Release Notes and Announcements

Release Notes

Type Definition

Focus Mode

Font Size

Last updated: 2026-03-27 15:38:14

Type Definition
StructType
FuncList
DESC
TRTCParams
Room entry parameters.
TRTCVideoEncParam
Video encoding parameters.
TRTCNetworkQosParam
Network QoS control parameter set.
TRTCRenderParams
Rendering parameters of video image.
TRTCQualityInfo
Network quality.
TRTCVolumeInfo
Volume.
TRTCSpeedTestParams
Network speed testing parameters.
TRTCSpeedTestResult
Network speed test result.
TRTCVideoFrame
Video frame information.
TRTCAudioFrame
Audio frame data.
TRTCMixUser
Description information of each video image in On-Cloud MixTranscoding.
TRTCTranscodingConfig
Layout and transcoding parameters of On-Cloud MixTranscoding.
TRTCPublishCDNParam
Push parameters required to be set when publishing audio/video streams to non-Tencent Cloud CDN.
TRTCAudioRecordingParams
Local audio file recording parameters.
TRTCLocalRecordingParams
Local media file recording parameters.
TRTCAudioEffectParam
Sound effect parameter (disused).
TRTCSwitchRoomConfig
Room switch parameter.
TRTCAudioFrameDelegateFormat
Format parameter of custom audio callback.
TRTCUser
The users whose streams to publish.
TRTCPublishCdnUrl
The destination URL when you publish to Tencent Cloud or a third-party CDN.
TRTCPublishTarget
The publishing destination.
TRTCVideoLayout
The video layout of the transcoded stream.
TRTCWatermark
The watermark layout.
TRTCStreamEncoderParam
The encoding parameters.
TRTCStreamMixingConfig
The transcoding parameters.
TRTCPayloadPrivateEncryptionConfig
Media Stream Private Encryption Configuration.
TRTCAudioVolumeEvaluateParams
Volume evaluation and other related parameter settings.
EnumType
EnumType
DESC
TRTCVideoResolution
Video resolution.
TRTCVideoResolutionMode
Video aspect ratio mode.
TRTCVideoStreamType
Video stream type.
TRTCVideoFillMode
Video image fill mode.
TRTCVideoRotation
Video image rotation direction.
TRTCBeautyStyle
Beauty (skin smoothing) filter algorithm.
TRTCVideoPixelFormat
Video pixel format.
TRTCVideoBufferType
Video data transfer method.
TRTCVideoMirrorType
Video mirror type.
TRTCSnapshotSourceType
Data source of local video screenshot.
TRTCAppScene
Use cases.
TRTCRoleType
Role.
TRTCQosControlMode
QoS control mode (disused).
TRTCVideoQosPreference
Image quality preference.
TRTCQuality
Network quality.
TRTCAVStatusType
Audio/Video playback status.
TRTCAVStatusChangeReason
Reasons for playback status changes.
TRTCAudioSampleRate
Audio sample rate.
TRTCAudioQuality
Sound quality.
TRTCAudioRoute
Audio route (i.e., audio playback mode).
TRTCReverbType
Audio reverb mode.
TRTCVoiceChangerType
Voice changing type.
TRTCSystemVolumeType
System volume type (only for mobile devices).
TRTCAudioFrameOperationMode
Audio callback data operation mode.
TRTCLogLevel
Log level.
TRTCGSensorMode
G-sensor switch (for mobile devices only).
TRTCScreenCaptureSourceType
Screen sharing target type (for desktops only).
TRTCTranscodingConfigMode
Layout mode of On-Cloud MixTranscoding.
TRTCRecordType
Media recording type.
TRTCMixInputType
Stream mix input type.
TRTCAudioRecordingContent
Audio recording content type.
TRTCPublishMode
The publishing mode.
TRTCEncryptionAlgorithm
Encryption Algorithm.
TRTCSpeedTestScene
Speed Test Scene.
TRTCGravitySensorAdaptiveMode
Set the adaptation mode of gravity sensing (only applicable to mobile terminals).
TRTCVideoResolution
TRTCVideoResolution
Video resolution.Here, only the landscape resolution (e.g., 640x360) is defined. If the portrait resolution (e.g., 360x640) needs to be used,  Portrait  must be selected for ﻿TRTCVideoResolutionMode﻿.
Enum
Value
DESC
TRTCVideoResolution_120_120
1
Aspect ratio: 1:1; resolution: 120x120; recommended bitrate (VideoCall): 80 Kbps; recommended bitrate (LIVE): 120 Kbps.
TRTCVideoResolution_160_160
3
Aspect ratio: 1:1; resolution: 160x160; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
TRTCVideoResolution_270_270
5
Aspect ratio: 1:1; resolution: 270x270; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_480_480
7
Aspect ratio: 1:1; resolution: 480x480; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 500 Kbps.
TRTCVideoResolution_160_120
50
Aspect ratio: 4:3; resolution: 160x120; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
TRTCVideoResolution_240_180
52
Aspect ratio: 4:3; resolution: 240x180; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
TRTCVideoResolution_280_210
54
Aspect ratio: 4:3; resolution: 280x210; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_320_240
56
Aspect ratio: 4:3; resolution: 320x240; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 375 Kbps.
TRTCVideoResolution_400_300
58
Aspect ratio: 4:3; resolution: 400x300; recommended bitrate (VideoCall): 300 Kbps; recommended bitrate (LIVE): 450 Kbps.
TRTCVideoResolution_480_360
60
Aspect ratio: 4:3; resolution: 480x360; recommended bitrate (VideoCall): 400 Kbps; recommended bitrate (LIVE): 600 Kbps.
TRTCVideoResolution_640_480
62
Aspect ratio: 4:3; resolution: 640x480; recommended bitrate (VideoCall): 600 Kbps; recommended bitrate (LIVE): 900 Kbps.
TRTCVideoResolution_960_720
64
Aspect ratio: 4:3; resolution: 960x720; recommended bitrate (VideoCall): 1000kbps; recommended bitrate (LIVE): 1500kbps。
TRTCVideoResolution_160_90
100
Aspect ratio: 16:9; resolution: 160x90; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
TRTCVideoResolution_256_144
102
Aspect ratio: 16:9; resolution: 256x144; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_320_180
104
Aspect ratio: 16:9; resolution: 320x180; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 400 Kbps.
TRTCVideoResolution_480_270
106
Aspect ratio: 16:9; resolution: 480x270; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 550 Kbps.
TRTCVideoResolution_640_360
108
Aspect ratio: 16:9; resolution: 640x360; recommended bitrate (VideoCall): 500 Kbps; recommended bitrate (LIVE): 900 Kbps.
TRTCVideoResolution_960_540
110
Aspect ratio: 16:9; resolution: 960x540; recommended bitrate (VideoCall): 850 Kbps; recommended bitrate (LIVE): 1300 Kbps.
TRTCVideoResolution_1280_720
112
Aspect ratio: 16:9; resolution: 1280x720; recommended bitrate (VideoCall): 1200 Kbps; recommended bitrate (LIVE): 1800 Kbps.
TRTCVideoResolution_1920_1080
114
Aspect ratio: 16:9; resolution: 1920x1080; recommended bitrate (VideoCall): 2000 Kbps; recommended bitrate (LIVE): 3000 Kbps.
TRTCVideoResolutionMode
TRTCVideoResolutionMode
Video aspect ratio mode.Only the landscape resolution (e.g., 640x360) is defined in  TRTCVideoResolution . If the portrait resolution (e.g., 360x640) needs to be used,  Portrait  must be selected for  TRTCVideoResolutionMode .
Enum
Value
DESC
TRTCVideoResolutionModeLandscape
0
Landscape resolution, such as  TRTCVideoResolution_640_360 + TRTCVideoResolutionModeLandscape = 640x360 .
TRTCVideoResolutionModePortrait
1
Portrait resolution, such as  TRTCVideoResolution_640_360 + TRTCVideoResolutionModePortrait = 360x640 .
TRTCVideoStreamType
TRTCVideoStreamType
Video stream type.TRTC provides three different video streams, including:
 HD big image: it is generally used to transfer video data from the camera.
 Smooth small image: it has the same content as the big image, but with lower resolution and bitrate and thus lower definition.
 Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their own substream.
Note
The SDK does not support enabling the smooth small image alone, which must be enabled together with the big image. It will automatically set the resolution and bitrate of the small image.
Enum
Value
DESC
TRTCVideoStreamTypeBig
0
HD big image: it is generally used to transfer video data from the camera.
TRTCVideoStreamTypeSmall
1
Smooth small image: it has the same content as the big image, but with lower resolution and bitrate and thus lower definition.
TRTCVideoStreamTypeSub
2
Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their own substream.
TRTCVideoFillMode
TRTCVideoFillMode
Video image fill mode.If the aspect ratio of the video display area is not equal to that of the video image, you need to specify the fill mode:
Enum
Value
DESC
TRTCVideoFillMode_Fill
0
Fill mode: the video image will be centered and scaled to fill the entire display area, where parts that exceed the area will be cropped. The displayed image may be incomplete in this mode.
TRTCVideoFillMode_Fit
1
Fit mode: the video image will be scaled based on its long side to fit the display area, where the short side will be filled with black bars. The displayed image is complete in this mode, but there may be black bars.
TRTCVideoFillMode_ScaleFill
2
Scale-to-fill mode: This means that regardless of the aspect ratio of the image, it will be stretched or compressed to completely fill the display area. In this mode, the aspect ratio of the image may be altered, leading to distortion in the rendered image.
TRTCVideoRotation
TRTCVideoRotation
Video image rotation direction.TRTC provides rotation angle setting APIs for local and remote images. The following rotation angles are all clockwise.
Enum
Value
DESC
TRTCVideoRotation_0
0
No rotation
TRTCVideoRotation_90
1
Clockwise rotation by 90 degrees
TRTCVideoRotation_180
2
Clockwise rotation by 180 degrees
TRTCVideoRotation_270
3
Clockwise rotation by 270 degrees
TRTCBeautyStyle
TRTCBeautyStyle
Beauty (skin smoothing) filter algorithm.TRTC has multiple built-in skin smoothing algorithms. You can select the one most suitable for your product.
Enum
Value
DESC
TRTCBeautyStyleSmooth
0
Smooth style, which uses a more radical algorithm for more obvious effect and is suitable for show live streaming.
TRTCBeautyStyleNature
1
Natural style, which retains more facial details for more natural effect and is suitable for most live streaming use cases.
TRTCBeautyStylePitu
2
Pitu style, which is provided by YouTu Lab. Its skin smoothing effect is between the smooth style and the natural style, that is, it retains more skin details than the smooth style and has a higher skin smoothing degree than the natural style.
TRTCVideoPixelFormat
TRTCVideoPixelFormat
Video pixel format.TRTC provides custom video capturing and rendering features.
 For the custom capturing feature, you can use the following enumerated values to describe the pixel format of the video you capture.
 For the custom rendering feature, you can specify the pixel format of the video you expect the SDK to call back.
Enum
Value
DESC
TRTCVideoPixelFormat_Unknown
0
Undefined format
TRTCVideoPixelFormat_I420
1
YUV420P (I420) format
TRTCVideoPixelFormat_Texture_2D
7
OpenGL 2D texture format
TRTCVideoPixelFormat_32BGRA
6
BGRA32 format
TRTCVideoPixelFormat_NV12
5
YUV420SP (NV12) format
TRTCVideoBufferType
TRTCVideoBufferType
Video data transfer method.For custom capturing and rendering features, you need to use the following enumerated values to specify the method of transferring video data:
 Method 1. This method uses memory buffer to transfer video data. It is efficient on iOS but inefficient on Android. It is the only method supported on Windows currently.
 Method 2. This method uses texture to transfer video data. It is efficient on both iOS and Android but is not supported on Windows. To use this method, you should have a general familiarity with OpenGL programming.
Enum
Value
DESC
TRTCVideoBufferType_Unknown
0
Undefined transfer method
TRTCVideoBufferType_PixelBuffer
1
Use memory buffer to transfer video data. iOS:  PixelBuffer ; Android:  Direct Buffer  for JNI layer; Windows: memory data block.
TRTCVideoBufferType_NSData
2
Use memory buffer to transfer video data. iOS: more compact memory block in  NSData  type after additional processing; Android:  byte[]  for Java layer.
This transfer method has a lower efficiency than other methods.
TRTCVideoBufferType_Texture
3
Use OpenGL texture to transfer video data
TRTCVideoMirrorType
TRTCVideoMirrorType
Video mirror type.Video mirroring refers to the left-to-right flipping of the video image, especially for the local camera preview image. After mirroring is enabled, it can bring anchors a familiar "look into the mirror" experience.
Enum
Value
DESC
TRTCVideoMirrorTypeAuto
0
Auto mode: mirror the front camera's image but not the rear camera's image (for mobile devices only).
TRTCVideoMirrorTypeEnable
1
Mirror the images of both the front and rear cameras.
TRTCVideoMirrorTypeDisable
2
Disable mirroring for both the front and rear cameras.
TRTCSnapshotSourceType
TRTCSnapshotSourceType
Data source of local video screenshot.The SDK can take screenshots from the following two data sources and save them as local files:
 Video stream: the SDK screencaptures the native video content from the video stream. The screenshots are not controlled by the display of the rendering control.
 Rendering layer: the SDK screencaptures the displayed video content from the rendering control, which can achieve the effect of WYSIWYG, but if the display area is too small, the screenshots will also be very small.
Enum
Value
DESC
TRTCSnapshotSourceTypeStream
0
The SDK screencaptures the native video content from the video stream. The screenshots are not controlled by the display of the rendering control.
TRTCSnapshotSourceTypeView
1
The SDK screencaptures the displayed video content from the rendering control, which can achieve the effect of WYSIWYG, but if the display area is too small, the screenshots will also be very small.
TRTCSnapshotSourceTypeCapture
2
The SDK screencaptures the capture video content from the capture control, which can capture the captured high-definition screenshots.
TRTCAppScene
TRTCAppScene
Use cases.TRTC features targeted optimizations for common audio/video application scenarios to meet the differentiated requirements in various verticals. The main scenarios can be divided into the following two categories:
 Live streaming scenario (LIVE): including  LIVE  (audio + video) and  VoiceChatRoom  (pure audio).
In the live streaming scenario, users are divided into two roles: "anchor" and "audience". A single room can sustain up to 100,000 concurrent online users. This is suitable for live streaming to a large audience.
 Real-Time scenario (RTC): including  VideoCall  (audio + video) and  AudioCall  (pure audio).
In the real-time scenario, there is no role difference between users, but a single room can sustain only up to 300 concurrent online users. This is suitable for small-scale real-time communication.
Enum
Value
DESC
TRTCAppSceneVideoCall
0
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], [small class], [video interview], etc.
TRTCAppSceneLIVE
1
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 users in the audience role is supported with the playback latency down to 1,000 ms.
Use cases: [low-latency interactive live streaming], [big class], [anchor competition], [video dating room], [online interactive classroom], [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.
TRTCAppSceneAudioCall
2
Audio call scenario, where the  SPEECH  sound quality is used by default. 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], [audio chat], [online Werewolf], etc.
TRTCAppSceneVoiceChatRoom
3
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 users in the audience role is supported with the playback latency down to 1,000 ms.
Use cases: [audio club], [online karaoke room], [music live room], [FM radio], etc.
Note
In this scenario, you must use the  role  field in  TRTCParams  to specify the role of the current user.
TRTCRoleType
TRTCRoleType
Role.Role is applicable only to live streaming scenarios ( TRTCAppSceneLIVE  and  TRTCAppSceneVoiceChatRoom ). Users are divided into two roles:
 Anchor, who can publish their audio/video streams. There is a limit on the number of anchors. Up to 50 anchors are allowed to publish streams at the same time in one room.
 Audience, who can only listen to or watch audio/video 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 can sustain up to 100,000 concurrent online users in the audience role.
Enum
Value
DESC
TRTCRoleAnchor
20
An anchor can publish their audio/video streams. There is a limit on the number of anchors. Up to 50 anchors are allowed to publish streams at the same time in one room.
TRTCRoleAudience
21
Audience can only listen to or watch audio/video 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 can sustain up to 100,000 concurrent online users in the audience role.
TRTCQosControlMode(Deprecated)
TRTCQosControlMode(Deprecated)
QoS control mode (disused).Enum
Value
DESC
TRTCQosControlModeClient
0
Client-based control, which is for internal debugging of SDK and shall not be used by users.
TRTCQosControlModeServer
1
On-cloud control, which is the default and recommended mode.
TRTCVideoQosPreference
TRTCVideoQosPreference
Image quality preference.TRTC has two control modes in weak network environments: "ensuring clarity" and "ensuring smoothness". Both modes will give priority to the transfer of audio data.
Enum
Value
DESC
TRTCVideoQosPreferenceSmooth
1
Ensuring smoothness: in this mode, when the current network is unable to transfer a clear and smooth video image, the smoothness of the image will be given priority, but there will be blurs.
TRTCVideoQosPreferenceClear
2
Ensuring clarity (default value): in this mode, when the current network is unable to transfer a clear and smooth video image, the clarity of the image will be given priority, but there will be lags.
TRTCQuality
TRTCQuality
Network quality.TRTC evaluates the current network quality once every two seconds. The evaluation results are divided into six levels:  Excellent  indicates the best, and  Down  indicates the worst.
Enum
Value
DESC
TRTCQuality_Unknown
0
Undefined
TRTCQuality_Excellent
1
The current network is excellent
TRTCQuality_Good
2
The current network is good
TRTCQuality_Poor
3
The current network is fair
TRTCQuality_Bad
4
The current network is bad
TRTCQuality_Vbad
5
The current network is very bad
TRTCQuality_Down
6
The current network cannot meet the minimum requirements of TRTC
TRTCAVStatusType
TRTCAVStatusType
Audio/Video playback status.This enumerated type is used in the audio status changed API ﻿onRemoteAudioStatusUpdated﻿ and the video status changed API ﻿onRemoteVideoStatusUpdated﻿ to specify the current audio/video status.
Enum
Value
DESC
TRTCAVStatusStopped
0
Stopped
TRTCAVStatusPlaying
1
Playing
TRTCAVStatusLoading
2
Loading
TRTCAVStatusChangeReason
TRTCAVStatusChangeReason
Reasons for playback status changes.This enumerated type is used in the audio status changed API ﻿onRemoteAudioStatusUpdated﻿ and the video status changed API ﻿onRemoteVideoStatusUpdated﻿ to specify the reason for the current audio/video status change.
Enum
Value
DESC
TRTCAVStatusChangeReasonInternal
0
Default value
TRTCAVStatusChangeReasonBufferingBegin
1
The stream enters the  Loading  state due to network congestion
TRTCAVStatusChangeReasonBufferingEnd
2
The stream enters the  Playing  state after network recovery
TRTCAVStatusChangeReasonLocalStarted
3
As a start-related API was directly called locally, the stream enters the  Playing  state
TRTCAVStatusChangeReasonLocalStopped
4
As a stop-related API was directly called locally, the stream enters the  Stopped  state
TRTCAVStatusChangeReasonRemoteStarted
5
As the remote user started (or resumed) publishing the audio or video stream, the stream enters the  Loading  or  Playing  state
TRTCAVStatusChangeReasonRemoteStopped
6
As the remote user stopped (or paused) publishing the audio or video stream, the stream enters the "Stopped" state
TRTCAudioSampleRate
TRTCAudioSampleRate
Audio sample rate.The audio sample rate is used to measure the audio fidelity. A higher sample rate indicates higher fidelity. If there is music in the use case,  TRTCAudioSampleRate48000  is recommended.
Enum
Value
DESC
TRTCAudioSampleRate16000
16000
16 kHz sample rate
TRTCAudioSampleRate32000
32000
32 kHz sample rate
TRTCAudioSampleRate44100
44100
44.1 kHz sample rate
TRTCAudioSampleRate48000
48000
48 kHz sample rate
TRTCAudioQuality
TRTCAudioQuality
Sound quality.TRTC provides three well-tuned modes to meet the differentiated requirements for sound quality in various verticals:
 Speech mode (Speech): it is suitable for application scenarios that focus on human communication. In this mode, the audio transfer is more resistant, and TRTC uses various voice processing technologies to ensure the optimal smoothness even in weak network environments.
 Music mode (Music): it is suitable for scenarios with demanding requirements for music. In this mode, the amount of transferred audio data is very large, and TRTC uses various technologies to ensure that the high-fidelity details of music signals can be restored in each frequency band.
 Default mode (Default): it is between  Speech  and  Music . In this mode, the reproduction of music is better than that in  Speech  mode, and the amount of transferred data is much lower than that in  Music  mode; therefore, this mode has good adaptability to various scenarios.
Enum
Value
DESC
TRTCAudioQualitySpeech
1
Speech mode: mono channel; bitrate: 18 Kbps. This mode has the best resistance among all modes and is suitable for audio call scenarios, such as online meeting and audio call.
TRTCAudioQualityDefault
2
Default mode: mono channel; bitrate: 50 Kbps. This mode is between the speech mode and the music mode as the default mode in the SDK and is recommended.
TRTCAudioQualityMusic
3
Music mode: full-band stereo; bitrate: 128 Kbps. This mode is suitable for scenarios where Hi-Fi music transfer is required, such as online karaoke and music live streaming.
TRTCAudioRoute
TRTCAudioRoute
Audio route (i.e., audio playback mode)."Audio route" determines whether the sound is played back from the speaker or receiver of a mobile device; therefore, this API is applicable only 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 the 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 the 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.
Enum
Value
DESC
TRTCAudioModeUnknown
-1
Unknown：default router.
TRTCAudioModeSpeakerphone
0
Speakerphone: the speaker at the bottom is used for playback (hands-free). With relatively high volume, it is used to play music out loud.
TRTCAudioModeEarpiece
1
Earpiece: the receiver at the top is used for playback. With relatively low volume, it is suitable for call scenarios that require privacy.
TRTCAudioModeWiredHeadset
2
WiredHeadset：play using wired headphones.
TRTCAudioModeBluetoothHeadset
3
BluetoothHeadset：play with bluetooth headphones.
TRTCAudioModeSoundCard
4
SoundCard：play using a USB sound card.
TRTCReverbType
TRTCReverbType
Audio reverb mode.This enumerated value is used to set the audio reverb mode in the live streaming scenario and is often used in show live streaming.
Enum
Value
DESC
TRTCReverbType_0
0
Disable reverb
TRTCReverbType_1
1
KTV
TRTCReverbType_2
2
Small room
TRTCReverbType_3
3
Hall
TRTCReverbType_4
4
Deep
TRTCReverbType_5
5
Resonant
TRTCReverbType_6
6
Metallic
TRTCReverbType_7
7
Husky
TRTCVoiceChangerType
TRTCVoiceChangerType
Voice changing type.This enumerated value is used to set the voice changing mode in the live streaming scenario and is often used in show live streaming.
Enum
Value
DESC
TRTCVoiceChangerType_0
0
Disable voice changing
TRTCVoiceChangerType_1
1
Child
TRTCVoiceChangerType_2
2
Girl
TRTCVoiceChangerType_3
3
Middle-Aged man
TRTCVoiceChangerType_4
4
Heavy metal
TRTCVoiceChangerType_5
5
Nasal
TRTCVoiceChangerType_6
6
Punk
TRTCVoiceChangerType_7
7
Trapped beast
TRTCVoiceChangerType_8
8
Otaku
TRTCVoiceChangerType_9
9
Electronic
TRTCVoiceChangerType_10
10
Robot
TRTCVoiceChangerType_11
11
Ethereal
TRTCSystemVolumeType
TRTCSystemVolumeType
System volume type (only for mobile devices).Smartphones usually have two types of system volume: call volume and media volume.
 Call volume is designed for call scenarios. It comes with acoustic echo cancellation (AEC) and supports audio capturing by Bluetooth earphones, but its sound quality is average.
If you cannot turn the volume down to 0 (i.e., mute the phone) using the volume buttons, then your phone is using call volume.
 Media volume is designed for media scenarios such as music playback. AEC does not work when media volume is used, and Bluetooth earphones cannot be used for audio capturing. However, media volume delivers better music listening experience.
If you are able to mute your phone using the volume buttons, then your phone is using media volume.
﻿
The SDK offers three system volume control modes: auto, call volume, and media volume.
Enum
Value
DESC
TRTCSystemVolumeTypeAuto
0
Auto:
In the auto mode, call volume is used for anchors, and media volume for audience. This mode is suitable for live streaming scenarios.
If the scenario you select during  enterRoom  is  TRTCAppSceneLIVE  or  TRTCAppSceneVoiceChatRoom , the SDK will automatically use this mode.
TRTCSystemVolumeTypeMedia
1
Media volume:
In this mode, media volume is used in all scenarios. It is rarely used, mainly suitable for music scenarios with demanding requirements on audio quality.
Use this mode if most of your users use peripheral devices such as audio cards. Otherwise, it is not recommended.
TRTCSystemVolumeTypeVOIP
2
Call volume:
In this mode, the audio module does not change its work mode when users switch between anchors and audience, enabling seamless mic on/off. This mode is suitable for scenarios where users need to switch frequently between anchors and audience.
If the scenario you select during  enterRoom  is  TRTCAppSceneVideoCall  or  TRTCAppSceneAudioCall , the SDK will automatically use this mode.
TRTCAudioFrameOperationMode
TRTCAudioFrameOperationMode
Audio callback data operation mode.TRTC provides two modes of operation for audio callback data.
 Read-only mode (ReadOnly): Get audio data only from the callback.
 ReadWrite mode (ReadWrite): You can get and modify the audio data of the callback.
Enum
Value
DESC
TRTCAudioFrameOperationModeReadWrite
0
Read-write mode: You can get and modify the audio data of the callback, the default mode.
TRTCAudioFrameOperationModeReadOnly
1
Read-only mode: Get audio data from callback only.
TRTCLogLevel
TRTCLogLevel
Log level.Different log levels indicate different levels of details and number of logs. We recommend you set the log level to  TRTCLogLevelInfo  generally.
Enum
Value
DESC
TRTCLogLevelVerbose
0
Output logs at all levels
TRTCLogLevelDebug
1
Output logs at the DEBUG, INFO, WARNING, ERROR, and FATAL levels
TRTCLogLevelInfo
2
Output logs at the INFO, WARNING, ERROR, and FATAL levels
TRTCLogLevelWarn
3
Output logs at the WARNING, ERROR, and FATAL levels
TRTCLogLevelError
4
Output logs at the ERROR and FATAL levels
TRTCLogLevelFatal
5
Output logs at the FATAL level
TRTCLogLevelNone
6
Do not output any SDK logs
TRTCGSensorMode
TRTCGSensorMode
G-sensor switch (for mobile devices only).@deprecated Begin from v11.7 version，it is recommended to use the new gravity sensing enumeration ﻿TRTCGravitySensorAdaptiveMode﻿.
Enum
Value
DESC
TRTCGSensorMode_Disable
0
Do not adapt to G-sensor orientation
This mode is the default value for desktop platforms. In this mode, the video image published by the current user is not affected by the change of the G-sensor orientation.
TRTCGSensorMode_UIAutoLayout
1
Adapt to G-sensor orientation
This mode is the default value on mobile platforms. In this mode, the video image published by the current user is adjusted according to the G-sensor orientation, while the orientation of the local preview image remains unchanged.
One of the adaptation modes currently supported by the SDK is as follows: when the phone or tablet is upside down, in order to ensure that the screen orientation seen by the remote user is normal, the SDK will automatically rotate the published video image by 180 degrees.
If the UI layer of your application has enabled G-sensor adaption, we recommend you use the  UIFixLayout  mode.
TRTCGSensorMode_UIFixLayout
2
Adapt to G-sensor orientation
In this mode, the video image published by the current user is adjusted according to the G-sensor orientation, and the local preview image will also be rotated accordingly.
One of the features currently supported is as follows: when the phone or tablet is upside down, in order to ensure that the screen orientation seen by the remote user is normal, the SDK will automatically rotate the published video image by 180 degrees.
If the UI layer of your application doesn't support G-sensor adaption, but you want the video image in the SDK to adapt to the G-sensor orientation, we recommend you use the  UIFixLayout  mode.
@deprecated Begin from v11.5 version, it no longer supports TRTCGSensorMode_UIFixLayout and only supports the above two modes.
TRTCScreenCaptureSourceType
TRTCScreenCaptureSourceType
Screen sharing target type (for desktops only).Enum
Value
DESC
TRTCScreenCaptureSourceTypeUnknown
-1
Undefined
TRTCScreenCaptureSourceTypeWindow
0
The screen sharing target is the window of an application
TRTCScreenCaptureSourceTypeScreen
1
The screen sharing target is the entire screen
TRTCTranscodingConfigMode
TRTCTranscodingConfigMode
Layout mode of On-Cloud MixTranscoding.TRTC's On-Cloud MixTranscoding service can mix multiple audio/video streams in the room into one stream. Therefore, you need to specify the layout scheme of the video images. The following layout modes are provided:
Enum
Value
DESC
TRTCTranscodingConfigMode_Unknown
0
Undefined
TRTCTranscodingConfigMode_Manual
1
Manual layout mode
In this mode, you need to specify the precise position of each video image. This mode has the highest degree of freedom, but its ease of use is the worst:
 You need to enter all the parameters in ﻿TRTCTranscodingConfig﻿, including the position coordinates of each video image (TRTCMixUser).
 You need to listen on the ﻿onUserVideoAvailable﻿ and ﻿onUserAudioAvailable﻿ event callbacks in ﻿TRTCCloudDelegate﻿ and constantly adjust the  mixUsers  parameter according to the audio/video status of each user with mic on in the current room.
TRTCTranscodingConfigMode_Template_PureAudio
2
Pure audio mode
This mode is suitable for pure audio scenarios such as audio call (AudioCall) and audio 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 audio of all mic-on users in the room into the current user's live stream.
 You don't need to set the  mixUsers  parameter in ﻿TRTCTranscodingConfig﻿; instead, you only need to set the  audioSampleRate ,  audioBitrate  and  audioChannels  parameters.
TRTCTranscodingConfigMode_Template_PresetLayout
3
Preset layout mode
This is the most popular layout mode, because it allows you to set the position of each video image in advance through placeholders, and then the SDK automatically adjusts it dynamically according to the number of video images in the room.
In this mode, you still need to set the  mixUsers  parameter, but you can set  userId  as a "placeholder". Placeholder values include:
 "$PLACE_HOLDER_REMOTE$": image of remote user. Multiple images can be set.
 "$PLACE_HOLDER_LOCAL_MAIN$": local camera image. Only one image can be set.
 "$PLACE_HOLDER_LOCAL_SUB$": local screen sharing image. Only one image can be set.
In this mode, you don't need to listen on the ﻿onUserVideoAvailable﻿ and ﻿onUserAudioAvailable﻿ callbacks in ﻿TRTCCloudDelegate﻿ to make real-time adjustments.
Instead, you only need to call ﻿setMixTranscodingConfig﻿ once after successful room entry. Then, the SDK will automatically populate the placeholders you set with real  userId  values.
TRTCTranscodingConfigMode_Template_ScreenSharing
4
Screen sharing mode
This mode is suitable for screen sharing-based use cases such as online education and supported only by the SDKs for Windows and macOS.
In this mode, 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 teacher's camera image 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 layout mode 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 audio of mic-on students will be mixed into the teacher's audio/video stream by default.
Video content is primarily the shared screen in teaching mode, and it is a waste of bandwidth to transfer camera image and screen image at the same time.
Therefore, the recommended practice is to directly draw the camera image onto the current screen through the ﻿setLocalVideoRenderCallback﻿ API.
In this mode, you don't need to set the  mixUsers  parameter in ﻿TRTCTranscodingConfig﻿, and the SDK will not mix students' images so as not to interfere with the screen sharing effect.
You can set width x height in ﻿TRTCTranscodingConfig﻿ to 0 px x 0 px, and the SDK will automatically calculate a suitable resolution based on the aspect ratio of the user's current screen.
 If the teacher's current screen width is less than or equal to 1920 px, the SDK will use the actual resolution of the teacher's current screen.
 If the teacher's current screen width is greater than 1920 px, the SDK will select one of the three resolutions of 1920x1080 (16:9), 1920x1200 (16:10), and 1920x1440 (4:3) according to the current screen aspect ratio.
TRTCRecordType
TRTCRecordType
Media recording type.This enumerated type is used in the local media recording API ﻿startLocalRecording﻿ to specify whether to record audio/video files or pure audio files.
Enum
Value
DESC
TRTCRecordTypeAudio
0
Record audio only
TRTCRecordTypeVideo
1
Record video only
TRTCRecordTypeBoth
2
Record both audio and video
TRTCMixInputType
TRTCMixInputType
Stream mix input type.Enum
Value
DESC
TRTCMixInputTypeUndefined
0
Default.
Considering the compatibility with older versions, if you specify the inputType as Undefined, the SDK will determine the stream mix input type according to the value of the  pureAudio  parameter
TRTCMixInputTypeAudioVideo
1
Mix both audio and video
TRTCMixInputTypePureVideo
2
Mix video only
TRTCMixInputTypePureAudio
3
Mix audio only
TRTCMixInputTypeWatermark
4
Mix watermark
In this case, you don't need to specify the  userId  parameter, but you need to specify the  image  parameter. It is recommended to use png format.
TRTCAudioRecordingContent
TRTCAudioRecordingContent
Audio recording content type.This enumerated type is used in the audio recording API ﻿startAudioRecording﻿ to specify the content of the recorded audio.
Enum
Value
DESC
TRTCAudioRecordingContentAll
0
Record both local and remote audio
TRTCAudioRecordingContentLocal
1
Record local audio only
TRTCAudioRecordingContentRemote
2
Record remote audio only
TRTCPublishMode
TRTCPublishMode
The publishing mode.This enum type is used by the publishing API ﻿startPublishMediaStream﻿.
TRTC can mix multiple streams in a room and publish the mixed stream to a CDN or to a TRTC room. It can also publish the stream of the local user to Tencent Cloud or a third-party CDN.
You can specify one of the following publishing modes to use:
Enum
Value
DESC
TRTCPublishModeUnknown
0
Undefined
TRTCPublishBigStreamToCdn
1
Use this parameter to publish the primary stream (﻿TRTCVideoStreamTypeBig﻿) in the room to Tencent Cloud or a third-party CDN (only RTMP is supported).
TRTCPublishSubStreamToCdn
2
Use this parameter to publish the substream (﻿TRTCVideoStreamTypeSub﻿) in the room to Tencent Cloud or a third-party CDN (only RTMP is supported).
TRTCPublishMixStreamToCdn
3
Use this parameter together with the encoding parameter ﻿TRTCStreamEncoderParam﻿ and On-Cloud MixTranscoding parameter ﻿TRTCStreamMixingConfig﻿ to transcode the streams you specify and publish the mixed stream to Tencent Cloud or a third-party CDN (only RTMP is supported).
TRTCPublishMixStreamToRoom
4
Use this parameter together with the encoding parameter ﻿TRTCStreamEncoderParam﻿ and On-Cloud MixTranscoding parameter ﻿TRTCStreamMixingConfig﻿ to transcode the streams you specify and publish the mixed stream to the room you specify.
 Use  TRTCUser  in ﻿TRTCPublishTarget﻿ to specify the robot that publishes the transcoded stream to a TRTC room.
TRTCEncryptionAlgorithm
TRTCEncryptionAlgorithm
Encryption Algorithm.This enumeration type is used for media stream private encryption algorithm selection.
Enum
Value
DESC
TRTCEncryptionAlgorithmAes128Gcm
0
AES GCM 128。
TRTCEncryptionAlgorithmAes256Gcm
1
AES GCM 256。
TRTCSpeedTestScene
TRTCSpeedTestScene
Speed Test Scene.This enumeration type is used for speed test scene selection.
Enum
Value
DESC
TRTCSpeedTestScene_DelayTesting
1
Delay testing.
TRTCSpeedTestScene_DelayAndBandwidthTesting
2
Delay and bandwidth testing.
TRTCSpeedTestScene_OnlineChorusTesting
3
Online chorus testing.
TRTCGravitySensorAdaptiveMode
TRTCGravitySensorAdaptiveMode
Set the adaptation mode of gravity sensing (only applicable to mobile terminals).Begin from v11.7 version，It only takes effect when the camera capture scene inside SDK is used.
Enum
Value
DESC
TRTCGravitySensorAdaptiveMode_Disable
0
Turn off the gravity sensor and make a decision based on the current acquisition resolution and the set encoding resolution. If the two are inconsistent, rotate 90 degrees to ensure the maximum frame.
TRTCGravitySensorAdaptiveMode_FillByCenterCrop
1
Turn on the gravity sensor to always ensure that the remote screen image is positive. When the intermediate process needs to deal with inconsistent resolutions, use the center cropping mode.
TRTCGravitySensorAdaptiveMode_FitWithBlackBorder
2
Turn on the gravity sensor to always ensure that the remote screen image is positive. When the resolution needs to be processed inconsistently in the intermediate process, use the superimposed black border mode.
TRTCParams
TRTCParams
Room entry parameters.As the room entry parameters in the TRTC SDK, these parameters must be correctly set so that the user can successfully enter the audio/video room specified by  roomId  or  strRoomId .
For historical reasons, TRTC supports two types of room IDs:  roomId  and  strRoomId .
Note: do not mix  roomId  and  strRoomId , because they are not interchangeable. For example, the number  123  and the string  123  are two completely different rooms in TRTC.
EnumType
DESC
bussInfo
Field description: business data, which is optional. This field is needed only by some advanced features.
Recommended value: do not set this field on your own.
privateMapKey
Field description: permission credential used for permission control, which is optional. If you want only users with the specified  userId  values to enter a room, you need to use  privateMapKey  to restrict the permission.
Recommended value: we recommend you use this parameter only if you have high security requirements. For more information, please see ﻿Enabling Advanced Permission Control﻿.
role
Field description: role in the live streaming scenario, which is applicable only to the live streaming scenario (﻿TRTCAppSceneLIVE﻿ or ﻿TRTCAppSceneVoiceChatRoom﻿) but doesn't take effect in the call scenario.
Recommended value: default value: anchor (﻿TRTCRoleAnchor﻿).
roomId
Field description: numeric room ID. Users (userId) in the same room can see one another and make audio/video calls.
Recommended value: value range: 1–4294967294.
@note  roomId  and  strRoomId  are mutually exclusive. If you decide to use  strRoomId , then  roomId  should be entered as 0. If both are entered,  roomId  will be used.
Note
do not mix  roomId  and  strRoomId , because they are not interchangeable. For example, the number  123  and the string  123  are two completely different rooms in TRTC.
sdkAppId
Field description: application ID, which is required. Tencent Cloud generates bills based on  sdkAppId .
Recommended value: the ID can be obtained on the account information page in the ﻿TRTC console﻿ after the corresponding application is created.
strRoomId
Field description: string-type room ID. Users (userId) in the same room can see one another and make audio/video calls.
@note  roomId  and  strRoomId  are mutually exclusive. If you decide to use  strRoomId , then  roomId  should be entered as 0. If both are entered,  roomId  will be used.
Note
do not mix  roomId  and  strRoomId , because they are not interchangeable. For example, the number  123  and the string  123  are two completely different rooms in TRTC.
Recommended value: the length limit is 64 bytes. The following 89 characters are supported:
 Uppercase and lowercase letters (a–z and A–Z)
 Digits (0–9)
 Space, "!", "#", "$", "%", "&", "(", ")", "+", "-", ":", ";", "<", "=", ".", ">", "?", "@", "[", "]", "^", "_", "{", "}", "|", "~", and ",".
streamId
Field description: specified  streamId  in Tencent Cloud CSS, which is optional. After setting this field, you can play back the user's audio/video stream on Tencent Cloud CSS CDN through a standard pull scheme (FLV or HLS).
Recommended value: this parameter can contain up to 64 bytes and can be left empty. We recommend you use  sdkappid_roomid_userid_main  as the  streamid , which is easier to identify and will not cause conflicts in your multiple applications.
Note
to use Tencent Cloud CSS CDN, you need to enable the auto-relayed live streaming feature on the "Function Configuration" page in the ﻿console﻿ first.
For more information, please see ﻿Relay to CDN﻿.
userDefineRecordId
Field description: on-cloud recording field, which is optional and used to specify whether to record the user's audio/video stream in the cloud.
For more information, please see ﻿On-Cloud Recording﻿.
Recommended value: it can contain up to 64 bytes. Letters (a–z and A–Z), digits (0–9), underscores, and hyphens are allowed.
Scheme 1. Manual recording
1. Enable on-cloud recording in "Application Management" > "On-cloud Recording Configuration" in the ﻿console﻿.
2. Set "Recording Mode" to "Manual Recording".
3. After manual recording is set, in a TRTC room, only users with the  userDefineRecordId  parameter set will have video recording files in the cloud, while users without this parameter set will not.
4. The recording file will be named in the format of "userDefineRecordId_start time_end time" in the cloud.
Scheme 2. Auto-recording
1. You need to enable on-cloud recording in "Application Management" > "On-cloud Recording Configuration" in the ﻿console﻿.
2. Set "Recording Mode" to "Auto-recording".
3. After auto-recording is set, any user who upstreams audio/video in a TRTC room will have a video recording file in the cloud.
4. The file will be named in the format of "userDefineRecordId_start time_end time". If  userDefineRecordId  is not specified, the file will be named in the format of "streamId_start time_end time".
userId
Field description: user ID, which is required. It is the  userId  of the local user in UTF-8 encoding and acts as the username.
Recommended value: if the ID of a user in your account system is "mike",  userId  can be set to "mike".
userSig
Field description: user signature, which is required. It is the authentication signature corresponding to the current  userId  and acts as the login password for Tencent Cloud services.
Recommended value: for the calculation method, please see ﻿UserSig﻿.
TRTCVideoEncParam
TRTCVideoEncParam
Video encoding parameters.These settings determine the quality of image viewed by remote users as well as the image quality of recorded video files in the cloud.
EnumType
DESC
enableAdjustRes
Field description: whether to allow dynamic resolution adjustment. Once enabled, this field will affect on-cloud recording.
Recommended value: this feature is suitable for scenarios that don't require on-cloud recording. After it is enabled, the SDK will intelligently select a suitable resolution according to the current network conditions to avoid the inefficient encoding mode of "large resolution + small bitrate".
Note
default value: NO. If you need on-cloud recording, please do not enable this feature, because if the video resolution changes, the MP4 file recorded in the cloud cannot be played back normally by common players.
minVideoBitrate
Field description: minimum video bitrate. The SDK will reduce the bitrate to as low as the value specified by  minVideoBitrate  to ensure the smoothness only if the network conditions are poor.
Note: default value: 0, indicating that a reasonable value of the lowest bitrate will be automatically calculated by the SDK according to the resolution you specify.
Recommended value: you can set the  videoBitrate  and  minVideoBitrate  parameters at the same time to restrict the SDK's adjustment range of the video bitrate:
 If you want to "ensure clarity while allowing lag in weak network environments", you can set  minVideoBitrate  to 60% of  videoBitrate .
 If you want to "ensure smoothness while allowing blur in weak network environments", you can set  minVideoBitrate  to a low value, for example, 100 Kbps.
 If you set  videoBitrate  and  minVideoBitrate  to the same value, it is equivalent to disabling the adaptive adjustment capability of the SDK for the video bitrate.
resMode
Field description: resolution mode (landscape/portrait)
Recommended value: for mobile platforms (iOS and Android),  Portrait  is recommended; for desktop platforms (Windows and macOS),  Landscape  is recommended.
Note
to use a portrait resolution, please specify  resMode  as  Portrait ; for example, when used together with  Portrait , 640x360 represents 360x640.
videoBitrate
Field description: target video bitrate. The SDK encodes streams at the target video bitrate and will actively reduce the bitrate only in weak network environments.
Recommended value: please see the optimal bitrate for each specification in  TRTCVideoResolution . You can also slightly increase the optimal bitrate.
For example,  TRTCVideoResolution_1280_720  corresponds to the target bitrate of 1,200 Kbps. You can also set the bitrate to 1,500 Kbps for higher definition.
Note
you can set the  videoBitrate  and  minVideoBitrate  parameters at the same time to restrict the SDK's adjustment range of the video bitrate:
 If you want to "ensure clarity while allowing lag in weak network environments", you can set  minVideoBitrate  to 60% of  videoBitrate .
 If you want to "ensure smoothness while allowing blur in weak network environments", you can set  minVideoBitrate  to a low value, for example, 100 Kbps.
 If you set  videoBitrate  and  minVideoBitrate  to the same value, it is equivalent to disabling the adaptive adjustment capability of the SDK for the video bitrate.
videoFps
Field description: video capturing frame rate
Recommended value: 15 or 20 fps. If the frame rate is lower than 5 fps, there will be obvious lagging; if lower than 10 fps but higher than 5 fps, there will be slight lagging; if higher than 20 fps, the bandwidth will be wasted (the frame rate of movies is generally 24 fps).
Note
the front cameras on certain Android phones do not support a capturing frame rate higher than 15 fps. For some Android phones that focus on beautification features, the capturing frame rate of the front cameras may be lower than 10 fps.
videoResolution
Field description: video resolution
Recommended value
 For mobile video call, we recommend you select a resolution of 360x640 or below and select  Portrait  (portrait resolution) for  resMode .
 For mobile live streaming, we recommend you select a resolution of 540x960 and select  Portrait  (portrait resolution) for  resMode .
 For desktop platforms (Windows and macOS), we recommend you select a resolution of 640x360 or above and select  Landscape  (landscape resolution) for  resMode .
Note
to use a portrait resolution, please specify  resMode  as  Portrait ; for example, when used together with  Portrait , 640x360 represents 360x640.
TRTCNetworkQosParam
TRTCNetworkQosParam
Network QoS control parameter set.Network QoS control parameter. The settings determine the QoS control policy of the SDK in weak network conditions (e.g., whether to "ensure clarity" or "ensure smoothness").
EnumType
DESC
controlMode
Field description: QoS control mode (disused)
Recommended value: on-cloud control
Note
please set the on-cloud control mode (TRTCQosControlModeServer).
preference
Field description: whether to ensure smoothness or clarity
Recommended value: ensuring clarity
Note
this parameter mainly affects the audio/video performance of TRTC in weak network environments:
 Ensuring smoothness: in this mode, when the current network is unable to transfer a clear and smooth video image, the smoothness of the image will be given priority, but there will be blurs. See ﻿TRTCVideoQosPreferenceSmooth﻿
 Ensuring clarity (default value): in this mode, when the current network is unable to transfer a clear and smooth video image, the clarity of the image will be given priority, but there will be lags. See ﻿TRTCVideoQosPreferenceClear﻿
TRTCRenderParams
TRTCRenderParams
Rendering parameters of video image.You can use these parameters to control the video image rotation angle, fill mode, and mirror mode.
EnumType
DESC
fillMode
Field description: image fill mode
Recommended value: fill (the image may be stretched or cropped) or fit (there may be black bars in unmatched areas). Default value: ﻿TRTCVideoFillMode_Fill﻿
mirrorType
Field description: image mirror mode
Recommended value: default value: ﻿TRTCVideoMirrorTypeAuto﻿
rotation
Field description: clockwise image rotation angle
Recommended value: rotation angles of 90, 180, and 270 degrees are supported. Default value: ﻿TRTCVideoRotation_0﻿
TRTCQualityInfo
TRTCQualityInfo
Network quality.This indicates the quality of the network. You can use it to display the network quality of each user on the UI.
EnumType
DESC
quality
Network quality
userId
User ID
TRTCVolumeInfo
TRTCVolumeInfo
Volume.This indicates the audio volume value. You can use it to display the volume of each user in the UI.
EnumType
DESC
pitch
The local user's vocal frequency (unit: Hz), the value range is [0 - 4000]. For remote users, this value is always 0.
spectrumData
Audio spectrum data, which divides the sound frequency into 256 frequency domains, spectrumData records the energy value of each frequency domain,
The value range of each energy value is [-300, 0] in dBFS.
Note
The local spectrum is calculated using the audio data before encoding, which will be affected by the capture volume, BGM, etc.; the remote spectrum is calculated using the received audio data, and operations such as adjusting the remote playback volume locally will not affect it.
userId
 userId  of the speaker. An empty value indicates the local user.
vad
Vad result of the local user. 0: not speech 1: speech.
volume
Volume of the speaker. Value range: 0–100.
TRTCSpeedTestParams
TRTCSpeedTestParams
Network speed testing parameters.You can test the network speed through the ﻿startSpeedTest:﻿ interface before the user enters the room (this API cannot be called during a call).
EnumType
DESC
expectedDownBandwidth
Expected downstream bandwidth (kbps, value range: 10 to 5000, no downlink bandwidth test when it is 0).
Note
When the parameter  scene  is set to ﻿TRTCSpeedTestScene_OnlineChorusTesting﻿, in order to obtain more accurate information such as  rtt / jitter , the value range is limited to [10, 1000].
expectedUpBandwidth
Expected upstream bandwidth (kbps, value range: 10 to 5000, no uplink bandwidth test when it is 0).
Note
When the parameter  scene  is set to ﻿TRTCSpeedTestScene_OnlineChorusTesting﻿, in order to obtain more accurate information such as  rtt / jitter , the value range is limited to [10, 1000].
scene
Speed test scene.
sdkAppId
Application identification, please refer to the relevant instructions in ﻿TRTCParams﻿.
userId
User identification, please refer to the relevant instructions in ﻿TRTCParams﻿.
userSig
User signature, please refer to the relevant instructions in ﻿TRTCParams﻿.
TRTCSpeedTestResult
TRTCSpeedTestResult
Network speed test result.The ﻿startSpeedTest:﻿ API can be used to test the network speed before a user enters a room (this API cannot be called during a call).
EnumType
DESC
availableDownBandwidth
Downstream bandwidth (in kbps, -1: invalid value).
availableUpBandwidth
Upstream bandwidth (in kbps, -1: invalid value).
downJitter
Downlink data packet jitter (ms) refers to the stability of data communication in the user's current network environment. The smaller the value, the better. The normal value range is [0, 100]. -1 means that the speed test failed to obtain an effective value. Generally, the Jitter of the WiFi network will be slightly larger than that of the 4G/5G environment.
downLostRate
Downstream packet loss rate between 0 and 1.0. For example, 0.2 indicates that 2 data packets may be lost in every 10 packets received from the server.
errMsg
Error message for network speed test.
ip
Server IP address.
quality
Network quality, which is tested and calculated based on the internal evaluation algorithm. For more information, please see ﻿TRTCQuality﻿
rtt
Delay in milliseconds, which is the round-trip time between the current device and TRTC server. The smaller the value, the better. The normal value range is 10–100 ms.
success
Whether the network speed test is successful.
upJitter
Uplink data packet jitter (ms) refers to the stability of data communication in the user's current network environment. The smaller the value, the better. The normal value range is [0, 100]. -1 means that the speed test failed to obtain an effective value. Generally, the Jitter of the WiFi network will be slightly larger than that of the 4G/5G environment.
upLostRate
Upstream packet loss rate between 0 and 1.0. For example, 0.3 indicates that 3 data packets may be lost in every 10 packets sent to the server.
TRTCVideoFrame
TRTCVideoFrame
Video frame information. TRTCVideoFrame  is used to describe the raw data of a frame of the video image, which is the image data before frame encoding or after frame decoding.
EnumType
DESC
bufferType
Field description: video data structure type
data
Field description: video data when  bufferType  is ﻿TRTCVideoBufferType_NSData﻿, which carries the memory data blocks in  NSData  type.
height
Field description: video height
Recommended value: please enter the height of the video data passed in.
pixelBuffer
Field description: video data when  bufferType  is ﻿TRTCVideoBufferType_PixelBuffer﻿, which carries the  PixelBuffer  unique to iOS.
pixelFormat
Field description: video pixel format
rotation
Field description: clockwise rotation angle of video pixels
textureId
Field description: video texture ID, i.e., video data when  bufferType  is ﻿TRTCVideoBufferType_Texture﻿, which carries the texture data used for OpenGL rendering.
timestamp
Field description: video frame timestamp in milliseconds
Recommended value: this parameter can be set to 0 for custom video capturing. In this case, the SDK will automatically set the  timestamp  field. However, please "evenly" set the calling interval of ﻿sendCustomVideoData﻿.
width
Field description: video width
Recommended value: please enter the width of the video data passed in.
TRTCAudioFrame
TRTCAudioFrame
Audio frame data.EnumType
DESC
channels
Field description: number of sound channels
data
Field description: audio data
extraData
Field description: extra data in audio frame, message sent by remote users through ﻿onLocalProcessedAudioFrame﻿ that add to audio frame will be callback through this field.
sampleRate
Field description: sample rate
timestamp
Field description: timestamp in ms
TRTCMixUser
TRTCMixUser
Description information of each video image in On-Cloud MixTranscoding. TRTCMixUser  is used to specify the location, size, layer, and stream type of each video image in On-Cloud MixTranscoding.
EnumType
DESC
image
Field description: specify the placeholder or watermark image. The placeholder image will be displayed when there is no upstream video.A watermark image is a semi-transparent image posted in the mixed image, and this image will always be overlaid on the mixed image.
 When the  inputType  field is set to ﻿TRTCMixInputTypePureAudio﻿, the image is a placeholder image, and you need to specify  userId .
 When the  inputType  field is set to ﻿TRTCMixInputTypeWatermark﻿, the image is a watermark image, and you don't need to specify  userId .
Recommended value: default value: null, indicating not to set the placeholder or watermark image.
Note
TRTC's backend service will mix the image specified by the URL address into the final stream.URL link length is limited to 512 bytes. The image size is limited to 10MB.Support png, jpg, jpeg, bmp format. Take effects iff the  inputType  field is set to ﻿TRTCMixInputTypePureAudio﻿ or ﻿TRTCMixInputTypeWatermark﻿.
inputType
Field description: specify the mixed content of this stream (audio only, video only, audio and video, or watermark).
Recommended value: default value: ﻿TRTCMixInputTypeUndefined﻿.
Note
 When specifying  inputType  as ﻿TRTCMixInputTypeUndefined﻿ and specifying  pureAudio  to YES, it is equivalent to setting  inputType  to  TRTCMixInputTypePureAudio .
 When specifying  inputType  as ﻿TRTCMixInputTypeUndefined﻿ and specifying  pureAudio  to NO, it is equivalent to setting  inputType  to  TRTCMixInputTypeAudioVideo .
 When specifying  inputType  as ﻿TRTCMixInputTypeWatermark﻿, you don't need to specify the  userId  field, but you need to specify the  image  field.
pureAudio
Field description: specify whether this stream mixes audio only
Recommended value: default value: NO
Note
this field has been disused. We recommend you use the new field  inputType  introduced in v8.5.
rect
Field description: specify the coordinate area of this video image in px
renderMode
Field description: specify the display mode of this stream.
Recommended value: default value: 0. 0 is cropping, 1 is zooming, 2 is zooming and displaying black background.
Note
image doesn't support setting  renderMode  temporarily, the default display mode is forced stretch.
roomID
Field description: ID of the room where this audio/video stream is located (an empty value indicates the local room ID)
soundLevel
Field description: specify the target volumn level of On-Cloud MixTranscoding. (value range: 0-100)
Recommended value: default value: 100.
streamType
Field description: specify whether this video image is the primary stream image (﻿TRTCVideoStreamTypeBig﻿) or substream image (﻿TRTCVideoStreamTypeSub﻿).
userId
Field description: user ID
zOrder
Field description: specify the level of this video image (value range: [1, 15]; the value must be unique)
TRTCTranscodingConfig
TRTCTranscodingConfig
Layout and transcoding parameters of On-Cloud MixTranscoding.These parameters are used to specify the layout position information of each video image and the encoding parameters of mixtranscoding during On-Cloud MixTranscoding.
EnumType
DESC
appId
Field description:  appId  of Tencent Cloud CSS
Recommended value: please click  Application Management  >  Application Information  in the ﻿TRTC console﻿ and get the  appId  in  Relayed Live Streaming Info .
Note
applications created on or after January 9, 2020 do not need to fill in this field.
audioBitrate
Field description: specify the target audio bitrate of On-Cloud MixTranscoding
Recommended value: default value: 64 Kbps. Value range: [32,192].
audioChannels
Field description: specify the number of sound channels of On-Cloud MixTranscoding
Recommended value: default value: 1, which means mono channel. Valid values: 1: mono channel; 2: dual channel.
audioCodec
Field description: specify the audio encoding type of On-Cloud MixTranscoding
Recommended value: default value: 0, which means LC-AAC. Valid values: 0:  LC-AAC; 1: HE-AAC; 2: HE-AACv2.
Note
 HE-AAC and HE-AACv2 only support [48000, 44100, 32000, 24000, 16000]  sample rate.
 HE-AACv2  only support dual channel.
 HE-AAC and HE-AACv2 take effects iff the output streamId is specified.
audioSampleRate
Field description: specify the target audio sample rate of On-Cloud MixTranscoding
Recommended value: default value: 48000 Hz. Valid values: 12000 Hz, 16000 Hz, 22050 Hz, 24000 Hz, 32000 Hz, 44100 Hz, 48000 Hz.
backgroundColor
Field description: specify the background color of the mixed video image.
Recommended value: default value: 0x000000, which means black and is in the format of hex number; for example: "0x61B9F1" represents the RGB color (97,158,241).
backgroundImage
Field description: specify the background image of the mixed video image.
**Recommended value: default value: null, indicating not to set the background image.
Note
TRTC's backend service will mix the image specified by the URL address into the final stream.URL link length is limited to 512 bytes. The image size is limited to 10MB.Support png, jpg, jpeg, bmp format.
bizId
Field description:  bizId  of Tencent Cloud CSS
Recommended value: please click  Application Management  >  Application Information  in the ﻿TRTC console﻿ and get the  bizId  in  Relayed Live Streaming Info .
Note
applications created on or after January 9, 2020 do not need to fill in this field.
mixUsers
Field description: specify the position, size, layer, and stream type of each video image in On-Cloud MixTranscoding
Recommended value: this field is an array in  TRTCMixUser  type, where each element represents the information of a video image.
mode
Field description: layout mode
Recommended value: please choose a value according to your business needs. The preset mode has better applicability.
streamId
Field description: ID of the live stream output to CDN
Recommended value: default value: null, that is, the audio/video streams in the room will be mixed into the audio/video stream of the caller of this API.
 If you don't set this parameter, the SDK will execute the default logic, that is, it will mix the multiple audio/video streams in the room into the audio/video stream of the caller of this API, i.e., A + B => A.
 If you set this parameter, the SDK will mix the audio/video streams in the room into the live stream you specify, i.e., A + B => C (C is the  streamId  you specify).
videoBitrate
Field description: specify the target video bitrate (Kbps) of On-Cloud MixTranscoding
Recommended value: if you enter 0, TRTC will estimate a reasonable bitrate value based on  videoWidth  and  videoHeight . You can also refer to the recommended bitrate value in the video resolution enumeration definition (in the comment section).
videoFramerate
Field description: specify the target video frame rate (fps) of On-Cloud MixTranscoding
Recommended value: default value: 15 fps. Value range: (0,30].
videoGOP
Field description: specify the target video keyframe interval (GOP) of On-Cloud MixTranscoding
Recommended value: default value: 2 (in seconds). Value range: [1,8].
videoHeight
Field description: specify the target resolution (height) of On-Cloud MixTranscoding
Recommended value: 640 px. If you only mix audio streams, please set both  width  and  height  to 0; otherwise, there will be a black background in the live stream after mixtranscoding.
videoSeiParams
Field description: SEI parameters. default value: null
Note
the parameter is passed in the form of a JSON string. Here is an example to use it:
The currently supported fields and their meanings are as follows:
 payloadContent: Required. The payload content of the passthrough SEI, which cannot be empty.
 payloadType: Required. The type of the SEI message, with a value range of 5 or an integer within the range of [100, 254] (excluding 244, which is an internally defined timestamp SEI).
 payloadUuid: Required when payloadType is 5, and ignored in other cases. The value must be a 32-digit hexadecimal number.
 interval: Optional, default is 1000. The sending interval of the SEI, in milliseconds.
 followIdr: Optional, default is false. When this value is true, the SEI will be ensured to be carried when sending a key frame, otherwise it is not guaranteed.
videoWidth
Field description: specify the target resolution (width) of On-Cloud MixTranscoding
Recommended value: 360 px. If you only mix audio streams, please set both  width  and  height  to 0; otherwise, there will be a black background in the live stream after mixtranscoding.
TRTCPublishCDNParam
TRTCPublishCDNParam
Push parameters required to be set when publishing audio/video streams to non-Tencent Cloud CDN.TRTC's backend service supports publishing audio/video streams to third-party live CDN service providers through the standard RTMP protocol.
If you use the Tencent Cloud CSS CDN service, you don't need to care about this parameter; instead, just use the ﻿startPublish﻿ API.
EnumType
DESC
appId
Field description:  appId  of Tencent Cloud CSS
Recommended value: please click  Application Management  >  Application Information  in the ﻿TRTC console﻿ and get the  appId  in  Relayed Live Streaming Info .
bizId
Field description:  bizId  of Tencent Cloud CSS
Recommended value: please click  Application Management  >  Application Information  in the ﻿TRTC console﻿ and get the  bizId  in  Relayed Live Streaming Info .
streamId
Field description: specify the push address (in RTMP format) of this audio/video stream at the third-party live streaming service provider
Recommended value: default value: null,that is, the audio/video streams in the room will be pushed to the target service provider of the caller of this API.
url
Field description: specify the push address (in RTMP format) of this audio/video stream at the third-party live streaming service provider
Recommended value: the push URL rules vary greatly by service provider. Please enter a valid push URL according to the requirements of the target service provider. TRTC's backend server will push audio/video streams in the standard format to the third-party service provider according to the URL you enter.
Note
the push URL must be in RTMP format and meet the specifications of your target live streaming service provider; otherwise, the target service provider will reject the push requests from TRTC's backend service.
TRTCAudioRecordingParams
TRTCAudioRecordingParams
Local audio file recording parameters.This parameter is used to specify the recording parameters in the audio recording API ﻿startAudioRecording﻿.
EnumType
DESC
filePath
Field description: storage path of the audio recording file, which is required.
Note
this path must be accurate to the file name and extension. The extension determines the format of the audio recording file. Currently, supported formats include PCM, WAV, and AAC.
For example, if you specify the path as  mypath/record/audio.aac , it means that you want the SDK to generate an audio recording file in AAC format.Please specify a valid path with read/write permissions; otherwise, the audio recording file cannot be generated.
maxDurationPerFile
Field description:  maxDurationPerFile  is the max duration of each recorded file segments, in milliseconds, with a minimum value of 10000. The default value is 0, indicating no segmentation.
recordingContent
Field description: Audio recording content type.
Note: Record all local and remote audio by default.
TRTCLocalRecordingParams
TRTCLocalRecordingParams
Local media file recording parameters.This parameter is used to specify the recording parameters in the local media file recording API ﻿startLocalRecording﻿.
The ﻿startLocalRecording﻿ API is an enhanced version of the ﻿startAudioRecording﻿ API. The former can record video files, while the latter can only record audio files.
EnumType
DESC
filePath
Field description: address of the recording file, which is required. Please ensure that the path is valid with read/write permissions; otherwise, the recording file cannot be generated.
Note
this path must be accurate to the file name and extension. The extension determines the format of the recording file. Currently, only the MP4 format is supported.
For example, if you specify the path as  mypath/record/test.mp4 , it means that you want the SDK to generate a local video file in MP4 format.
Please specify a valid path with read/write permissions; otherwise, the recording file cannot be generated.
interval
Field description:  interval  is the update frequency of the recording information in milliseconds. Value range: [1000, 10000]. Default value: -1, indicating not to call back
maxDurationPerFile
Field description:  maxDurationPerFile  is the max duration of each recorded file segments, in milliseconds, with a minimum value of 10000. The default value is 0, indicating no segmentation.
recordType
Field description: media recording type, which is  TRTCRecordTypeBoth  by default, indicating to record both audio and video.
TRTCAudioEffectParam(Deprecated)
TRTCAudioEffectParam(Deprecated)
Sound effect parameter (disused)."Sound effects" in TRTC refer to some short audio files (usually only a few seconds), such as "applause" and "laughter".
This parameter is used to specify the path and number of playback times of a sound effect file (short audio file) in the sound effect playback API ﻿TRTCCloud#playAudioEffect﻿ on legacy versions.
After v7.3, the sound effect API has been replaced by a new ﻿TXAudioEffectManager#startPlayMusic﻿ API.
When you specify the ﻿TXAudioMusicParam﻿ parameter of  startPlayMusic , if  isShortFile  is set to  YES , the file is a "sound effect" file.
EnumType
DESC
effectId
Field description: sound effect ID
Note: the SDK supports playing multiple sound effects. IDs are used to distinguish different sound effects and control their start, end, volume, etc.
loopCount
Field description: number of times the sound effect is looped
Valid values: 0 or any positive integer. 0 (default) indicates that the sound effect is played once, 1 twice, and so on.
path
Field description: sound effect file path. Supported file formats include AAC, MP3, and M4A.
publish
Field description: whether the sound effect is upstreamed
Recommended value: YES: when the sound effect is played back locally, it will be upstreamed to the cloud and can be heard by remote users. NO: the sound effect will not be upstreamed to the cloud and can only be heard locally. Default value: NO
volume
Field description: sound effect volume
Recommended value: value range: 0–100. Default value: 100
TRTCSwitchRoomConfig
TRTCSwitchRoomConfig
Room switch parameter.This parameter is used for the room switch API ﻿switchRoom﻿, which can quickly switch a user from one room to another.
EnumType
DESC
privateMapKey
Field description: permission credential used for permission control, which is optional. If you want only users with the specified  userId  values to enter a room, you need to use  privateMapKey  to restrict the permission.
Recommended value: we recommend you use this parameter only if you have high security requirements. For more information, please see ﻿Enabling Advanced Permission Control﻿.
roomId
Field description: numeric room ID, which is optional. Users in the same room can see one another and make audio/video calls.
Recommended value: value range: 1–4294967294.
Note
either  roomId  or  strRoomId  must be entered. If both are entered,  roomId  will be used.
strRoomId
Field description: string-type room ID, which is optional. Users in the same room can see one another and make audio/video calls.
Note
either  roomId  or  strRoomId  must be entered. If both are entered,  roomId  will be used.
userSig
Field description: user signature, which is optional. It is the authentication signature corresponding to the current  userId  and acts as the login password.
If you don't specify the newly calculated  userSig  during room switch, the SDK will continue to use the  userSig  you specified during room entry (enterRoom).
This requires you to ensure that the old  userSig  is still within the validity period allowed by the signature at the moment of room switch; otherwise, room switch will fail.
Recommended value: for the calculation method, please see ﻿UserSig﻿.
TRTCAudioFrameDelegateFormat
TRTCAudioFrameDelegateFormat
Format parameter of custom audio callback.This parameter is used to set the relevant format (including sample rate and number of channels) of the audio data called back by the SDK in the APIs related to custom audio callback.
EnumType
DESC
channels
Field description: number of sound channels
Recommended value: default value: 1, which means mono channel. Valid values: 1: mono channel; 2: dual channel.
mode
Field description: audio callback data operation mode
Recommended value: ﻿TRTCAudioFrameOperationModeReadOnly﻿, get audio data from callback only. The modes that can be set are ﻿TRTCAudioFrameOperationModeReadOnly﻿, ﻿TRTCAudioFrameOperationModeReadWrite﻿.
sampleRate
Field description: sample rate
Recommended value: default value: 48000 Hz. Valid values: 16000, 32000, 44100, 48000.
samplesPerCall
Field description: number of sample points
Recommended value: the value must be an integer multiple of  sampleRate/100 .
TRTCUser
TRTCUser
The users whose streams to publish.You can use this parameter together with the publishing destination parameter ﻿TRTCPublishTarget﻿ and On-Cloud MixTranscoding parameter ﻿TRTCStreamMixingConfig﻿ to transcode the streams you specify and publish the mixed stream to the destination you specify.
EnumType
DESC
intRoomId
Description: Numeric room ID. The room ID must be of the same type as that in ﻿TRTCParams﻿.
Value: Value range: 1-4294967294
Note: You cannot use both  intRoomId  and  strRoomId . If you specify  strRoomId , you need to set  intRoomId  to  0 . If you set both, only  intRoomId  will be used.
strRoomId
Description: String-type room ID. The room ID must be of the same type as that in ﻿TRTCParams﻿.
Note: You cannot use both  intRoomId  and  strRoomId . If you specify  roomId , you need to leave  strRoomId  empty. If you set both, only  intRoomId  will be used.
Value: 64 bytes or shorter; supports the following character set (89 characters):
 Uppercase and lowercase letters (a-z and A-Z)
 Numbers (0-9)
 Space, "!", "#", "$", "%", "&", "(", ")", "+", "-", ":", ";", "<", "=", ".", ">", "?", "@", "[", "]", "^", "_", "{", "}", "|", "~", ","
userId
/Description: UTF-8-encoded user ID (required)
Value: For example, if the ID of a user in your account system is "mike", set it to  mike .
TRTCPublishCdnUrl
TRTCPublishCdnUrl
The destination URL when you publish to Tencent Cloud or a third-party CDN.This enum type is used by the publishing destination parameter ﻿TRTCPublishTarget﻿ of the publishing API ﻿startPublishMediaStream﻿.
EnumType
DESC
isInternalLine
Description: Whether to publish to Tencent Cloud
Value: The default value is  true .
Note: If the destination URL you set is provided by Tencent Cloud, set this parameter to  true , and you will not be charged relaying fees.
rtmpUrl
Description: The destination URL (RTMP) when you publish to Tencent Cloud or a third-party CDN.
Value: The URLs of different CDN providers may vary greatly in format. Please enter a valid URL as required by your service provider. TRTC's backend server will push audio/video streams in the standard format to the URL you provide.
Note: The URL must be in RTMP format. It must also meet the requirements of your service provider, or your service provider may reject push requests from the TRTC backend.
TRTCPublishTarget
TRTCPublishTarget
The publishing destination.This enum type is used by the publishing API ﻿startPublishMediaStream﻿.
EnumType
DESC
cdnUrlList
 Description:  The destination URLs (RTMP) when you publish to Tencent Cloud or third-party CDNs.
 Note:  You don’t need to set this parameter if you set the publishing mode to ﻿TRTCPublishMixStreamToRoom﻿.
mixStreamIdentity
 Description:  The information of the robot that publishes the transcoded stream to a TRTC room.
 Note:  You need to set this parameter only if you set the publishing mode to ﻿TRTCPublishMixStreamToRoom﻿`.
 Note:  After you set this parameter, the stream will be pushed to the room you specify. We recommend you set it to a special user ID to distinguish the robot from the anchor who enters the room via the TRTC SDK.
 Note:  Users whose streams are transcoded cannot subscribe to the transcoded stream.
 Note:  If you set the subscription mode (﻿setDefaultStreamRecvMode﻿) to manual before room entry, you need to manage the streams to receive by yourself (normally, if you receive the transcoded stream, you need to unsubscribe from the streams that are transcoded).
 Note:  If you set the subscription mode (﻿setDefaultStreamRecvMode﻿) to auto before room entry, users whose streams are not transcoded will receive the transcoded stream automatically and will unsubscribe from the users whose streams are transcoded. You call ﻿muteRemoteVideoStream﻿ and ﻿muteRemoteAudio﻿ to unsubscribe from the transcoded stream.
mode
 Description:  The publishing mode.
 Value:  You can relay streams to a CDN, transcode streams, or publish streams to an RTC room. Select the mode that fits your needs.
Note
If you need to use more than one publishing mode, you can call ﻿startPublishMediaStream﻿ multiple times and set  TRTCPublishTarget  to a different value each time.You can use one mode each time you call the ﻿startPublishMediaStream﻿) API. To modify the configuration, call ﻿updatePublishCDNStream﻿.
TRTCVideoLayout
TRTCVideoLayout
The video layout of the transcoded stream.This enum type is used by the On-Cloud MixTranscoding parameter ﻿TRTCStreamMixingConfig﻿ of the publishing API ﻿startPublishMediaStream﻿.
You can use this parameter to specify the position, size, layer, and stream type of each video in the transcoded stream.
EnumType
DESC
backgroundColor
 Description:  The background color of the mixed stream.
 Value:  The value must be a hex number. For example, "0x61B9F1" represents the RGB color value (97,158,241). Default value: 0x000000 (black).
fillMode
 Description:  The rendering mode.
 Value:  The rendering mode may be fill (the image may be stretched or cropped) or fit (there may be black bars). Default value: ﻿TRTCVideoFillMode_Fill﻿.
fixedVideoStreamType
 Description:  Whether the video is the primary stream (﻿TRTCVideoStreamTypeBig﻿) or substream (﻿TRTCVideoStreamTypeSub﻿).
fixedVideoUser
 Description:  The users whose streams are transcoded.
Note
If you do not specify ﻿TRTCUser﻿ ( userId ,  intRoomId ,  strRoomId ), the TRTC backend will automatically mix the streams of anchors who are sending audio/video in the room according to the video layout you specify.
placeHolderImage
 Description:  The URL of the placeholder image. If a user sends only audio, the image specified by the URL will be mixed during On-Cloud MixTranscoding.
 Value:  This parameter is left empty by default, which means no placeholder image will be used.
Note
 You need to specify the  userId  parameter in  fixedVideoUser .
 The URL can be 512 bytes long at most, and the image must not exceed 2 MB.
 The image can be in PNG, JPG, JPEG, or BMP format. We recommend you use a semitransparent image in PNG format.
rect
 Description:  The coordinates (in pixels) of the video.
zOrder
 Description:  The layer of the video, which must be unique. Value range: 0-15.
TRTCWatermark
TRTCWatermark
The watermark layout.This enum type is used by the On-Cloud MixTranscoding parameter ﻿TRTCStreamMixingConfig﻿ of the publishing API ﻿startPublishMediaStream﻿.
EnumType
DESC
rect
 Description:  The coordinates (in pixels) of the watermark.
watermarkUrl
 Description:  The URL of the watermark image. The image specified by the URL will be mixed during On-Cloud MixTranscoding.
Note
 The URL can be 512 bytes long at most, and the image must not exceed 2 MB.
 The image can be in PNG, JPG, JPEG, or BMP format. We recommend you use a semitransparent image in PNG format.
zOrder
 Description:  The layer of the watermark, which must be unique. Value range: 0-15.
TRTCStreamEncoderParam
TRTCStreamEncoderParam
The encoding parameters. Description:  This enum type is used by the publishing API ﻿startPublishMediaStream﻿.
 Note:  This parameter is required if you set the publishing mode to  TRTCPublish_MixStream_ToCdn  or  TRTCPublish_MixStream_ToRoom  in ﻿TRTCPublishTarget﻿.
 Note:  If you use the relay to CDN feature (the publishing mode set to  RTCPublish_BigStream_ToCdn  or  TRTCPublish_SubStream_ToCdn ), to improve the relaying stability and playback compatibility, we also recommend you set this parameter.
EnumType
DESC
audioEncodedChannelNum
 Description:  The sound channels of the stream to publish.
 Value:  Valid values: 1 (mono channel); 2 (dual-channel). Default: 1.
audioEncodedCodecType
 Description:  The audio codec of the stream to publish.
 Value:  Valid values: 0 (LC-AAC); 1 (HE-AAC); 2 (HE-AACv2). Default: 0.
Note
 The audio sample rates supported by HE-AAC and HE-AACv2 are 48000, 44100, 32000, 24000, and 16000.
 When HE-AACv2 is used, the output stream can only be dual-channel.
audioEncodedKbps
 Description:  The audio bitrate (Kbps) of the stream to publish.
 Value:  Value range: [32,192]. Default: 50.
audioEncodedSampleRate
 Description:  The audio sample rate of the stream to publish.
 Value:  Valid values: [48000, 44100, 32000, 24000, 16000, 8000]. Default: 48000 (Hz).
videoEncodedCodecType
 Description:  The video codec of the stream to publish.
 Value:  Valid values: 0 (H264); 1 (H265). Default: 0.
videoEncodedFPS
 Description:  The frame rate (fps) of the stream to publish.
 Value:  Value range: (0,30]. Default: 20.
videoEncodedGOP
 Description:  The keyframe interval (GOP) of the stream to publish.
 Value:  Value range: [1,5]. Default: 3 (seconds).
videoEncodedHeight
 Description:  The resolution (height) of the stream to publish.
 Value:  Recommended value: 640. If you mix only audio streams, to avoid displaying a black video in the transcoded stream, set both  width  and  height  to  0 .
videoEncodedKbps
 Description:  The video bitrate (Kbps) of the stream to publish.
 Value:  If you set this parameter to  0 , TRTC will work out a bitrate based on  videoWidth  and  videoHeight . For details, refer to the recommended bitrates for the constants of the resolution enum type (see comment).
videoEncodedWidth
 Description:  The resolution (width) of the stream to publish.
 Value:  Recommended value: 368. If you mix only audio streams, to avoid displaying a black video in the transcoded stream, set both  width  and  height  to  0 .
videoSeiParams
 Description:  SEI parameters. Default: null
 Note:  the parameter is passed in the form of a JSON string. Here is an example to use it:
{
  "payLoadContent":"xxx",
  "payloadType":5,
  "payloadUuid":"1234567890abcdef1234567890abcdef",
  "interval":1000,
  "followIdr":false
}
The currently supported fields and their meanings are as follows:
 payloadContent: Required. The payload content of the passthrough SEI, which cannot be empty.
 payloadType: Required. The type of the SEI message, with a value range of 5 or an integer within the range of [100, 254] (excluding 244, which is an internally defined timestamp SEI).
 payloadUuid: Required when payloadType is 5, and ignored in other cases. The value must be a 32-digit hexadecimal number.
 interval: Optional, default is 1000. The sending interval of the SEI, in milliseconds.
 followIdr: Optional, default is false. When this value is true, the SEI will be ensured to be carried when sending a key frame, otherwise it is not guaranteed.
TRTCStreamMixingConfig
TRTCStreamMixingConfig
The transcoding parameters.This enum type is used by the publishing API ﻿startPublishMediaStream﻿.
You can use this parameter to specify the video layout and input audio information for On-Cloud MixTranscoding.
EnumType
DESC
audioMixUserList
 Description:  The information of each audio stream to mix.
 Value:  This parameter is an array. Each  TRTCUser  element in the array indicates the information of an audio stream.
Note
If you do not specify this array, the TRTC backend will automatically mix all streams of the anchors who are sending audio in the room according to the audio encode param ﻿TRTCStreamEncoderParam﻿ you specify (currently only supports up to 16 audio and video inputs).
backgroundColor
 Description:  The background color of the mixed stream.
 Value:  The value must be a hex number. For example, "0x61B9F1" represents the RGB color value (97,158,241). Default value: 0x000000 (black).
backgroundImage
 Description:  The URL of the background image of the mixed stream. The image specified by the URL will be mixed during On-Cloud MixTranscoding.
 Value:  This parameter is left empty by default, which means no background image will be used.
Note
 The URL can be 512 bytes long at most, and the image must not exceed 2 MB.
 The image can be in PNG, JPG, JPEG, or BMP format. We recommend you use a semitransparent image in PNG format.
videoLayoutList
 Description:  The position, size, layer, and stream type of each video in On-Cloud MixTranscoding.
 Value:  This parameter is an array. Each  TRTCVideoLayout  element in the array indicates the information of a video in On-Cloud MixTranscoding.
watermarkList
 Description:  The position, size, and layer of each watermark image in On-Cloud MixTranscoding.
 Value:  This parameter is an array. Each  TRTCWatermark  element in the array indicates the information of a watermark.
TRTCPayloadPrivateEncryptionConfig
TRTCPayloadPrivateEncryptionConfig
Media Stream Private Encryption Configuration.This configuration is used to set the algorithm and key for media stream private encryption.
EnumType
DESC
encryptionAlgorithm
 Description:  Encryption algorithm, the default is ﻿TRTCEncryptionAlgorithmAes128Gcm﻿.
encryptionKey
 Description:  encryption key, string type.
 Value:  If the encryption algorithm is ﻿TRTCEncryptionAlgorithmAes128Gcm﻿, the key length must be 16 bytes;
if the encryption algorithm is ﻿TRTCEncryptionAlgorithmAes256Gcm﻿, the key length must be 32 bytes.
encryptionSalt
 Description:  Salt, initialization vector for encryption.
 Value:  It is necessary to ensure that the array filled in this parameter is not empty, not all 0 and the data length is 32 bytes.
TRTCAudioVolumeEvaluateParams
TRTCAudioVolumeEvaluateParams
Volume evaluation and other related parameter settings.This setting is used to enable vocal detection and sound spectrum calculation.
EnumType
DESC
enablePitchCalculation
 Description:  Whether to enable local vocal frequency calculation.
enableSpectrumCalculation
 Description:  Whether to enable sound spectrum calculation.
enableVadDetection
 Description:  Whether to enable local voice detection.
Note
Call before startLocalAudio.
interval
 Description:  Set the trigger interval of the ﻿onUserVoiceVolume﻿ callback, the unit is milliseconds, the minimum interval is 100ms, if it is less than or equal to 0, the callback will be closed.
 Value:  Recommended value: 300, in milliseconds.
Note
When the interval is greater than 0, the volume prompt will be enabled by default, no additional setting is required.

Help and Support

Was this page helpful?

You can also Contact sales or Submit a Ticket for help.

Help us improve! Rate your documentation experience in 5 mins.

Feedback

FuncList	DESC
TRTCParams	Room entry parameters.
TRTCVideoEncParam	Video encoding parameters.
TRTCNetworkQosParam	Network QoS control parameter set.
TRTCRenderParams	Rendering parameters of video image.
TRTCQualityInfo	Network quality.
TRTCVolumeInfo	Volume.
TRTCSpeedTestParams	Network speed testing parameters.
TRTCSpeedTestResult	Network speed test result.
TRTCVideoFrame	Video frame information.
TRTCAudioFrame	Audio frame data.
TRTCMixUser	Description information of each video image in On-Cloud MixTranscoding.
TRTCTranscodingConfig	Layout and transcoding parameters of On-Cloud MixTranscoding.
TRTCPublishCDNParam	Push parameters required to be set when publishing audio/video streams to non-Tencent Cloud CDN.
TRTCAudioRecordingParams	Local audio file recording parameters.
TRTCLocalRecordingParams	Local media file recording parameters.
TRTCAudioEffectParam	Sound effect parameter (disused).
TRTCSwitchRoomConfig	Room switch parameter.
TRTCAudioFrameDelegateFormat	Format parameter of custom audio callback.
TRTCUser	The users whose streams to publish.
TRTCPublishCdnUrl	The destination URL when you publish to Tencent Cloud or a third-party CDN.
TRTCPublishTarget	The publishing destination.
TRTCVideoLayout	The video layout of the transcoded stream.
TRTCWatermark	The watermark layout.
TRTCStreamEncoderParam	The encoding parameters.
TRTCStreamMixingConfig	The transcoding parameters.
TRTCPayloadPrivateEncryptionConfig	Media Stream Private Encryption Configuration.
TRTCAudioVolumeEvaluateParams	Volume evaluation and other related parameter settings.

EnumType	DESC
TRTCVideoResolution	Video resolution.
TRTCVideoResolutionMode	Video aspect ratio mode.
TRTCVideoStreamType	Video stream type.
TRTCVideoFillMode	Video image fill mode.
TRTCVideoRotation	Video image rotation direction.
TRTCBeautyStyle	Beauty (skin smoothing) filter algorithm.
TRTCVideoPixelFormat	Video pixel format.
TRTCVideoBufferType	Video data transfer method.
TRTCVideoMirrorType	Video mirror type.
TRTCSnapshotSourceType	Data source of local video screenshot.
TRTCAppScene	Use cases.
TRTCRoleType	Role.
TRTCQosControlMode	QoS control mode (disused).
TRTCVideoQosPreference	Image quality preference.
TRTCQuality	Network quality.
TRTCAVStatusType	Audio/Video playback status.
TRTCAVStatusChangeReason	Reasons for playback status changes.
TRTCAudioSampleRate	Audio sample rate.
TRTCAudioQuality	Sound quality.
TRTCAudioRoute	Audio route (i.e., audio playback mode).
TRTCReverbType	Audio reverb mode.
TRTCVoiceChangerType	Voice changing type.
TRTCSystemVolumeType	System volume type (only for mobile devices).
TRTCAudioFrameOperationMode	Audio callback data operation mode.
TRTCLogLevel	Log level.
TRTCGSensorMode	G-sensor switch (for mobile devices only).
TRTCScreenCaptureSourceType	Screen sharing target type (for desktops only).
TRTCTranscodingConfigMode	Layout mode of On-Cloud MixTranscoding.
TRTCRecordType	Media recording type.
TRTCMixInputType	Stream mix input type.
TRTCAudioRecordingContent	Audio recording content type.
TRTCPublishMode	The publishing mode.
TRTCEncryptionAlgorithm	Encryption Algorithm.
TRTCSpeedTestScene	Speed Test Scene.
TRTCGravitySensorAdaptiveMode	Set the adaptation mode of gravity sensing (only applicable to mobile terminals).

Enum	Value	DESC
TRTCVideoResolution_120_120	1	Aspect ratio: 1:1; resolution: 120x120; recommended bitrate (VideoCall): 80 Kbps; recommended bitrate (LIVE): 120 Kbps.
TRTCVideoResolution_160_160	3	Aspect ratio: 1:1; resolution: 160x160; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
TRTCVideoResolution_270_270	5	Aspect ratio: 1:1; resolution: 270x270; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_480_480	7	Aspect ratio: 1:1; resolution: 480x480; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 500 Kbps.
TRTCVideoResolution_160_120	50	Aspect ratio: 4:3; resolution: 160x120; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
TRTCVideoResolution_240_180	52	Aspect ratio: 4:3; resolution: 240x180; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
TRTCVideoResolution_280_210	54	Aspect ratio: 4:3; resolution: 280x210; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_320_240	56	Aspect ratio: 4:3; resolution: 320x240; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 375 Kbps.
TRTCVideoResolution_400_300	58	Aspect ratio: 4:3; resolution: 400x300; recommended bitrate (VideoCall): 300 Kbps; recommended bitrate (LIVE): 450 Kbps.
TRTCVideoResolution_480_360	60	Aspect ratio: 4:3; resolution: 480x360; recommended bitrate (VideoCall): 400 Kbps; recommended bitrate (LIVE): 600 Kbps.
TRTCVideoResolution_640_480	62	Aspect ratio: 4:3; resolution: 640x480; recommended bitrate (VideoCall): 600 Kbps; recommended bitrate (LIVE): 900 Kbps.
TRTCVideoResolution_960_720	64	Aspect ratio: 4:3; resolution: 960x720; recommended bitrate (VideoCall): 1000kbps; recommended bitrate (LIVE): 1500kbps。
TRTCVideoResolution_160_90	100	Aspect ratio: 16:9; resolution: 160x90; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
TRTCVideoResolution_256_144	102	Aspect ratio: 16:9; resolution: 256x144; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_320_180	104	Aspect ratio: 16:9; resolution: 320x180; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 400 Kbps.
TRTCVideoResolution_480_270	106	Aspect ratio: 16:9; resolution: 480x270; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 550 Kbps.
TRTCVideoResolution_640_360	108	Aspect ratio: 16:9; resolution: 640x360; recommended bitrate (VideoCall): 500 Kbps; recommended bitrate (LIVE): 900 Kbps.
TRTCVideoResolution_960_540	110	Aspect ratio: 16:9; resolution: 960x540; recommended bitrate (VideoCall): 850 Kbps; recommended bitrate (LIVE): 1300 Kbps.
TRTCVideoResolution_1280_720	112	Aspect ratio: 16:9; resolution: 1280x720; recommended bitrate (VideoCall): 1200 Kbps; recommended bitrate (LIVE): 1800 Kbps.
TRTCVideoResolution_1920_1080	114	Aspect ratio: 16:9; resolution: 1920x1080; recommended bitrate (VideoCall): 2000 Kbps; recommended bitrate (LIVE): 3000 Kbps.

EnumType	DESC
bussInfo	Field description: business data, which is optional. This field is needed only by some advanced features. Recommended value: do not set this field on your own.
privateMapKey	Field description: permission credential used for permission control, which is optional. If you want only users with the specified `userId` values to enter a room, you need to use `privateMapKey` to restrict the permission. Recommended value: we recommend you use this parameter only if you have high security requirements. For more information, please see Enabling Advanced Permission Control.
role	Field description: role in the live streaming scenario, which is applicable only to the live streaming scenario (TRTCAppSceneLIVE or TRTCAppSceneVoiceChatRoom) but doesn't take effect in the call scenario. Recommended value: default value: anchor (TRTCRoleAnchor).
roomId	Field description: numeric room ID. Users (userId) in the same room can see one another and make audio/video calls. Recommended value: value range: 1–4294967294. @note `roomId` and `strRoomId` are mutually exclusive. If you decide to use `strRoomId` , then `roomId` should be entered as 0. If both are entered, `roomId` will be used. Note do not mix `roomId` and `strRoomId` , because they are not interchangeable. For example, the number `123` and the string `123` are two completely different rooms in TRTC.
sdkAppId	Field description: application ID, which is required. Tencent Cloud generates bills based on `sdkAppId` . Recommended value: the ID can be obtained on the account information page in the TRTC console after the corresponding application is created.
strRoomId	Field description: string-type room ID. Users (userId) in the same room can see one another and make audio/video calls. @note `roomId` and `strRoomId` are mutually exclusive. If you decide to use `strRoomId` , then `roomId` should be entered as 0. If both are entered, `roomId` will be used. Note do not mix `roomId` and `strRoomId` , because they are not interchangeable. For example, the number `123` and the string `123` are two completely different rooms in TRTC. Recommended value: the length limit is 64 bytes. The following 89 characters are supported: Uppercase and lowercase letters (a–z and A–Z) Digits (0–9) Space, "!", "#", "$", "%", "&", "(", ")", "+", "-", ":", ";", "<", "=", ".", ">", "?", "@", "[", "]", "^", "_", "{", "}", "\|", "~", and ",".
streamId	Field description: specified `streamId` in Tencent Cloud CSS, which is optional. After setting this field, you can play back the user's audio/video stream on Tencent Cloud CSS CDN through a standard pull scheme (FLV or HLS). Recommended value: this parameter can contain up to 64 bytes and can be left empty. We recommend you use `sdkappid_roomid_userid_main` as the `streamid` , which is easier to identify and will not cause conflicts in your multiple applications. Note to use Tencent Cloud CSS CDN, you need to enable the auto-relayed live streaming feature on the "Function Configuration" page in the console first. For more information, please see Relay to CDN.
userDefineRecordId	Field description: on-cloud recording field, which is optional and used to specify whether to record the user's audio/video stream in the cloud. For more information, please see On-Cloud Recording. Recommended value: it can contain up to 64 bytes. Letters (a–z and A–Z), digits (0–9), underscores, and hyphens are allowed. Scheme 1. Manual recording 1. Enable on-cloud recording in "Application Management" > "On-cloud Recording Configuration" in the console. 2. Set "Recording Mode" to "Manual Recording". 3. After manual recording is set, in a TRTC room, only users with the `userDefineRecordId` parameter set will have video recording files in the cloud, while users without this parameter set will not. 4. The recording file will be named in the format of "userDefineRecordId_start time_end time" in the cloud. Scheme 2. Auto-recording 1. You need to enable on-cloud recording in "Application Management" > "On-cloud Recording Configuration" in the console. 2. Set "Recording Mode" to "Auto-recording". 3. After auto-recording is set, any user who upstreams audio/video in a TRTC room will have a video recording file in the cloud. 4. The file will be named in the format of "userDefineRecordId_start time_end time". If `userDefineRecordId` is not specified, the file will be named in the format of "streamId_start time_end time".
userId	Field description: user ID, which is required. It is the `userId` of the local user in UTF-8 encoding and acts as the username. Recommended value: if the ID of a user in your account system is "mike", `userId` can be set to "mike".
userSig	Field description: user signature, which is required. It is the authentication signature corresponding to the current `userId` and acts as the login password for Tencent Cloud services. Recommended value: for the calculation method, please see UserSig.

EnumType	DESC
enableAdjustRes	Field description: whether to allow dynamic resolution adjustment. Once enabled, this field will affect on-cloud recording. Recommended value: this feature is suitable for scenarios that don't require on-cloud recording. After it is enabled, the SDK will intelligently select a suitable resolution according to the current network conditions to avoid the inefficient encoding mode of "large resolution + small bitrate". Note default value: NO. If you need on-cloud recording, please do not enable this feature, because if the video resolution changes, the MP4 file recorded in the cloud cannot be played back normally by common players.
minVideoBitrate	Field description: minimum video bitrate. The SDK will reduce the bitrate to as low as the value specified by `minVideoBitrate` to ensure the smoothness only if the network conditions are poor. Note: default value: 0, indicating that a reasonable value of the lowest bitrate will be automatically calculated by the SDK according to the resolution you specify. Recommended value: you can set the `videoBitrate` and `minVideoBitrate` parameters at the same time to restrict the SDK's adjustment range of the video bitrate: If you want to "ensure clarity while allowing lag in weak network environments", you can set `minVideoBitrate` to 60% of `videoBitrate` . If you want to "ensure smoothness while allowing blur in weak network environments", you can set `minVideoBitrate` to a low value, for example, 100 Kbps. If you set `videoBitrate` and `minVideoBitrate` to the same value, it is equivalent to disabling the adaptive adjustment capability of the SDK for the video bitrate.
resMode	Field description: resolution mode (landscape/portrait) Recommended value: for mobile platforms (iOS and Android), `Portrait` is recommended; for desktop platforms (Windows and macOS), `Landscape` is recommended. Note to use a portrait resolution, please specify `resMode` as `Portrait` ; for example, when used together with `Portrait` , 640x360 represents 360x640.
videoBitrate	Field description: target video bitrate. The SDK encodes streams at the target video bitrate and will actively reduce the bitrate only in weak network environments. Recommended value: please see the optimal bitrate for each specification in `TRTCVideoResolution` . You can also slightly increase the optimal bitrate. For example, `TRTCVideoResolution_1280_720` corresponds to the target bitrate of 1,200 Kbps. You can also set the bitrate to 1,500 Kbps for higher definition. Note you can set the `videoBitrate` and `minVideoBitrate` parameters at the same time to restrict the SDK's adjustment range of the video bitrate: If you want to "ensure clarity while allowing lag in weak network environments", you can set `minVideoBitrate` to 60% of `videoBitrate` . If you want to "ensure smoothness while allowing blur in weak network environments", you can set `minVideoBitrate` to a low value, for example, 100 Kbps. If you set `videoBitrate` and `minVideoBitrate` to the same value, it is equivalent to disabling the adaptive adjustment capability of the SDK for the video bitrate.
videoFps	Field description: video capturing frame rate Recommended value: 15 or 20 fps. If the frame rate is lower than 5 fps, there will be obvious lagging; if lower than 10 fps but higher than 5 fps, there will be slight lagging; if higher than 20 fps, the bandwidth will be wasted (the frame rate of movies is generally 24 fps). Note the front cameras on certain Android phones do not support a capturing frame rate higher than 15 fps. For some Android phones that focus on beautification features, the capturing frame rate of the front cameras may be lower than 10 fps.
videoResolution	Field description: video resolution Recommended value For mobile video call, we recommend you select a resolution of 360x640 or below and select `Portrait` (portrait resolution) for `resMode` . For mobile live streaming, we recommend you select a resolution of 540x960 and select `Portrait` (portrait resolution) for `resMode` . For desktop platforms (Windows and macOS), we recommend you select a resolution of 640x360 or above and select `Landscape` (landscape resolution) for `resMode` . Note to use a portrait resolution, please specify `resMode` as `Portrait` ; for example, when used together with `Portrait` , 640x360 represents 360x640.

EnumType	DESC
controlMode	Field description: QoS control mode (disused) Recommended value: on-cloud control Note please set the on-cloud control mode (TRTCQosControlModeServer).
preference	Field description: whether to ensure smoothness or clarity Recommended value: ensuring clarity Note this parameter mainly affects the audio/video performance of TRTC in weak network environments: Ensuring smoothness: in this mode, when the current network is unable to transfer a clear and smooth video image, the smoothness of the image will be given priority, but there will be blurs. See TRTCVideoQosPreferenceSmooth Ensuring clarity (default value): in this mode, when the current network is unable to transfer a clear and smooth video image, the clarity of the image will be given priority, but there will be lags. See TRTCVideoQosPreferenceClear

EnumType	DESC
fillMode	Field description: image fill mode Recommended value: fill (the image may be stretched or cropped) or fit (there may be black bars in unmatched areas). Default value: TRTCVideoFillMode_Fill
mirrorType	Field description: image mirror mode Recommended value: default value: TRTCVideoMirrorTypeAuto
rotation	Field description: clockwise image rotation angle Recommended value: rotation angles of 90, 180, and 270 degrees are supported. Default value: TRTCVideoRotation_0

EnumType	DESC
pitch	The local user's vocal frequency (unit: Hz), the value range is [0 - 4000]. For remote users, this value is always 0.
spectrumData	Audio spectrum data, which divides the sound frequency into 256 frequency domains, spectrumData records the energy value of each frequency domain, The value range of each energy value is [-300, 0] in dBFS. Note The local spectrum is calculated using the audio data before encoding, which will be affected by the capture volume, BGM, etc.; the remote spectrum is calculated using the received audio data, and operations such as adjusting the remote playback volume locally will not affect it.
userId	`userId` of the speaker. An empty value indicates the local user.
vad	Vad result of the local user. 0: not speech 1: speech.
volume	Volume of the speaker. Value range: 0–100.

EnumType	DESC
expectedDownBandwidth	Expected downstream bandwidth (kbps, value range: 10 to 5000, no downlink bandwidth test when it is 0). Note When the parameter `scene` is set to TRTCSpeedTestScene_OnlineChorusTesting, in order to obtain more accurate information such as `rtt / jitter` , the value range is limited to [10, 1000].
expectedUpBandwidth	Expected upstream bandwidth (kbps, value range: 10 to 5000, no uplink bandwidth test when it is 0). Note When the parameter `scene` is set to TRTCSpeedTestScene_OnlineChorusTesting, in order to obtain more accurate information such as `rtt / jitter` , the value range is limited to [10, 1000].
scene	Speed test scene.
sdkAppId	Application identification, please refer to the relevant instructions in TRTCParams.
userId	User identification, please refer to the relevant instructions in TRTCParams.
userSig	User signature, please refer to the relevant instructions in TRTCParams.

EnumType	DESC
availableDownBandwidth	Downstream bandwidth (in kbps, -1: invalid value).
availableUpBandwidth	Upstream bandwidth (in kbps, -1: invalid value).
downJitter	Downlink data packet jitter (ms) refers to the stability of data communication in the user's current network environment. The smaller the value, the better. The normal value range is [0, 100]. -1 means that the speed test failed to obtain an effective value. Generally, the Jitter of the WiFi network will be slightly larger than that of the 4G/5G environment.
downLostRate	Downstream packet loss rate between 0 and 1.0. For example, 0.2 indicates that 2 data packets may be lost in every 10 packets received from the server.
errMsg	Error message for network speed test.
ip	Server IP address.
quality	Network quality, which is tested and calculated based on the internal evaluation algorithm. For more information, please see TRTCQuality
rtt	Delay in milliseconds, which is the round-trip time between the current device and TRTC server. The smaller the value, the better. The normal value range is 10–100 ms.
success	Whether the network speed test is successful.
upJitter	Uplink data packet jitter (ms) refers to the stability of data communication in the user's current network environment. The smaller the value, the better. The normal value range is [0, 100]. -1 means that the speed test failed to obtain an effective value. Generally, the Jitter of the WiFi network will be slightly larger than that of the 4G/5G environment.
upLostRate	Upstream packet loss rate between 0 and 1.0. For example, 0.3 indicates that 3 data packets may be lost in every 10 packets sent to the server.

EnumType	DESC
bufferType	Field description: video data structure type
data	Field description: video data when `bufferType` is TRTCVideoBufferType_NSData, which carries the memory data blocks in `NSData` type.
height	Field description: video height Recommended value: please enter the height of the video data passed in.
pixelBuffer	Field description: video data when `bufferType` is TRTCVideoBufferType_PixelBuffer, which carries the `PixelBuffer` unique to iOS.
pixelFormat	Field description: video pixel format
rotation	Field description: clockwise rotation angle of video pixels
textureId	Field description: video texture ID, i.e., video data when `bufferType` is TRTCVideoBufferType_Texture, which carries the texture data used for OpenGL rendering.
timestamp	Field description: video frame timestamp in milliseconds Recommended value: this parameter can be set to 0 for custom video capturing. In this case, the SDK will automatically set the `timestamp` field. However, please "evenly" set the calling interval of sendCustomVideoData.
width	Field description: video width Recommended value: please enter the width of the video data passed in.

EnumType	DESC
channels	Field description: number of sound channels
data	Field description: audio data
extraData	Field description: extra data in audio frame, message sent by remote users through onLocalProcessedAudioFrame that add to audio frame will be callback through this field.
sampleRate	Field description: sample rate
timestamp	Field description: timestamp in ms

EnumType	DESC
image	Field description: specify the placeholder or watermark image. The placeholder image will be displayed when there is no upstream video.A watermark image is a semi-transparent image posted in the mixed image, and this image will always be overlaid on the mixed image. When the `inputType` field is set to TRTCMixInputTypePureAudio, the image is a placeholder image, and you need to specify `userId` . When the `inputType` field is set to TRTCMixInputTypeWatermark, the image is a watermark image, and you don't need to specify `userId` . Recommended value: default value: null, indicating not to set the placeholder or watermark image. Note TRTC's backend service will mix the image specified by the URL address into the final stream.URL link length is limited to 512 bytes. The image size is limited to 10MB.Support png, jpg, jpeg, bmp format. Take effects iff the `inputType` field is set to TRTCMixInputTypePureAudio or TRTCMixInputTypeWatermark.
inputType	Field description: specify the mixed content of this stream (audio only, video only, audio and video, or watermark). Recommended value: default value: TRTCMixInputTypeUndefined. Note When specifying `inputType` as TRTCMixInputTypeUndefined and specifying `pureAudio` to YES, it is equivalent to setting `inputType` to `TRTCMixInputTypePureAudio` . When specifying `inputType` as TRTCMixInputTypeUndefined and specifying `pureAudio` to NO, it is equivalent to setting `inputType` to `TRTCMixInputTypeAudioVideo` . When specifying `inputType` as TRTCMixInputTypeWatermark, you don't need to specify the `userId` field, but you need to specify the `image` field.
pureAudio	Field description: specify whether this stream mixes audio only Recommended value: default value: NO Note this field has been disused. We recommend you use the new field `inputType` introduced in v8.5.
rect	Field description: specify the coordinate area of this video image in px
renderMode	Field description: specify the display mode of this stream. Recommended value: default value: 0. 0 is cropping, 1 is zooming, 2 is zooming and displaying black background. Note image doesn't support setting `renderMode` temporarily, the default display mode is forced stretch.
roomID	Field description: ID of the room where this audio/video stream is located (an empty value indicates the local room ID)
soundLevel	Field description: specify the target volumn level of On-Cloud MixTranscoding. (value range: 0-100) Recommended value: default value: 100.
streamType	Field description: specify whether this video image is the primary stream image (TRTCVideoStreamTypeBig) or substream image (TRTCVideoStreamTypeSub).
userId	Field description: user ID
zOrder	Field description: specify the level of this video image (value range: [1, 15]; the value must be unique)

EnumType	DESC
appId	Field description: `appId` of Tencent Cloud CSS Recommended value: please click `Application Management` > `Application Information` in the TRTC console and get the `appId` in `Relayed Live Streaming Info` . Note applications created on or after January 9, 2020 do not need to fill in this field.
audioBitrate	Field description: specify the target audio bitrate of On-Cloud MixTranscoding Recommended value: default value: 64 Kbps. Value range: [32,192].
audioChannels	Field description: specify the number of sound channels of On-Cloud MixTranscoding Recommended value: default value: 1, which means mono channel. Valid values: 1: mono channel; 2: dual channel.
audioCodec	Field description: specify the audio encoding type of On-Cloud MixTranscoding Recommended value: default value: 0, which means LC-AAC. Valid values: 0: LC-AAC; 1: HE-AAC; 2: HE-AACv2. Note HE-AAC and HE-AACv2 only support [48000, 44100, 32000, 24000, 16000] sample rate. HE-AACv2 only support dual channel. HE-AAC and HE-AACv2 take effects iff the output streamId is specified.
audioSampleRate	Field description: specify the target audio sample rate of On-Cloud MixTranscoding Recommended value: default value: 48000 Hz. Valid values: 12000 Hz, 16000 Hz, 22050 Hz, 24000 Hz, 32000 Hz, 44100 Hz, 48000 Hz.
backgroundColor	Field description: specify the background color of the mixed video image. Recommended value: default value: 0x000000, which means black and is in the format of hex number; for example: "0x61B9F1" represents the RGB color (97,158,241).
backgroundImage	Field description: specify the background image of the mixed video image. Recommended value: default value: null, indicating not to set the background image. Note** TRTC's backend service will mix the image specified by the URL address into the final stream.URL link length is limited to 512 bytes. The image size is limited to 10MB.Support png, jpg, jpeg, bmp format.
bizId	Field description: `bizId` of Tencent Cloud CSS Recommended value: please click `Application Management` > `Application Information` in the TRTC console and get the `bizId` in `Relayed Live Streaming Info` . Note applications created on or after January 9, 2020 do not need to fill in this field.
mixUsers	Field description: specify the position, size, layer, and stream type of each video image in On-Cloud MixTranscoding Recommended value: this field is an array in `TRTCMixUser` type, where each element represents the information of a video image.
mode	Field description: layout mode Recommended value: please choose a value according to your business needs. The preset mode has better applicability.
streamId	Field description: ID of the live stream output to CDN Recommended value: default value: null, that is, the audio/video streams in the room will be mixed into the audio/video stream of the caller of this API. If you don't set this parameter, the SDK will execute the default logic, that is, it will mix the multiple audio/video streams in the room into the audio/video stream of the caller of this API, i.e., A + B => A. If you set this parameter, the SDK will mix the audio/video streams in the room into the live stream you specify, i.e., A + B => C (C is the `streamId` you specify).
videoBitrate	Field description: specify the target video bitrate (Kbps) of On-Cloud MixTranscoding Recommended value: if you enter 0, TRTC will estimate a reasonable bitrate value based on `videoWidth` and `videoHeight` . You can also refer to the recommended bitrate value in the video resolution enumeration definition (in the comment section).
videoFramerate	Field description: specify the target video frame rate (fps) of On-Cloud MixTranscoding Recommended value: default value: 15 fps. Value range: (0,30].
videoGOP	Field description: specify the target video keyframe interval (GOP) of On-Cloud MixTranscoding Recommended value: default value: 2 (in seconds). Value range: [1,8].
videoHeight	Field description: specify the target resolution (height) of On-Cloud MixTranscoding Recommended value: 640 px. If you only mix audio streams, please set both `width` and `height` to 0; otherwise, there will be a black background in the live stream after mixtranscoding.
videoSeiParams	Field description: SEI parameters. default value: null Note the parameter is passed in the form of a JSON string. Here is an example to use it: The currently supported fields and their meanings are as follows: payloadContent: Required. The payload content of the passthrough SEI, which cannot be empty. payloadType: Required. The type of the SEI message, with a value range of 5 or an integer within the range of [100, 254] (excluding 244, which is an internally defined timestamp SEI). payloadUuid: Required when payloadType is 5, and ignored in other cases. The value must be a 32-digit hexadecimal number. interval: Optional, default is 1000. The sending interval of the SEI, in milliseconds. followIdr: Optional, default is false. When this value is true, the SEI will be ensured to be carried when sending a key frame, otherwise it is not guaranteed.
videoWidth	Field description: specify the target resolution (width) of On-Cloud MixTranscoding Recommended value: 360 px. If you only mix audio streams, please set both `width` and `height` to 0; otherwise, there will be a black background in the live stream after mixtranscoding.

EnumType	DESC
filePath	Field description: storage path of the audio recording file, which is required. Note this path must be accurate to the file name and extension. The extension determines the format of the audio recording file. Currently, supported formats include PCM, WAV, and AAC. For example, if you specify the path as `mypath/record/audio.aac` , it means that you want the SDK to generate an audio recording file in AAC format.Please specify a valid path with read/write permissions; otherwise, the audio recording file cannot be generated.
maxDurationPerFile	Field description: `maxDurationPerFile` is the max duration of each recorded file segments, in milliseconds, with a minimum value of 10000. The default value is 0, indicating no segmentation.
recordingContent	Field description: Audio recording content type. Note: Record all local and remote audio by default.

EnumType	DESC
effectId	Field description: sound effect ID Note: the SDK supports playing multiple sound effects. IDs are used to distinguish different sound effects and control their start, end, volume, etc.
loopCount	Field description: number of times the sound effect is looped Valid values: 0 or any positive integer. 0 (default) indicates that the sound effect is played once, 1 twice, and so on.
path	Field description: sound effect file path. Supported file formats include AAC, MP3, and M4A.
publish	Field description: whether the sound effect is upstreamed Recommended value: YES: when the sound effect is played back locally, it will be upstreamed to the cloud and can be heard by remote users. NO: the sound effect will not be upstreamed to the cloud and can only be heard locally. Default value: NO
volume	Field description: sound effect volume Recommended value: value range: 0–100. Default value: 100

EnumType	DESC
intRoomId	Description: Numeric room ID. The room ID must be of the same type as that in TRTCParams. Value: Value range: 1-4294967294 Note: You cannot use both `intRoomId` and `strRoomId` . If you specify `strRoomId` , you need to set `intRoomId` to `0` . If you set both, only `intRoomId` will be used.
strRoomId	Description: String-type room ID. The room ID must be of the same type as that in TRTCParams. Note: You cannot use both `intRoomId` and `strRoomId` . If you specify `roomId` , you need to leave `strRoomId` empty. If you set both, only `intRoomId` will be used. Value: 64 bytes or shorter; supports the following character set (89 characters): Uppercase and lowercase letters (a-z and A-Z) Numbers (0-9) Space, "!", "#", "$", "%", "&", "(", ")", "+", "-", ":", ";", "<", "=", ".", ">", "?", "@", "[", "]", "^", "_", "{", "}", "\|", "~", ","
userId	/Description: UTF-8-encoded user ID (required) Value: For example, if the ID of a user in your account system is "mike", set it to `mike` .

EnumType	DESC
isInternalLine	Description: Whether to publish to Tencent Cloud Value: The default value is `true` . Note: If the destination URL you set is provided by Tencent Cloud, set this parameter to `true` , and you will not be charged relaying fees.
rtmpUrl	Description: The destination URL (RTMP) when you publish to Tencent Cloud or a third-party CDN. Value: The URLs of different CDN providers may vary greatly in format. Please enter a valid URL as required by your service provider. TRTC's backend server will push audio/video streams in the standard format to the URL you provide. Note: The URL must be in RTMP format. It must also meet the requirements of your service provider, or your service provider may reject push requests from the TRTC backend.

EnumType	DESC
cdnUrlList	`Description:` The destination URLs (RTMP) when you publish to Tencent Cloud or third-party CDNs. `Note:` You don’t need to set this parameter if you set the publishing mode to TRTCPublishMixStreamToRoom.
mixStreamIdentity	`Description:` The information of the robot that publishes the transcoded stream to a TRTC room. `Note:` You need to set this parameter only if you set the publishing mode to TRTCPublishMixStreamToRoom`. `Note:` After you set this parameter, the stream will be pushed to the room you specify. We recommend you set it to a special user ID to distinguish the robot from the anchor who enters the room via the TRTC SDK. `Note:` Users whose streams are transcoded cannot subscribe to the transcoded stream. `Note:` If you set the subscription mode (setDefaultStreamRecvMode) to manual before room entry, you need to manage the streams to receive by yourself (normally, if you receive the transcoded stream, you need to unsubscribe from the streams that are transcoded). `Note:` If you set the subscription mode (setDefaultStreamRecvMode) to auto before room entry, users whose streams are not transcoded will receive the transcoded stream automatically and will unsubscribe from the users whose streams are transcoded. You call muteRemoteVideoStream and muteRemoteAudio to unsubscribe from the transcoded stream.
mode	`Description:` The publishing mode. `Value:` You can relay streams to a CDN, transcode streams, or publish streams to an RTC room. Select the mode that fits your needs. Note If you need to use more than one publishing mode, you can call startPublishMediaStream multiple times and set `TRTCPublishTarget` to a different value each time.You can use one mode each time you call the startPublishMediaStream) API. To modify the configuration, call updatePublishCDNStream.

EnumType	DESC
audioEncodedChannelNum	`Description:` The sound channels of the stream to publish. `Value:` Valid values: 1 (mono channel); 2 (dual-channel). Default: 1.
audioEncodedCodecType	`Description:` The audio codec of the stream to publish. `Value:` Valid values: 0 (LC-AAC); 1 (HE-AAC); 2 (HE-AACv2). Default: 0. Note The audio sample rates supported by HE-AAC and HE-AACv2 are 48000, 44100, 32000, 24000, and 16000. When HE-AACv2 is used, the output stream can only be dual-channel.
audioEncodedKbps	`Description:` The audio bitrate (Kbps) of the stream to publish. `Value:` Value range: [32,192]. Default: 50.
audioEncodedSampleRate	`Description:` The audio sample rate of the stream to publish. `Value:` Valid values: [48000, 44100, 32000, 24000, 16000, 8000]. Default: 48000 (Hz).
videoEncodedCodecType	`Description:` The video codec of the stream to publish. `Value:` Valid values: 0 (H264); 1 (H265). Default: 0.
videoEncodedFPS	`Description:` The frame rate (fps) of the stream to publish. `Value:` Value range: (0,30]. Default: 20.
videoEncodedGOP	`Description:` The keyframe interval (GOP) of the stream to publish. `Value:` Value range: [1,5]. Default: 3 (seconds).
videoEncodedHeight	`Description:` The resolution (height) of the stream to publish. `Value:` Recommended value: 640. If you mix only audio streams, to avoid displaying a black video in the transcoded stream, set both `width` and `height` to `0` .
videoEncodedKbps	`Description:` The video bitrate (Kbps) of the stream to publish. `Value:` If you set this parameter to `0` , TRTC will work out a bitrate based on `videoWidth` and `videoHeight` . For details, refer to the recommended bitrates for the constants of the resolution enum type (see comment).
videoEncodedWidth	`Description:` The resolution (width) of the stream to publish. `Value:` Recommended value: 368. If you mix only audio streams, to avoid displaying a black video in the transcoded stream, set both `width` and `height` to `0` .
videoSeiParams	`Description:` SEI parameters. Default: null `Note:` the parameter is passed in the form of a JSON string. Here is an example to use it: { "payLoadContent":"xxx", "payloadType":5, "payloadUuid":"1234567890abcdef1234567890abcdef", "interval":1000, "followIdr":false } The currently supported fields and their meanings are as follows: payloadContent: Required. The payload content of the passthrough SEI, which cannot be empty. payloadType: Required. The type of the SEI message, with a value range of 5 or an integer within the range of [100, 254] (excluding 244, which is an internally defined timestamp SEI). payloadUuid: Required when payloadType is 5, and ignored in other cases. The value must be a 32-digit hexadecimal number. interval: Optional, default is 1000. The sending interval of the SEI, in milliseconds. followIdr: Optional, default is false. When this value is true, the SEI will be ensured to be carried when sending a key frame, otherwise it is not guaranteed.

tencent cloud

Tencent Real-Time Communication

Type Definition

StructType

EnumType

TRTCVideoResolution

TRTCVideoResolutionMode

TRTCVideoStreamType

TRTCVideoFillMode

TRTCVideoRotation

TRTCBeautyStyle

TRTCVideoPixelFormat

TRTCVideoBufferType

TRTCVideoMirrorType

TRTCSnapshotSourceType

TRTCAppScene

TRTCRoleType

TRTCQosControlMode(Deprecated)

TRTCVideoQosPreference

TRTCQuality

TRTCAVStatusType

TRTCAVStatusChangeReason

TRTCAudioSampleRate

TRTCAudioQuality

TRTCAudioRoute

TRTCReverbType

TRTCVoiceChangerType

TRTCSystemVolumeType

TRTCAudioFrameOperationMode

TRTCLogLevel

TRTCGSensorMode

TRTCScreenCaptureSourceType

TRTCTranscodingConfigMode

TRTCRecordType

TRTCMixInputType

TRTCAudioRecordingContent

TRTCPublishMode

TRTCEncryptionAlgorithm

TRTCSpeedTestScene

TRTCGravitySensorAdaptiveMode

TRTCParams

TRTCVideoEncParam

TRTCNetworkQosParam

TRTCRenderParams

TRTCQualityInfo

TRTCVolumeInfo

TRTCSpeedTestParams

TRTCSpeedTestResult

TRTCVideoFrame

TRTCAudioFrame

TRTCMixUser

TRTCTranscodingConfig

TRTCPublishCDNParam

TRTCAudioRecordingParams

TRTCLocalRecordingParams

TRTCAudioEffectParam(Deprecated)

TRTCSwitchRoomConfig

TRTCAudioFrameDelegateFormat

TRTCUser

TRTCPublishCdnUrl

TRTCPublishTarget

TRTCVideoLayout

TRTCWatermark

TRTCStreamEncoderParam

TRTCStreamMixingConfig

TRTCPayloadPrivateEncryptionConfig

TRTCAudioVolumeEvaluateParams

Help and Support

EnumType	DESC
audioMixUserList	`Description:` The information of each audio stream to mix. `Value:` This parameter is an array. Each `TRTCUser` element in the array indicates the information of an audio stream. Note If you do not specify this array, the TRTC backend will automatically mix all streams of the anchors who are sending audio in the room according to the audio encode param TRTCStreamEncoderParam you specify (currently only supports up to 16 audio and video inputs).
backgroundColor	`Description:` The background color of the mixed stream. `Value:` The value must be a hex number. For example, "0x61B9F1" represents the RGB color value (97,158,241). Default value: 0x000000 (black).
backgroundImage	`Description:` The URL of the background image of the mixed stream. The image specified by the URL will be mixed during On-Cloud MixTranscoding. `Value:` This parameter is left empty by default, which means no background image will be used. Note The URL can be 512 bytes long at most, and the image must not exceed 2 MB. The image can be in PNG, JPG, JPEG, or BMP format. We recommend you use a semitransparent image in PNG format.
videoLayoutList	`Description:` The position, size, layer, and stream type of each video in On-Cloud MixTranscoding. `Value:` This parameter is an array. Each `TRTCVideoLayout` element in the array indicates the information of a video in On-Cloud MixTranscoding.
watermarkList	`Description:` The position, size, and layer of each watermark image in On-Cloud MixTranscoding. `Value:` This parameter is an array. Each `TRTCWatermark` element in the array indicates the information of a watermark.

EnumType	DESC
encryptionAlgorithm	`Description:` Encryption algorithm, the default is TRTCEncryptionAlgorithmAes128Gcm.
encryptionKey	`Description:` encryption key, string type. `Value:` If the encryption algorithm is TRTCEncryptionAlgorithmAes128Gcm, the key length must be 16 bytes; if the encryption algorithm is TRTCEncryptionAlgorithmAes256Gcm, the key length must be 32 bytes.
encryptionSalt	`Description:` Salt, initialization vector for encryption. `Value:` It is necessary to ensure that the array filled in this parameter is not empty, not all 0 and the data length is 32 bytes.

EnumType	DESC
enablePitchCalculation	`Description:` Whether to enable local vocal frequency calculation.
enableSpectrumCalculation	`Description:` Whether to enable sound spectrum calculation.
enableVadDetection	`Description:` Whether to enable local voice detection. Note Call before startLocalAudio.
interval	`Description:` Set the trigger interval of the onUserVoiceVolume callback, the unit is milliseconds, the minimum interval is 100ms, if it is less than or equal to 0, the callback will be closed. `Value:` Recommended value: 300, in milliseconds. Note When the interval is greater than 0, the volume prompt will be enabled by default, no additional setting is required.