud = mainCloud.createSubCloud();
    subCloud.enterRoom(params2, TRTCCloudDef.TRTC_APP_SCENE_LIVE);
    subCloud.switchRole(TRTCCloudDef.TRTCRoleAnchor);
    subCloud.muteLocalVideo(false);
    subCloud.muteLocalAudio(false);
    //...
    //Exit from new room and release it.
    subCloud.exitRoom();
    mainCloud.destroySubCloud(subCloud);
    Note
    The same user can enter multiple rooms with different roomId values by using the same userId .
    Two devices cannot use the same userId to enter the same room with a specified roomId .
    The same user can push a stream in only one TRTCCloud instance at any time. If streams are pushed simultaneously in different rooms, a status mess will be caused in the cloud, leading to various bugs.
    The TRTCCloud instance created by the createSubCloud API cannot call APIs related to the local audio/video in the subinstance, except switchRole , muteLocalVideo , and muteLocalAudio . To use APIs such as the beauty filter, please use the original TRTCCloud instance object.

    destroySubCloud

    destroySubCloud
    void destroySubCloud
    (final TRTCCloud subCloud)

    Terminate room subinstance

    param
    desc
    subCloud
    

    startPublishing

    startPublishing
    void startPublishing
    (final String streamId
    
    final int streamType)

    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.
    param
    desc
    streamId
    Custom stream ID.
    streamType
    Only TRTCVideoStreamTypeBig and TRTCVideoStreamTypeSub are supported.
    Note
    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.

    stopPublishing

    stopPublishing

    Stop publishing audio/video streams to Tencent Cloud CSS CDN

    startPublishCDNStream

    startPublishCDNStream
    void startPublishCDNStream
    (TRTCCloudDef.TRTCPublishCDNParam param)

    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.
    param
    desc
    param
    CDN relaying parameter. For more information, please see TRTCPublishCDNParam
    Note
    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.

    stopPublishCDNStream

    stopPublishCDNStream

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

    setMixTranscodingConfig

    setMixTranscodingConfig
    void setMixTranscodingConfig
    (TRTCCloudDef.TRTCTranscodingConfig config)

    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.
    
    
    param
    desc
    config
    If config is not empty, On-Cloud MixTranscoding will be started; otherwise, it will be stopped. For more information, please see TRTCTranscodingConfig.
    Note
    Notes on On-Cloud MixTranscoding:
    Mixed-stream transcoding is a chargeable function, calling the interface will incur cloud-based mixed-stream transcoding fees, see Billing of On-Cloud MixTranscoding.
    If the user calling this API does not set streamId in the config parameter, TRTC will mix the multiple channels of images in the room into the audio/video streams corresponding to the current user, i.e., A + B => A.
    If the user calling this API sets streamId in the config parameter, TRTC will mix the multiple channels of images in the room into the specified streamId , i.e., A + B => streamId.
    Please note that if you are still in the room but do not need mixtranscoding anymore, be sure to call this API again and leave config empty to cancel it; otherwise, additional fees may be incurred.
    Please rest assured that TRTC will automatically cancel the mixtranscoding status upon room exit.

    startPublishMediaStream

    startPublishMediaStream
    void startPublishMediaStream
    (TRTCCloudDef.TRTCPublishTarget target
    
    TRTCCloudDef.TRTCStreamEncoderParam params
    
    TRTCCloudDef.TRTCStreamMixingConfig config)

    Publish a stream

    After this API is called, the TRTC server will relay the stream of the local user to a CDN (after transcoding or without transcoding), or transcode and publish the stream to a TRTC room.
    You can use the TRTCPublishMode parameter in TRTCPublishTarget to specify the publishing mode.
    param
    desc
    config
    The On-Cloud MixTranscoding settings. This parameter is invalid in the relay-to-CDN mode. It is required if you transcode and publish the stream to a CDN or to a TRTC room. For details, see TRTCStreamMixingConfig.
    params
    The encoding settings. This parameter is required if you transcode and publish the stream to a CDN or to a TRTC room. If you relay to a CDN without transcoding, to improve the relaying stability and playback compatibility, we also recommend you set this parameter. For details, see TRTCStreamEncoderParam.
    target
    The publishing destination. You can relay the stream to a CDN (after transcoding or without transcoding) or transcode and publish the stream to a TRTC room. For details, see TRTCPublishTarget.
    Note
    1. The SDK will send a task ID to you via the onStartPublishMediaStream callback.
    2. You can start a publishing task only once and cannot initiate two tasks that use the same publishing mode and publishing cdn url. Note the task ID returned, which you need to pass to updatePublishMediaStream to modify the publishing parameters or stopPublishMediaStream to stop the task.
    3. You can specify up to 10 CDN URLs in target . You will be charged only once for transcoding even if you relay to multiple CDNs.
    4. To avoid causing errors, do not specify the same URLs for different publishing tasks executed at the same time. We recommend you add "sdkappid_roomid_userid_main" to URLs to distinguish them from one another and avoid application conflicts.

    updatePublishMediaStream

    updatePublishMediaStream
    void updatePublishMediaStream
    (final String taskId
    
    TRTCCloudDef.TRTCPublishTarget target
    
    TRTCCloudDef.TRTCStreamEncoderParam params
    
    TRTCCloudDef.TRTCStreamMixingConfig config)

    Modify publishing parameters

    You can use this API to change the parameters of a publishing task initiated by startPublishMediaStream.
    param
    desc
    config
    The On-Cloud MixTranscoding settings. This parameter is invalid in the relay-to-CDN mode. It is required if you transcode and publish the stream to a CDN or to a TRTC room. For details, see TRTCStreamMixingConfig.
    params
    The encoding settings. This parameter is required if you transcode and publish the stream to a CDN or to a TRTC room. If you relay to a CDN without transcoding, to improve the relaying stability and playback compatibility, we recommend you set this parameter. For details, see TRTCStreamEncoderParam.
    target
    The publishing destination. You can relay the stream to a CDN (after transcoding or without transcoding) or transcode and publish the stream to a TRTC room. For details, see TRTCPublishTarget.
    taskId
    The task ID returned to you via the onStartPublishMediaStream callback.
    Note
    1. You can use this API to add or remove CDN URLs to publish to (you can publish to up to 10 CDNs at a time). To avoid causing errors, do not specify the same URLs for different tasks executed at the same time.
    2. You can use this API to switch a relaying task to transcoding or vice versa. For example, in cross-room communication, you can first call startPublishMediaStream to relay to a CDN. When the anchor requests cross-room communication, call this API, passing in the task ID to switch the relaying task to a transcoding task. This can ensure that the live stream and CDN playback are not interrupted (you need to keep the encoding parameters consistent).
    3. You can not switch output between "only audio" 、 "only video" and "audio and video" for the same task.

    stopPublishMediaStream

    stopPublishMediaStream
    void stopPublishMediaStream
    (final String taskId)

    Stop publishing

    You can use this API to stop a task initiated by startPublishMediaStream.
    param
    desc
    taskId
    The task ID returned to you via the onStartPublishMediaStream callback.
    Note
    1. If the task ID is not saved to your backend, you can call startPublishMediaStream again when an anchor re-enters the room after abnormal exit. The publishing will fail, but the TRTC backend will return the task ID to you.
    2. If taskId is left empty, the TRTC backend will end all tasks initiated by startPublishMediaStream. You can leave it empty if you have started only one task or want to stop all publishing tasks.

    startLocalPreview

    startLocalPreview
    void startLocalPreview
    (boolean frontCamera
    
    TXCloudVideoView view)

    Enable the preview image of local camera (mobile)

    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 onCameraDidReady callback in TRTCCloudDelegate.
    param
    desc
    frontCamera
    true: front camera; false: rear camera
    view
    Control that carries the video image
    Note
    If you want to preview the camera image and adjust the beauty filter parameters through BeautyManager before going live, you can:
    Scheme 1. Call startLocalPreview before calling enterRoom
    Scheme 2. Call startLocalPreview and muteLocalVideo(true) after calling enterRoom

    updateLocalView

    updateLocalView
    void updateLocalView
    (TXCloudVideoView view)

    Update the preview image of local camera

    stopLocalPreview

    stopLocalPreview

    Stop camera preview

    muteLocalVideo

    muteLocalVideo
    void muteLocalVideo
    (int streamType
    
    boolean mute)

    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, false) callback notification.
    After local video publishing is resumed, other members in the same room will receive the onUserVideoAvailable(userId, true) callback notification.
    param
    desc
    mute
    true: pause; false: resume
    streamType
    Specify for which video stream to pause (or resume). Only TRTCVideoStreamTypeBig and TRTCVideoStreamTypeSub are supported

    setVideoMuteImage

    setVideoMuteImage
    void setVideoMuteImage
    (Bitmap image
    
    int fps)

    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.
    param
    desc
    fps
    Frame rate of the placeholder image. Minimum value: 5. Maximum value: 10. Default value: 5
    image
    Placeholder image. A null value means that no more video stream data will be sent after muteLocalVideo . The default value is null.

    startRemoteView

    startRemoteView
    void startRemoteView
    (String userId
    
    int streamType
    
    TXCloudVideoView view)

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

    Calling this API allows the SDK to 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 through setRemoteRenderParams.
    If you already know the userId of a user who has a video stream in the room, you can directly call startRemoteView to subscribe to the user's video image.
    If you don't know which users in the room are publishing video streams, you can wait for the notification from onUserVideoAvailable after enterRoom .
    
    Calling this API only starts pulling the video stream, and the image needs to be loaded and buffered at this time. After the buffering is completed, you will receive a notification from onFirstVideoFrame.
    param
    desc
    streamType
    Video stream type of the userId specified for watching:
    HD big image: TRTCVideoStreamTypeBig
    Smooth small image: TRTCVideoStreamTypeSmall (the remote user should enable dual-channel encoding through enableEncSmallVideoStream for this parameter to take effect)
    Substream image (usually used for screen sharing): TRTCVideoStreamTypeSub
    userId
    ID of the specified remote user
    view
    Rendering control that carries the video image
    Note
    The following requires your attention:
    1. The SDK supports watching the big image and substream image or small image and substream image of a userId at the same time, but does not support watching the big image and small image at the same time.
    2. Only when the specified userId enables dual-channel encoding through enableEncSmallVideoStream can the user's small image be viewed.
    3. If the small image of the specified userId does not exist, the SDK will switch to the big image of the user by default.

    updateRemoteView

    updateRemoteView
    void updateRemoteView
    (String userId
    
    int streamType
    
    TXCloudVideoView view)

    Update remote user's video rendering control

    This API can be used to update the rendering control of the remote video image. It is often used in interactive scenarios where the display area needs to be switched.
    param
    desc
    streamType
    Type of the stream for which to set the preview window (only TRTCVideoStreamTypeBig and TRTCVideoStreamTypeSub are supported)
    userId
    ID of the specified remote user
    view
    Control that carries the video image

    stopRemoteView

    stopRemoteView
    void stopRemoteView
    (String userId
    
    int streamType)

    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.
    param
    desc
    streamType
    Video stream type of the userId specified for watching:
    HD big image: TRTCVideoStreamTypeBig
    Smooth small image: TRTCVideoStreamTypeSmall
    Substream image (usually used for screen sharing): TRTCVideoStreamTypeSub
    userId
    ID of the specified remote user

    stopAllRemoteView

    stopAllRemoteView

    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.
    Note
    If a substream image (screen sharing) is being displayed, it will also be stopped.

    muteRemoteVideoStream

    muteRemoteVideoStream
    void muteRemoteVideoStream
    (String userId
    
    int streamType
    
    boolean mute)

    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.
    param
    desc
    mute
    Whether to pause receiving
    streamType
    Specify for which video stream to pause (or resume):
    HD big image: TRTCVideoStreamTypeBig
    Smooth small image: TRTCVideoStreamTypeSmall
    Substream image (usually used for screen sharing): TRTCVideoStreamTypeSub
    userId
    ID of the specified remote user
    Note
    This API can be called before room entry (enterRoom), and the pause status will be reset after room exit (exitRoom).

    muteAllRemoteVideoStreams

    muteAllRemoteVideoStreams
    void muteAllRemoteVideoStreams
    (boolean mute)

    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.
    param
    desc
    mute
    Whether to pause receiving
    Note
    This API can be called before room entry (enterRoom), and the pause status will be reset after room exit (exitRoom).

    setVideoEncoderParam

    setVideoEncoderParam
    void setVideoEncoderParam
    (TRTCCloudDef.TRTCVideoEncParam param)

    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.
    param
    desc
    param
    It is used to set relevant parameters for the video encoder. For more information, please see TRTCVideoEncParam.

    setNetworkQosParam

    setNetworkQosParam
    void setNetworkQosParam
    (TRTCCloudDef.TRTCNetworkQosParam param)

    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".
    param
    desc
    param
    It is used to set relevant parameters for network quality control. For details, please refer to TRTCNetworkQosParam.

    setLocalRenderParams

    setLocalRenderParams
    void setLocalRenderParams
    (TRTCCloudDef.TRTCRenderParams renderParams)

    Set the rendering parameters of local video image

    The parameters that can be set include video image rotation angle, fill mode, and mirror mode.
    param
    desc
    params
    Video image rendering parameters. For more information, please see TRTCRenderParams.

    setRemoteRenderParams

    setRemoteRenderParams
    void setRemoteRenderParams
    (String userId
    
    int streamType
    
    TRTCCloudDef.TRTCRenderParams renderParams)

    Set the rendering mode of remote video image

    The parameters that can be set include video image rotation angle, fill mode, and mirror mode.
    param
    desc
    params
    Video image rendering parameters. For more information, please see TRTCRenderParams.
    streamType
    It can be set to the primary stream image (TRTCVideoStreamTypeBig) or substream image (TRTCVideoStreamTypeSub).
    userId
    ID of the specified remote user

    setVideoEncoderRotation

    setVideoEncoderRotation
    void setVideoEncoderRotation
    (int rotation)

    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.
    param
    desc
    rotation
    Currently, rotation angles of 0 and 180 degrees are supported. Default value: TRTCVideoRotation_0 (no rotation)

    setVideoEncoderMirror

    setVideoEncoderMirror
    void setVideoEncoderMirror
    (boolean mirror)

    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).
    param
    desc
    mirror
    Whether to enable remote mirror mode. true: yes; false: no. Default value: false

    setGSensorMode

    setGSensorMode
    void setGSensorMode
    (int mode)

    Set the adaptation mode of G-sensor

    You can achieve the following user-friendly interactive experience through this API:
    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 let the SDK automatically adjust the rotation direction of the local video image and the image output by the encoder according to the direction of the device's gyroscope, so that remote viewers can see the image in the normal direction.
    param
    desc
    mode
    G-sensor mode. For more information, please see TRTCGSensorMode. Default value: TRTCGSensorMode_UIAutoLayout

    enableEncSmallVideoStream

    enableEncSmallVideoStream
    int enableEncSmallVideoStream
    (boolean enable
    
    TRTCCloudDef.TRTCVideoEncParam smallVideoEncParam)

    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.
    param
    desc
    enable
    Whether to enable small image encoding. Default value: false
    smallVideoEncParam
    Video parameters of small image stream
    Note
    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.

    setRemoteVideoStreamType

    setRemoteVideoStreamType
    int setRemoteVideoStreamType
    (String userId
    
    int streamType)

    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.
    param
    desc
    streamType
    Video stream type, i.e., big image or small image. Default value: big image
    userId
    ID of the specified remote user
    Note
    To implement this feature, the target user must have enabled the dual-channel encoding mode through enableEncSmallVideoStream; otherwise, this API will not work.

    snapshotVideo

    snapshotVideo
    void snapshotVideo
    (String userId
    
    int streamType
    
    TRTCCloudListener.TRTCSnapshotListener listener)

    Screencapture video

    You can use this API to screencapture the local video image or the primary stream image and substream (screen sharing) image of a remote user.
    param
    desc
    sourceType
    Video image source, which can be the video stream image (TRTCSnapshotSourceTypeStream, generally in higher definition) or the video rendering image (TRTCSnapshotSourceTypeView)
    streamType
    Video stream type, which can be the primary stream image (TRTCVideoStreamTypeBig, generally for camera) or substream image (TRTCVideoStreamTypeSub, generally for screen sharing)
    userId
    User ID. A null value indicates to screencapture the local video.
    Note
    On Windows, only video image from the TRTCSnapshotSourceTypeStream source can be screencaptured currently.

    startLocalAudio

    startLocalAudio
    void startLocalAudio
    (int quality)

    Enable local audio capturing and publishing

    The SDK does not enable the mic by default. When a user wants to publish the local audio, the user needs to call this API to enable mic 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, true) notification.
    param
    desc
    quality
    Sound quality
    TRTCAudioQualitySpeech - Smooth: sample rate: 16 kHz; mono channel; audio bitrate: 16 Kbps. This is suitable for audio call scenarios, such as online meeting and audio call.
    TRTCAudioQualityDefault - Default: sample rate: 48 kHz; mono channel; audio bitrate: 50 Kbps. This is the default sound quality of the SDK and recommended if there are no special requirements.
    TRTCAudioQualityMusic - HD: sample rate: 48 kHz; dual channel + full band; audio bitrate: 128 Kbps. This is suitable for scenarios where Hi-Fi music transfer is required, such as online karaoke and music live streaming.
    Note
    This API will check the mic permission. If the current application does not have permission to use the mic, the SDK will automatically ask the user to grant the mic permission.

    stopLocalAudio

    stopLocalAudio

    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) notification.

    muteLocalAudio

    muteLocalAudio
    void muteLocalAudio
    (boolean mute)

    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.
    param
    desc
    mute
    true: mute; false: unmute

    muteRemoteAudio

    muteRemoteAudio
    void muteRemoteAudio
    (String userId
    
    boolean mute)

    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.
    param
    desc
    mute
    true: mute; false: unmute
    userId
    ID of the specified remote user
    Note
    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).

    muteAllRemoteAudio

    muteAllRemoteAudio
    void muteAllRemoteAudio
    (boolean mute)

    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.
    param
    desc
    mute
    true: mute; false: unmute
    Note
    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).

    setAudioRoute

    setAudioRoute
    void setAudioRoute
    (int route)

    Set audio route

    Setting "audio route" is to determine whether the sound is played back from the speaker or receiver of a mobile device; therefore, this API is only applicable to mobile devices such as phones.
    
    Generally, a phone has two speakers: one is the receiver at the top, and the other is the stereo speaker at the bottom.
    If audio route is set to the receiver, the volume is relatively low, and the sound can be heard clearly only when the phone is put near the ear. This mode has a high level of privacy and is suitable for answering calls.
    If audio route is set to the speaker, the volume is relatively high, so there is no need to put the phone near the ear. Therefore, this mode can implement the "hands-free" feature.
    param
    desc
    route
    Audio route, i.e., whether the audio is output by speaker or receiver. Default value: TRTCAudioModeSpeakerphone

    setRemoteAudioVolume

    setRemoteAudioVolume
    void setRemoteAudioVolume
    (String userId
    
    int volume)

    Set the audio playback volume of remote user

    You can mute the audio of a remote user through setRemoteAudioVolume(userId, 0) .
    param
    desc
    userId
    ID of the specified remote user
    volume
    Volume. 100 is the original volume. Value range: [0,150]. Default value: 100
    Note
    If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.

    setAudioCaptureVolume

    setAudioCaptureVolume
    void setAudioCaptureVolume
    (int volume)

    Set the capturing volume of local audio

    param
    desc
    volume
    Volume. 100 is the original volume. Value range: [0,150]. Default value: 100
    Note
    If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.

    getAudioCaptureVolume

    getAudioCaptureVolume

    Get the capturing volume of local audio

    setAudioPlayoutVolume

    setAudioPlayoutVolume
    void setAudioPlayoutVolume
    (int volume)

    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.
    param
    desc
    volume
    Volume. 100 is the original volume. Value range: [0,150]. Default value: 100
    Note
    If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.

    getAudioPlayoutVolume

    getAudioPlayoutVolume

    Get the playback volume of remote audio

    enableAudioVolumeEvaluation

    enableAudioVolumeEvaluation
    void enableAudioVolumeEvaluation
    (int interval
    
    boolean enable_vad)

    Enable volume reminder

    After this feature is enabled, the SDK will return the volume of local user who sends stream and remote users in the onUserVoiceVolume callback of TRTCCloudDelegate.
    param
    desc
    enable_vad
    true: Enable the voice detection of the local user false: Disable the voice detection of the local user
    interval
    Set the interval in ms for triggering the onUserVoiceVolume callback. The minimum interval is 100 ms. If the value is smaller than or equal to 0, the callback will be disabled. We recommend you set this parameter to 300 ms.
    Note
    To enable this feature, call this API before calling startLocalAudio .

    startAudioRecording

    startAudioRecording
    int startAudioRecording
    (TRTCCloudDef.TRTCAudioRecordingParams param)

    Start audio recording

    After you call this API, the SDK will selectively record local and remote audio streams (such as local audio, remote audio, background music, and sound effects) into a local file.
    This API works when called either before or after room entry. If a recording task has not been stopped through stopAudioRecording before room exit, it will be automatically stopped after room exit.
    param
    desc
    param
    Recording parameter. For more information, please see TRTCAudioRecordingParams

    stopAudioRecording

    stopAudioRecording

    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.

    startLocalRecording

    startLocalRecording
    void startLocalRecording
    (TRTCCloudDef.TRTCLocalRecordingParams params)

    Start local media recording

    This API records the audio/video content during live streaming into a local file.
    param
    desc
    params
    Recording parameter. For more information, please see TRTCLocalRecordingParams

    stopLocalRecording

    stopLocalRecording

    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.

    setRemoteAudioParallelParams

    setRemoteAudioParallelParams
    void setRemoteAudioParallelParams
    (TRTCCloudDef.TRTCAudioParallelParams params)

    Set the parallel strategy of remote audio streams

    For room with many speakers.
    param
    desc
    params
    Audio parallel parameter. For more information, please see TRTCAudioParallelParams

    enable3DSpatialAudioEffect

    enable3DSpatialAudioEffect
    void enable3DSpatialAudioEffect
    (boolean enabled)

    Enable 3D spatial effect

    Enable 3D spatial effect. Note that TRTCAudioQualitySpeech smooth or TRTCAudioQualityDefault default audio quality should be used.
    param
    desc
    enabled
    Whether to enable 3D spatial effect. It’s disabled by default.

    updateSelf3DSpatialPosition

    updateSelf3DSpatialPosition
    void updateSelf3DSpatialPosition
    (int[] position
    
    float[] axisForward
    
    float[] axisRight
    
    float[] axisUp)

    Update self position and orientation for 3D spatial effect

    Update self position and orientation in the world coordinate system. The SDK will calculate the relative position between self and the remote users according to the parameters of this method, and then render the spatial sound effect. Note that the length of array should be 3.
    param
    desc
    axisForward
    The unit vector of the forward axis of user coordinate system. The three values represent the forward, right and up coordinate values in turn.
    axisRight
    The unit vector of the right axis of user coordinate system. The three values represent the forward, right and up coordinate values in turn.
    axisUp
    The unit vector of the up axis of user coordinate system. The three values represent the forward, right and up coordinate values in turn.
    position
    The coordinate of self in the world coordinate system. The three values represent the forward, right and up coordinate values in turn.
    Note
    Please limit the calling frequency appropriately. It's recommended that the interval between two operations be at least 100ms.

    updateRemote3DSpatialPosition

    updateRemote3DSpatialPosition
    void updateRemote3DSpatialPosition
    (String userId
    
    int[] position)

    Update the specified remote user's position for 3D spatial effect

    Update the specified remote user's position in the world coordinate system. The SDK will calculate the relative position between self and the remote users according to the parameters of this method, and then render the spatial sound effect. Note that the length of array should be 3.
    param
    desc
    position
    The coordinate of self in the world coordinate system. The three values represent the forward, right and up coordinate values in turn.
    userId
    ID of the specified remote user.
    Note
    Please limit the calling frequency appropriately. It's recommended that the interval between two operations of the same remote user be at least 100ms.

    set3DSpatialReceivingRange

    set3DSpatialReceivingRange
    void set3DSpatialReceivingRange
    (String userId
    
    int range)

    Set the maximum 3D spatial attenuation range for userId's audio stream

    After set the range, the specified user's audio stream will attenuate to zero within the range.
    param
    desc
    range
    Maximum attenuation range of the audio stream.
    userId
    ID of the specified user.

    getDeviceManager

    getDeviceManager

    Get device management class (TXDeviceManager)

    getBeautyManager

    getBeautyManager

    Get beauty filter management class (TXBeautyManager)

    You can use the following features with beauty filter management:
    Set beauty effects such as "skin smoothing", "brightening", and "rosy skin".
    Set face adjustment effects such as "eye enlarging", "face slimming", "chin slimming", "chin lengthening/shortening", "face shortening", "nose narrowing", "eye brightening", "teeth whitening", "eye bag removal", "wrinkle removal", and "smile line removal".
    Set face adjustment effects such as "hairline", "eye distance", "eye corners", "mouth shape", "nose wing", "nose position", "lip thickness", and "face shape".
    Set makeup effects such as "eye shadow" and "blush".
    Set animated effects such as animated sticker and facial pendant.

    setWatermark

    setWatermark
    void setWatermark
    (Bitmap image
    
    int streamType
    
    float x
    
    float y
    
    float width)

    Add watermark

    The watermark position is determined by the rect parameter, which is a quadruple in the format of (x, y, width, height).
    x: X coordinate of watermark, which is a floating-point number between 0 and 1.
    y: Y coordinate of watermark, which is a floating-point number between 0 and 1.
    width: width of watermark, which is a floating-point number between 0 and 1.
    height: it does not need to be set. The SDK will automatically calculate it according to the watermark image's aspect ratio.
    
    Sample parameter:
    If the encoding resolution of the current video is 540x960, and the rect parameter is set to (0.1, 0.1, 0.2, 0.0),
    then the coordinates of the top-left point of the watermark will be (540 * 0.1, 960 * 0.1), i.e., (54, 96), the watermark width will be 540 * 0.2 = 108 px, and the watermark height will be calculated automatically by the SDK based on the watermark image's aspect ratio.
    param
    desc
    image
    Watermark image, which must be a PNG image with transparent background
    rect
    Unified coordinates of the watermark relative to the encoded resolution. Value range of x , y , width , and height : 0–1.
    streamType
    Specify for which image to set the watermark. For more information, please see TRTCVideoStreamType.
    Note
    If you want to set watermarks for both the primary image (generally for the camera) and the substream image (generally for screen sharing), you need to call this API twice with streamType set to different values.

    getAudioEffectManager

    getAudioEffectManager

    Get sound effect management class (TXAudioEffectManager)

    TXAudioEffectManager is a sound effect management API, through which you can implement the following features:
    Background music: both online music and local music can be played back with various features such as speed adjustment, pitch adjustment, original voice, accompaniment, and loop.
    In-ear monitoring: the sound captured by the mic is played back in the headphones in real time, which is generally used for music live streaming.
    Reverb effect: karaoke room, small room, big hall, deep, resonant, and other effects.
    Voice changing effect: young girl, middle-aged man, heavy metal, and other effects.
    Short sound effect: short sound effect files such as applause and laughter are supported (for files less than 10 seconds in length, please set the isShortFile parameter to true ).

    startSystemAudioLoopback

    startSystemAudioLoopback

    Enable system audio capturing (for android system only)

    This API captures audio data from another app 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 another app.
    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.
    Note
    1. This interface only works on Android API 29 and above.
    2. You need to use this interface to enable system sound capture first, and it will take effect only when you call startScreenCapture to enable screen sharing.
    3. You need to add a foreground service to ensure that the system sound capture is not silenced, and set android:foregroundServiceType="mediaProjection".
    4. The SDK only capture audio of applications that satisfies the capture strategy and audio usage. Currently, the audio usage captured by the SDK includes USAGE_MEDIA, USAGE_GAME。

    stopSystemAudioLoopback

    stopSystemAudioLoopback

    Stop system audio capturing (for desktop systems and android system)

    startScreenCapture

    startScreenCapture
    void startScreenCapture
    (int streamType
    
    TRTCCloudDef.TRTCVideoEncParam encParams
    
    TRTCCloudDef.TRTCScreenShareParams shareParams)

    Start screen sharing

    This API supports capturing the screen of the entire Android system, which can implement system-wide screen sharing similar to VooV Meeting.
    
    For more information, please see Android
    
    Video encoding parameters recommended for screen sharing on Android (TRTCVideoEncParam):
    Resolution (videoResolution): 1280x720
    Frame rate (videoFps): 10 fps
    Bitrate (videoBitrate): 1200 Kbps
    Resolution adaption (enableAdjustRes): false
    param
    desc
    encParams
    Encoding parameters. For more information, please see TRTCCloudDef#TRTCVideoEncParam. If encParams is set to null , the SDK will automatically use the previously set encoding parameter.
    shareParams
    For more information, please see TRTCCloudDef#TRTCScreenShareParams. You can use the floatingView parameter to pop up a floating window (you can also use Android's WindowManager parameter to configure automatic pop-up).

    stopScreenCapture

    stopScreenCapture

    Stop screen sharing

    pauseScreenCapture

    pauseScreenCapture

    Pause screen sharing

    resumeScreenCapture

    resumeScreenCapture

    Resume screen sharing

    setSubStreamEncoderParam

    setSubStreamEncoderParam
    void setSubStreamEncoderParam
    (TRTCCloudDef.TRTCVideoEncParam param)

    Set the video encoding parameters of screen sharing (i.e., substream) (for desktop and mobile systems)

    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).
    param
    desc
    param
    Substream encoding parameters. For more information, please see TRTCVideoEncParam.
    Note
    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.

    enableCustomVideoCapture

    enableCustomVideoCapture
    void enableCustomVideoCapture
    (int streamType
    
    boolean enable)

    Enable/Disable custom video capturing mode

    After this mode is enabled, the SDK will not run the original video capturing process (i.e., stopping camera data capturing and beauty filter operations) and will retain only the video encoding and sending capabilities.
    You need to use sendCustomVideoData to continuously insert the captured video image into the SDK.
    param
    desc
    enable
    Whether to enable. Default value: false
    streamType
    Specify video stream type (TRTCVideoStreamTypeBig: HD big image; TRTCVideoStreamTypeSub: substream image).

    sendCustomVideoData

    sendCustomVideoData
    void sendCustomVideoData
    (int streamType
    
    TRTCCloudDef.TRTCVideoFrame frame)

    Deliver captured video frames to SDK

    You can use this API to deliver video frames you capture to the SDK, and the SDK will encode and transfer them through its own network module.
    There are two delivery schemes for Android:
    Memory-based delivery scheme: its connection is easy but its performance is poor, so it is not suitable for scenarios with high resolution.
    Video memory-based delivery scheme: its connection requires certain knowledge in OpenGL, but its performance is good. For resolution higher than 640x360, please use this scheme.
    
    For more information, please see Custom Capturing and Rendering.
    param
    desc
    frame
    Video data. If the memory-based delivery scheme is used, please set the data field; if the video memory-based delivery scheme is used, please set the TRTCTexture field. For more information, please see com::tencent::trtc::TRTCCloudDef::TRTCVideoFrame TRTCVideoFrame.
    streamType
    Specify video stream type (TRTCVideoStreamTypeBig: HD big image; TRTCVideoStreamTypeSub: substream image).
    Note
    1. We recommend you call the generateCustomPTS API to get the timestamp value of a video frame immediately after capturing it, so as to achieve the best audio/video sync effect.
    2. The video frame rate eventually encoded by the SDK is not determined by the frequency at which you call this API, but by the FPS you set in setVideoEncoderParam.
    3. Please try to keep the calling interval of this API even; otherwise, problems will be caused, such as unstable output frame rate of the encoder or out-of-sync audio/video.

    enableCustomAudioCapture

    enableCustomAudioCapture
    void enableCustomAudioCapture
    (boolean enable)

    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.
    param
    desc
    enable
    Whether to enable. Default value: false
    Note
    As acoustic echo cancellation (AEC) requires strict control over the audio capturing and playback time, after custom audio capturing is enabled, AEC may fail.

    sendCustomAudioData

    sendCustomAudioData
    void sendCustomAudioData
    (TRTCCloudDef.TRTCAudioFrame frame)

    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.
    
    For more information, please see Custom Capturing and Rendering.
    param
    desc
    frame
    Audio data
    Note
    Please call this API accurately at intervals of the frame length; otherwise, sound lag may occur due to uneven data delivery intervals.

    enableMixExternalAudioFrame

    enableMixExternalAudioFrame
    void enableMixExternalAudioFrame
    (boolean enablePublish
    
    boolean enablePlayout)

    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.
    param
    desc
    enablePlayout
    Whether the mixed audio track should be played back locally. Default value: false
    enablePublish
    Whether the mixed audio track should be played back remotely. Default value: false
    Note
    If you specify both enablePublish and enablePlayout as false , the custom audio track will be completely closed.

    mixExternalAudioFrame

    mixExternalAudioFrame
    int mixExternalAudioFrame
    (TRTCCloudDef.TRTCAudioFrame frame)

    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.
    param
    desc
    frame
    Audio data

    setMixExternalAudioVolume

    setMixExternalAudioVolume
    void setMixExternalAudioVolume
    (int publishVolume
    
    int playoutVolume)

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

    param
    desc
    playoutVolume
    set the play volume,from 0 to 100, -1 means no change
    publishVolume
    set the publish volume,from 0 to 100, -1 means no change

    generateCustomPTS

    generateCustomPTS

    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 sendCustomVideoData or 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 sendCustomVideoData or sendCustomAudioData for delivery, assign the PTS timestamp recorded when the frame was captured to the timestamp field in TRTCVideoFrame or TRTCAudioFrame.

    setLocalVideoProcessListener

    setLocalVideoProcessListener
    int setLocalVideoProcessListener
    (int pixelFormat
    
    int bufferType
    
    TRTCCloudListener.TRTCVideoFrameListener listener)

    Set video data callback for third-party beauty filters

    After this callback is set, the SDK will call back the captured video frames through the listener you set and use them for further processing by a third-party beauty filter component. Then, the SDK will encode and send the processed video frames.
    param
    desc
    bufferType
    Specify the format of the data called back. Currently, it supports:
    TRTC_VIDEO_BUFFER_TYPE_TEXTURE: suitable when pixelFormat is set to TRTC_VIDEO_PIXEL_FORMAT_Texture_2D.
    TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER: suitable when pixelFormat is set to TRTC_VIDEO_PIXEL_FORMAT_I420.
    TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY: suitable when pixelFormat is set to TRTC_VIDEO_PIXEL_FORMAT_I420.
    listener
    Custom preprocessing callback. For more information, please see TRTCVideoFrameListener
    pixelFormat
    Specify the format of the pixel called back. Currently, it supports:
    TRTC_VIDEO_PIXEL_FORMAT_Texture_2D: video memory-based texture scheme.
    TRTC_VIDEO_PIXEL_FORMAT_I420: memory-based data scheme.

    setLocalVideoRenderListener

    setLocalVideoRenderListener
    int setLocalVideoRenderListener
    (int pixelFormat
    
    int bufferType
    
    TRTCCloudListener.TRTCVideoRenderListener listener)

    Set the callback of custom rendering for local video

    After this callback is set, the SDK will skip its own rendering process and call back the captured data. Therefore, you need to complete image rendering on your own.
    pixelFormat specifies the format of the data called back. Currently, Texture2D, I420, and RGBA formats are supported.
    bufferType specifies the buffer type. BYTE_BUFFER is suitable for the JNI layer, while BYTE_ARRAY can be used in direct operations at the Java layer.
    
    For more information, please see Custom Capturing and Rendering.
    param
    desc
    bufferType
    Specify the data structure of the video frame:
    TRTC_VIDEO_BUFFER_TYPE_TEXTURE: suitable when pixelFormat is set to TRTC_VIDEO_PIXEL_FORMAT_Texture_2D.
    TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER: suitable when pixelFormat is set to TRTC_VIDEO_PIXEL_FORMAT_I420 or TRTC_VIDEO_PIXEL_FORMAT_RGBA.
    TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY: suitable when pixelFormat is set to TRTC_VIDEO_PIXEL_FORMAT_I420 or TRTC_VIDEO_PIXEL_FORMAT_RGBA.
    listener
    Callback of custom video rendering. The callback is returned once for each video frame
    pixelFormat
    Specify the format of the video frame, such as:
    TRTC_VIDEO_PIXEL_FORMAT_Texture_2D: OpenGL texture format, which is suitable for GPU processing and has a high processing efficiency.
    TRTC_VIDEO_PIXEL_FORMAT_I420: standard I420 format, which is suitable for CPU processing and has a poor processing efficiency.
    TRTC_VIDEO_PIXEL_FORMAT_RGBA: RGBA format, which is suitable for CPU processing and has a poor processing efficiency.

    setRemoteVideoRenderListener

    setRemoteVideoRenderListener
    int setRemoteVideoRenderListener
    (String userId
    
    int pixelFormat
    
    int bufferType
    
    TRTCCloudListener.TRTCVideoRenderListener listener)

    Set the callback of custom rendering for remote video

    After this callback is set, the SDK will skip its own rendering process and call back the captured data. Therefore, you need to complete image rendering on your own.
    pixelFormat specifies the format of the called back data, such as NV12, I420, and 32BGRA.
    bufferType specifies the buffer type. PixelBuffer has the highest efficiency, while NSData makes the SDK perform a memory conversion internally, which will result in extra performance loss.
    
    For more information, please see Custom Capturing and Rendering.
    param
    desc
    bufferType
    Specify video data structure type.
    listener
    listen for custom rendering
    pixelFormat
    Specify the format of the pixel called back
    userId
    ID of the specified remote user
    Note
    Before this API is called, startRemoteView(nil) needs to be called to get the video stream of the remote user ( view can be set to nil for this end); otherwise, there will be no data called back.

    setAudioFrameListener

    setAudioFrameListener
    void setAudioFrameListener
    (TRTCCloudListener.TRTCAudioFrameListener listener)

    Set custom audio data callback

    After this callback is set, the SDK will internally call back the audio data (in PCM format), including:
    onCapturedRawAudioFrame: callback of the original audio data captured by the local mic
    onLocalProcessedAudioFrame: callback of the audio data captured by the local mic and preprocessed by the audio module
    onRemoteUserAudioFrame: 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
    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.

    setCapturedRawAudioFrameCallbackFormat

    setCapturedRawAudioFrameCallbackFormat
    int setCapturedRawAudioFrameCallbackFormat
    (TRTCCloudDef.TRTCAudioFrameCallbackFormat format)

    Set the callback format of original audio frames captured by local mic

    This API is used to set the AudioFrame format called back by onCapturedRawAudioFrame:
    sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
    channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
    samplesPerCall: number of sample points, which defines the frame length of the callback data. The frame length must be an integer multiple of 10 ms.
    
    If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: number of sample points = number of milliseconds * sample rate / 1000
    For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as 960 = 20 * 48000 / 1000
    Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: number of bytes = number of sample points * number of channels * 2 (bit width)
    For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is 3840 = 960 * 2 * 2
    param
    desc
    format
    Audio data callback format

    setLocalProcessedAudioFrameCallbackFormat

    setLocalProcessedAudioFrameCallbackFormat
    int setLocalProcessedAudioFrameCallbackFormat
    (TRTCCloudDef.TRTCAudioFrameCallbackFormat format)

    Set the callback format of preprocessed local audio frames

    This API is used to set the AudioFrame format called back by onLocalProcessedAudioFrame:
    sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
    channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
    samplesPerCall: number of sample points, which defines the frame length of the callback data. The frame length must be an integer multiple of 10 ms.
    
    If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: number of sample points = number of milliseconds * sample rate / 1000
    For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as 960 = 20 * 48000 / 1000
    Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: number of bytes = number of sample points * number of channels * 2 (bit width)
    For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is 3840 = 960 * 2 * 2
    param
    desc
    format
    Audio data callback format

    setMixedPlayAudioFrameCallbackFormat

    setMixedPlayAudioFrameCallbackFormat
    int setMixedPlayAudioFrameCallbackFormat
    (TRTCCloudDef.TRTCAudioFrameCallbackFormat format)

    Set the callback format of audio frames to be played back by system

    This API is used to set the AudioFrame format called back by onMixedPlayAudioFrame:
    sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
    channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
    samplesPerCall: number of sample points, which defines the frame length of the callback data. The frame length must be an integer multiple of 10 ms.
    
    If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: number of sample points = number of milliseconds * sample rate / 1000
    For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as 960 = 20 * 48000 / 1000
    Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: number of bytes = number of sample points * number of channels * 2 (bit width)
    For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is 3840 = 960 * 2 * 2
    param
    desc
    format
    Audio data callback format

    enableCustomAudioRendering

    enableCustomAudioRendering
    void enableCustomAudioRendering
    (boolean enable)

    Enabling custom audio playback

    You can use this API to enable custom audio playback if you want to connect to an external audio device or control the audio playback logic by yourself.
    After you enable custom audio playback, the SDK will stop using its audio API to play back audio. You need to call getCustomAudioRenderingFrame to get audio frames and play them by yourself.
    param
    desc
    enable
    Whether to enable custom audio playback. It’s disabled by default.
    Note
    The parameter must be set before room entry to take effect.

    getCustomAudioRenderingFrame

    getCustomAudioRenderingFrame
    void getCustomAudioRenderingFrame
    (final TRTCCloudDef.TRTCAudioFrame audioFrame)

    Getting playable audio data

    Before calling this API, you need to first enable custom audio playback using enableCustomAudioRendering.
    
    Fill the fields in TRTCAudioFrame as follows (other fields are not required):
    sampleRate : sample rate (required). Valid values: 16000, 24000, 32000, 44100, 48000
    channel : number of sound channels (required). 1 : mono-channel; 2 : dual-channel; if dual-channel is used, data is interleaved.
    data : the buffer used to get audio data. You need to allocate memory for the buffer based on the duration of an audio frame.
    The PCM data obtained can have a frame duration of 10 ms or 20 ms. 20 ms is recommended.
    Assume that the sample rate is 48000, and sound channels mono-channel. The buffer size for a 20 ms audio frame would be 48000 x 0.02s x 1 x 16 bit = 15360 bit = 1920 bytes.
    param
    desc
    audioFrame
    Audio frames
    Note
    1. You must set sampleRate and channel in audioFrame , and allocate memory for one frame of audio in advance.
    2. The SDK will fill the data automatically based on sampleRate and channel .
    3. We recommend that you use the system’s audio playback thread to drive the calling of this API, so that it is called each time the playback of an audio frame is complete.

    sendCustomCmdMsg

    sendCustomCmdMsg
    boolean sendCustomCmdMsg
    (int cmdID
    
    byte[] data
    
    boolean reliable
    
    boolean ordered)

    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.
    Other users in the room can receive the message through the onRecvCustomCmdMsg callback in TRTCCloudDelegate.
    param
    desc
    cmdID
    Message ID. Value range: 1–10
    data
    Message to be sent. The maximum length of one single message is 1 KB.
    ordered
    Whether orderly sending is enabled, i.e., whether the data packets should be received in the same order in which they are sent; if so, a certain delay will be caused.
    reliable
    Whether reliable sending is enabled. Reliable sending can achieve a higher success rate but with a longer reception delay than unreliable sending.
    Note
    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.
    6. Currently only the anchor role is supported.

    sendSEIMsg

    sendSEIMsg
    boolean sendSEIMsg
    (byte[] data
    
    int repeatCount)

    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 callback in TRTCCloudDelegate.
    param
    desc
    data
    Data to be sent, which can be up to 1 KB (1,000 bytes)
    repeatCount
    Data sending count
    Note
    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 callback; therefore, deduplication is required.

    startSpeedTest

    startSpeedTest
    int startSpeedTest
    (TRTCCloudDef.TRTCSpeedTestParams params)

    Start network speed test (used before room entry)

    param
    desc
    params
    speed test options
    Note
    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.

    stopSpeedTest

    stopSpeedTest

    Stop network speed test

    getSDKVersion

    getSDKVersion

    Get SDK version information

    setLogLevel

    setLogLevel
    void setLogLevel
    (int level)

    Set log output level

    param
    desc
    level
    For more information, please see TRTCLogLevel. Default value: TRTCLogLevelNone

    setConsoleEnabled

    setConsoleEnabled
    void setConsoleEnabled
    (boolean enabled)

    Enable/Disable console log printing

    param
    desc
    enabled
    Specify whether to enable it, which is disabled by default

    setLogCompressEnabled

    setLogCompressEnabled
    void setLogCompressEnabled
    (boolean enabled)

    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.
    param
    desc
    enabled
    Specify whether to enable it, which is enabled by default

    setLogDirPath

    setLogDirPath
    void setLogDirPath
    (String path)

    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 .
    iOS or macOS: under sandbox Documents/log .
    Android: under /app directory/files/log/liteav/ .
    param
    desc
    path
    Log storage path
    Note
    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.

    setLogListener

    setLogListener
    void setLogListener
    (final TRTCCloudListener.TRTCLogListener logListener)

    Set log callback

    showDebugView

    showDebugView
    void showDebugView
    (int showType)

    Display dashboard

    "Dashboard" is a semi-transparent floating layer for debugging information on top of the video rendering control. It is used to display audio/video information and event information to facilitate integration and debugging.
    param
    desc
    showType
    0: does not display; 1: displays lite edition (only with audio/video information); 2: displays full edition (with audio/video information and event information).

    TRTCViewMargin

    TRTCViewMargin
    public TRTCViewMargin
    (float leftMargin
    
    float rightMargin
    
    float topMargin
    
    float bottomMargin)

    Set dashboard margin

    This API is used to adjust the position of the dashboard in the video rendering control. It must be called before showDebugView for it to take effect.
    param
    desc
    margin
    Inner margin of the dashboard. It should be noted that this is based on the percentage of parentView . Value range: 0–1
    userId
    User ID

    callExperimentalAPI

    callExperimentalAPI
    void callExperimentalAPI
    (String jsonStr)

    Call experimental APIs

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