TRTCCloudDef class Null safety

Key class definition variable

Constructors

TRTCCloudDef()

Properties

hashCode → int
The hash code for this object. [...]
read-only, inherited
runtimeType → Type
A representation of the runtime type of the object.
read-only, inherited

Methods

noSuchMethod(Invocation invocation) → dynamic
Invoked when a non-existent method or property is accessed. [...]
inherited
toString() → String
A string representation of this object. [...]
inherited

Operators

operator ==(Object other) → bool
The equality operator. [...]
inherited

Static Properties

TRTC_APP_SCENE_AUDIOCALL → int
In the audio call scenario, 48 kHz dual-channel audio call is supported. A single room can sustain up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
Use cases: one-to-one audio call, audio conferencing with up to 300 participants, voice chat, online Werewolf, etc.
final
TRTC_APP_SCENE_LIVE → int
In the interactive video live streaming scenario, mic can be turned on/off smoothly without waiting for switchover, and the anchor latency is as low as less than 300 ms. Live streaming to hundreds of thousands of concurrent audience users is supported with the playback latency down to 1,000 ms.
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.
Note: in this scenario, you must use the role field in TRTCParams to specify the role of the current user.
final
TRTC_APP_SCENE_VIDEOCALL → int
In the video call scenario, 720p and 1080p HD image quality is supported. A single room can sustain up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
Use cases: one-to-one video call, video conferencing with up to 300 participants, online medical diagnosis, video chat, video interview, etc.
final
TRTC_APP_SCENE_VOICE_CHATROOM → int
In the interactive audio live streaming scenario, mic can be turned on/off smoothly without waiting for switchover, and the anchor latency is as low as less than 300 ms. Live streaming to hundreds of thousands of concurrent audience users is supported with the playback latency down to 1,000 ms.
Use cases: low-latency audio live streaming, live audio co-anchoring, voice chat room, karaoke room, FM radio, etc.
Note: in this scenario, you must use the role field in TRTCParams to specify the role of the current user.
final
TRTC_AUDIO_FRAME_FORMAT_PCM → int
PCM
final
TRTC_AUDIO_QUALITY_DEFAULT → int
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.
final
TRTC_AUDIO_QUALITY_MUSIC → int
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 karaoke and music live streaming.
final
TRTC_AUDIO_QUALITY_SPEECH → int
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.
final
TRTC_AUDIO_ROUTE_EARPIECE → int
Headphones
final
TRTC_AUDIO_ROUTE_SPEAKER → int
Speaker
final
TRTC_BEAUTY_STYLE_NATURE → int
Natural style, which retains more facial details and seems more natural subjectively.
final
TRTC_BEAUTY_STYLE_PITU → int
Pitu style, which is more natural and retains more skin details than the smooth style.
final
TRTC_BEAUTY_STYLE_SMOOTH → int
Smooth style, which is suitable for shows since it has more obvious effect.
final
TRTC_DEBUG_VIEW_LEVEL_ALL → int
The upper part of the UI displays the status logs, and the lower part displays the key events
final
TRTC_DEBUG_VIEW_LEVEL_GONE → int
The UI doesn't display logs
final
TRTC_DEBUG_VIEW_LEVEL_STATUS → int
The upper part of the UI displays the status logs
final
TRTC_GSENSOR_MODE_DISABLE → int
Disable G-sensor
final
TRTC_GSENSOR_MODE_UIAUTOLAYOUT → int
Enable G-sensor (default value).
final
TRTC_GSENSOR_MODE_UIFIXLAYOUT → int
This is to be disused and equivalent to UIAutoLayout.
final
TRTC_LOG_LEVEL_DEBUG → int
Output logs at the DEBUG, INFO, WARNING, ERROR, and FATAL levels
final
TRTC_LOG_LEVEL_ERROR → int
Output logs at the ERROR and FATAL levels
final
TRTC_LOG_LEVEL_FATAL → int
Output logs at the FATAL level
final
TRTC_LOG_LEVEL_INFO → int
Output logs at the INFO, WARNING, ERROR, and FATAL levels
final
TRTC_LOG_LEVEL_NULL → int
Do not output any SDK logs
final
TRTC_LOG_LEVEL_VERBOSE → int
Output logs at all levels
final
TRTC_LOG_LEVEL_WARN → int
Output logs at the WARNING, ERROR, and FATAL levels
final
TRTC_QUALITY_Bad → int
Bad
final
TRTC_QUALITY_Down → int
Unavailable
final
TRTC_QUALITY_Excellent → int
Excellent
final
TRTC_QUALITY_Good → int
Good
final
TRTC_QUALITY_Poor → int
Poor
final
TRTC_QUALITY_UNKNOWN → int
Undefined
final
TRTC_QUALITY_Vbad → int
Very bad
final
TRTC_REVERB_TYPE_0 → int
Disable reverb
final
TRTC_REVERB_TYPE_1 → int
KTV
final
TRTC_REVERB_TYPE_2 → int
Small room
final
TRTC_REVERB_TYPE_3 → int
Big hall
final
TRTC_REVERB_TYPE_4 → int
Deep
final
TRTC_REVERB_TYPE_5 → int
Resonant
final
TRTC_REVERB_TYPE_6 → int
Metallic
final
TRTC_REVERB_TYPE_7 → int
Husky
final
TRTC_SDK_VERSION → String
final
TRTC_TranscodingConfigMode_Manual → int
Manual mode. It is most flexible and can implement various mixtranscoding schemes through free combinations, but it is most difficult to use. In this mode, you need to enter all the parameters in TRTCTranscodingConfig and listen on the onUserVideoAvailable() and onUserAudioAvailable() callbacks in TRTCCloudDelegate so as to constantly adjust the mixUsers parameter according to the audio/video status of each user with mic on in the current room; otherwise, mixtranscoding will fail.
final
TRTC_TranscodingConfigMode_Template_PresetLayout → int
Preset layout mode, where the layout of each channel of image is arranged in advance through placeholders. In this mode, you still need to set the mixUsers parameter, but you can set userId as a placeholder. Placeholder values include: [...]
final
TRTC_TranscodingConfigMode_Template_PureAudio → int
PureAudio mode. It is suitable for pure audio scenarios such as audio call (AudioCall) and voice chat room (VoiceChatRoom). You only need to set it once through the setMixTranscodingConfig() API after room entry, and then the SDK will automatically mix the audios of all mic-on users in the room into the current user's live stream. In this mode, you don't need to set the mixUsers parameter in TRTCTranscodingConfig; instead, you only need to set the audioSampleRate, audioBitrate and audioChannels parameters.
final
TRTC_TranscodingConfigMode_Template_ScreenSharing → int
Screen sharing mode, which is suitable for screen sharing-based use cases such as online education and supported only by the SDKs for Windows and macOS. The SDK will first build a canvas according to the target resolution you set (through the videoWidth and videoHeight parameters). Before the teacher enables screen sharing, the SDK will scale up the camera image proportionally and draw it onto the canvas. After the teacher enables screen sharing, the SDK will draw the video image shared on the screen onto the same canvas. The purpose of this procedure is to ensure consistency in the output resolution of the mixtranscoding module and avoid problems with blurred screen during course replay and webpage playback (web players don't support adjustable resolution). Meanwhile, the audios of mic-on students will be mixed into the teacher's audio/video stream by default. [...]
final
TRTC_VIDEO_MIRROR_TYPE_DISABLE → int
Do not mirror the images of both the front and rear cameras
final
TRTC_VIDEO_MIRROR_TYPE_ENABLE → int
Mirror the images of both the front and rear cameras
final
TRTC_VIDEO_PIXEL_FORMAT_I420 → int
YUV I420
final
TRTC_VIDEO_PIXEL_FORMAT_NV21 → int
final
TRTC_VIDEO_PIXEL_FORMAT_Texture_2D → int
OpenGL 2D texture
final
TRTC_VIDEO_PIXEL_FORMAT_TEXTURE_EXTERNAL_OES → int
final
TRTC_VIDEO_PIXEL_FORMAT_UNKNOWN → int
Unknown
final
TRTC_VIDEO_QOS_PREFERENCE_CLEAR → int
Ensure definition on a weak network (default value)
final
TRTC_VIDEO_QOS_PREFERENCE_SMOOTH → int
Ensure smoothness on a weak network
final
TRTC_VIDEO_RESOLUTION_120_120 → int
Here, only the landscape resolution is defined. If the portrait resolution (e.g., 360x640) needs to be used, Portrait must be selected for TRTCVideoResolutionMode. Recommended bitrate: VideoCall: 80 Kbps, LIVE: 120 Kbps
final
TRTC_VIDEO_RESOLUTION_160_90 → int
Recommended bitrate: VideoCall: 150 Kbps, LIVE: 250 Kbps
final
TRTC_VIDEO_RESOLUTION_160_120 → int
Recommended bitrate: VideoCall: 100 Kbps, LIVE: 150 Kbps
final
TRTC_VIDEO_RESOLUTION_160_160 → int
Recommended bitrate: VideoCall: 100 Kbps, LIVE: 150 Kbps
final
TRTC_VIDEO_RESOLUTION_240_180 → int
Recommended bitrate: VideoCall: 150 Kbps, LIVE: 225 Kbps
final
TRTC_VIDEO_RESOLUTION_256_144 → int
Recommended bitrate: VideoCall: 200 Kbps, LIVE: 300 Kbps
final
TRTC_VIDEO_RESOLUTION_270_270 → int
Recommended bitrate: VideoCall: 200 Kbps, LIVE: 120 Kbps
final
TRTC_VIDEO_RESOLUTION_280_210 → int
Recommended bitrate: VideoCall: 200 Kbps, LIVE: 300 Kbps
final
TRTC_VIDEO_RESOLUTION_320_180 → int
Recommended bitrate: VideoCall: 250 Kbps, LIVE: 400 Kbps
final
TRTC_VIDEO_RESOLUTION_320_240 → int
Recommended bitrate: VideoCall: 250 Kbps, LIVE: 375 Kbps
final
TRTC_VIDEO_RESOLUTION_400_300 → int
Recommended bitrate: VideoCall: 300 Kbps, LIVE: 450 Kbps
final
TRTC_VIDEO_RESOLUTION_480_270 → int
Recommended bitrate: VideoCall: 350 Kbps, LIVE: 550 Kbps
final
TRTC_VIDEO_RESOLUTION_480_360 → int
Recommended bitrate: VideoCall: 400 Kbps, LIVE: 600 Kbps
final
TRTC_VIDEO_RESOLUTION_480_480 → int
Recommended bitrate: VideoCall: 350 Kbps, LIVE: 120 Kbps
final
TRTC_VIDEO_RESOLUTION_640_360 → int
Recommended bitrate: VideoCall: 550 Kbps, LIVE: 900 Kbps
final
TRTC_VIDEO_RESOLUTION_640_480 → int
Recommended bitrate: VideoCall: 600 Kbps, LIVE: 900 Kbps
final
TRTC_VIDEO_RESOLUTION_960_540 → int
Recommended bitrate: VideoCall: 850 Kbps, LIVE: 1300 Kbps
final
TRTC_VIDEO_RESOLUTION_960_720 → int
Recommended bitrate: VideoCall: 1000 Kbps, LIVE: 1500 Kbps
final
TRTC_VIDEO_RESOLUTION_1280_720 → int
Recommended bitrate: VideoCall: 1200 Kbps, LIVE: 1800 Kbps
final
TRTC_VIDEO_RESOLUTION_1920_1080 → int
Recommended bitrate: VideoCall: 2000 Kbps, LIVE: 3000 Kbps
final
TRTC_VIDEO_RESOLUTION_MODE_LANDSCAPE → int
Landscape resolution
final
TRTC_VIDEO_RESOLUTION_MODE_PORTRAIT → int
Portrait resolution
final
TRTC_VIDEO_ROTATION_90 → int
Rotate 90 degrees clockwise
final
TRTC_VIDEO_ROTATION_180 → int
Rotate 180 degrees clockwise
final
TRTC_VIDEO_ROTATION_270 → int
Rotate 270 degrees clockwise
final
TRTC_VIDEO_STREAM_TYPE_SMALL → int
Small image video stream
final
TRTC_VIDEO_STREAM_TYPE_SUB → int
Substream (screen sharing)
final
TRTC_VideoView_SurfaceView → int
Use SurfaceView for Android video rendering
final
TRTC_VideoView_TextureView → int
Use TextureView for Android video rendering
final
TRTC_VOICE_CHANGER_TYPE_0 → int
Disable voice changing
final
TRTC_VOICE_CHANGER_TYPE_1 → int
Naughty boy
final
TRTC_VOICE_CHANGER_TYPE_2 → int
Young girl
final
TRTC_VOICE_CHANGER_TYPE_3 → int
Middle-Aged man
final
TRTC_VOICE_CHANGER_TYPE_4 → int
Heavy metal
final
TRTC_VOICE_CHANGER_TYPE_5 → int
Cold
final
TRTC_VOICE_CHANGER_TYPE_6 → int
Punk
final
TRTC_VOICE_CHANGER_TYPE_7 → int
Furious animal
final
TRTC_VOICE_CHANGER_TYPE_8 → int
Chubby
final
TRTC_VOICE_CHANGER_TYPE_9 → int
Strong electric current
final
TRTC_VOICE_CHANGER_TYPE_10 → int
Robot
final
TRTC_VOICE_CHANGER_TYPE_11 → int
Ethereal
final
TRTCAudioSampleRate16000 → int
16 kHz sample rate
final
TRTCAudioSampleRate32000 → int
32 kHz sample rate
final
TRTCAudioSampleRate44100 → int
44.1 kHz sample rate
final
TRTCAudioSampleRate48000 → int
48 kHz sample rate
final
TRTCRecordTypeAudio → int
Record audio only
final
TRTCRecordTypeBoth → int
Record both audio and video
final
TRTCRecordTypeVideo → int
Record video only
final
TRTCSystemVolumeTypeAuto → int
"Call volume with mic and media volume without mic", i.e., the call volume mode will be used when the anchor mics on, while the media volume mode will be used when the audience user mics off. This is suitable for live streaming scenarios.
If the scenario you select during enterRoom is TRTC_APP_SCENE_LIVE or TRTC_APP_SCENE_VOICE_CHATROOM, the SDK will automatically select this mode.
final
TRTCSystemVolumeTypeMedia → int
The media volume mode is used throughout the call. This is not common and is suitable for scenarios with special requirements (for example, the anchor has an external sound card).
final
TRTCSystemVolumeTypeVOIP → int
The call volume mode will be always used, which is suitable for conferencing scenarios. If the scenario you select during enterRoom is TRTC_APP_SCENE_VIDEOCALL or TRTC_APP_SCENE_AUDIOCALL, the SDK will automatically select this mode.
final
TXMediaDeviceTypeCamera → int
Camera
final
TXMediaDeviceTypeMic → int
Mic
final
TXMediaDeviceTypeSpeaker → int
Speaker or receiver
final
TXMediaDeviceTypeUnknown → int
Unknown type
final
VIDEO_QOS_CONTROL_CLIENT → int
Client-based control (which is for internal debugging of the SDK and should not be used)
final
VIDEO_QOS_CONTROL_SERVER → int
On-cloud control (default value)
final

Constants

TRTC_TranscodingConfigMode_Unknown → const int
Invalid value
0
TRTC_VIDEO_MIRROR_TYPE_AUTO → const int
The SDK determines the mirror type: mirroring the front camera's image bur not the rear camera's image
0
TRTC_VIDEO_RENDER_MODE_FILL → const int
The entire screen will be covered by the image, where parts that exceed the screen will be cropped
0
TRTC_VIDEO_RENDER_MODE_FIT → const int
The long side of the image will fit the screen, while the short side will be proportionally scaled with unmatched areas being filled with black color blocks
1
TRTC_VIDEO_ROTATION_0 → const int
No rotation
0
TRTC_VIDEO_STREAM_TYPE_BIG → const int
Primary image video stream
0
TRTCRoleAnchor → const int
Anchor
20
TRTCRoleAudience → const int
Audience
21