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Module: TRTC key class definition

Description: definitions of enumerated and constant values such as resolution and quality level
Type Define

StructType

FuncList
DESC
Room entry parameters
Video encoding parameters
Network QoS control parameter set
Rendering parameters of video image
Network quality
Volume
Network speed testing parameters
Network speed test result
Video texture data
Video frame information
Audio frame data
Description information of each video image in On-Cloud MixTranscoding
Layout and transcoding parameters of On-Cloud MixTranscoding
Push parameters required to be set when publishing audio/video streams to non-Tencent Cloud CDN
Local audio file recording parameters
Local media file recording parameters
Sound effect parameter (disused)
Room switch parameter
Format parameter of custom audio callback
Screen sharing parameter (for Android only)
The users whose streams to publish
The destination URL when you publish to Tencent Cloud or a third-party CDN
The publishing destination
The video layout of the transcoded stream
The watermark layout
The encoding parameters
The transcoding parameters
Media Stream Private Encryption Configuration
Volume evaluation and other related parameter settings.

EnumType

EnumType
DESC
Video resolution
Video aspect ratio mode
Video stream type
Video image fill mode
Video image rotation direction
Beauty (skin smoothing) filter algorithm
Video pixel format
Video data transfer method
Video mirror type
Data source of local video screenshot
Use cases
Role
QoS control mode (disused)
Image quality preference
Network quality
Audio/Video playback status
Reasons for playback status changes
Audio sample rate
Sound quality
Audio route (i.e., audio playback mode)
Audio reverb mode
Voice changing type
System volume type (only for mobile devices)
Audio frame content format
Audio capability type supported by the system (only for Android devices)
Audio callback data operation mode
Log level
G-sensor switch (for mobile devices only)
Layout mode of On-Cloud MixTranscoding
Media recording type
Stream mix input type
Debugging information displayed in the rendering control
Audio recording content type
The publishing mode
Encryption Algorithm
Speed Test Scene
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
TRTC_VIDEO_RESOLUTION_120_120
1
Aspect ratio: 1:1; resolution: 120x120; recommended bitrate (VideoCall): 80 Kbps; recommended bitrate (LIVE): 120 Kbps.
TRTC_VIDEO_RESOLUTION_160_160
3
Aspect ratio: 1:1; resolution: 160x160; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
TRTC_VIDEO_RESOLUTION_270_270
5
Aspect ratio: 1:1; resolution: 270x270; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTC_VIDEO_RESOLUTION_480_480
7
Aspect ratio: 1:1; resolution: 480x480; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 500 Kbps.
TRTC_VIDEO_RESOLUTION_160_120
50
Aspect ratio: 4:3; resolution: 160x120; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
TRTC_VIDEO_RESOLUTION_240_180
52
Aspect ratio: 4:3; resolution: 240x180; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
TRTC_VIDEO_RESOLUTION_280_210
54
Aspect ratio: 4:3; resolution: 280x210; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTC_VIDEO_RESOLUTION_320_240
56
Aspect ratio: 4:3; resolution: 320x240; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 375 Kbps.
TRTC_VIDEO_RESOLUTION_400_300
58
Aspect ratio: 4:3; resolution: 400x300; recommended bitrate (VideoCall): 300 Kbps; recommended bitrate (LIVE): 450 Kbps.
TRTC_VIDEO_RESOLUTION_480_360
60
Aspect ratio: 4:3; resolution: 480x360; recommended bitrate (VideoCall): 400 Kbps; recommended bitrate (LIVE): 600 Kbps.
TRTC_VIDEO_RESOLUTION_640_480
62
Aspect ratio: 4:3; resolution: 640x480; recommended bitrate (VideoCall): 600 Kbps; recommended bitrate (LIVE): 900 Kbps.
TRTC_VIDEO_RESOLUTION_960_720
64
Aspect ratio: 4:3; resolution: 960x720; recommended bitrate (VideoCall): 1000kbps; recommended bitrate (LIVE): 1500kbps。
TRTC_VIDEO_RESOLUTION_160_90
100
Aspect ratio: 16:9; resolution: 160x90; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
TRTC_VIDEO_RESOLUTION_256_144
102
Aspect ratio: 16:9; resolution: 256x144; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTC_VIDEO_RESOLUTION_320_180
104
Aspect ratio: 16:9; resolution: 320x180; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 400 Kbps.
TRTC_VIDEO_RESOLUTION_480_270
106
Aspect ratio: 16:9; resolution: 480x270; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 550 Kbps.
TRTC_VIDEO_RESOLUTION_640_360
108
Aspect ratio: 16:9; resolution: 640x360; recommended bitrate (VideoCall): 500 Kbps; recommended bitrate (LIVE): 900 Kbps.
TRTC_VIDEO_RESOLUTION_960_540
110
Aspect ratio: 16:9; resolution: 960x540; recommended bitrate (VideoCall): 850 Kbps; recommended bitrate (LIVE): 1300 Kbps.
TRTC_VIDEO_RESOLUTION_1280_720
112
Aspect ratio: 16:9; resolution: 1280x720; recommended bitrate (VideoCall): 1200 Kbps; recommended bitrate (LIVE): 1800 Kbps.
TRTC_VIDEO_RESOLUTION_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
TRTC_VIDEO_RESOLUTION_MODE_LANDSCAPE
0
Landscape resolution, such as TRTCVideoResolution_640_360 + TRTCVideoResolutionModeLandscape = 640x360.
TRTC_VIDEO_RESOLUTION_MODE_PORTRAIT
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
TRTC_VIDEO_STREAM_TYPE_BIG
0
HD big image: it is generally used to transfer video data from the camera.
TRTC_VIDEO_STREAM_TYPE_SMALL
1
Smooth small image: it has the same content as the big image, but with lower resolution and bitrate and thus lower definition.
TRTC_VIDEO_STREAM_TYPE_SUB
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
TRTC_VIDEO_RENDER_MODE_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.
TRTC_VIDEO_RENDER_MODE_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.

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
TRTC_VIDEO_ROTATION_0
0
No rotation
TRTC_VIDEO_ROTATION_90
1
Clockwise rotation by 90 degrees
TRTC_VIDEO_ROTATION_180
2
Clockwise rotation by 180 degrees
TRTC_VIDEO_ROTATION_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
TRTC_BEAUTY_STYLE_SMOOTH
0
Smooth style, which uses a more radical algorithm for more obvious effect and is suitable for show live streaming.
TRTC_BEAUTY_STYLE_NATURE
1
Natural style, which retains more facial details for more natural effect and is suitable for most live streaming use cases.
TRTC_BEAUTY_STYLE_PITU
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
TRTC_VIDEO_PIXEL_FORMAT_UNKNOWN
0
Undefined format
TRTC_VIDEO_PIXEL_FORMAT_I420
1
YUV420P (I420) format
TRTC_VIDEO_PIXEL_FORMAT_Texture_2D
2
OpenGL 2D texture format
TRTC_VIDEO_PIXEL_FORMAT_TEXTURE_EXTERNAL_OES
3
OES external texture format (for Android)
TRTC_VIDEO_PIXEL_FORMAT_NV21
4
NV21 format
TRTC_VIDEO_PIXEL_FORMAT_RGBA
5
RGBA 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
TRTC_VIDEO_BUFFER_TYPE_UNKNOWN
0
Undefined transfer method
TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER
1
Use memory buffer to transfer video data. iOS: PixelBuffer ; Android: Direct Buffer for JNI layer; Windows: memory data block.
TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY
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.
TRTC_VIDEO_BUFFER_TYPE_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
TRTC_VIDEO_MIRROR_TYPE_AUTO
0
Auto mode: mirror the front camera's image but not the rear camera's image (for mobile devices only).
TRTC_VIDEO_MIRROR_TYPE_ENABLE
1
Mirror the images of both the front and rear cameras.
TRTC_VIDEO_MIRROR_TYPE_DISABLE
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
TRTC_SNAPSHOT_SOURCE_TYPE_STREAM
0
The SDK screencaptures the native video content from the video stream. The screenshots are not controlled by the display of the rendering control.
TRTC_SNAPSHOT_SOURCE_TYPE_VIEW
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.
TRTC_SNAPSHOT_SOURCE_TYPE_CAPTURE
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
TRTC_APP_SCENE_VIDEOCALL
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.
TRTC_APP_SCENE_LIVE
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.
TRTC_APP_SCENE_AUDIOCALL
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.
TRTC_APP_SCENE_VOICE_CHATROOM
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
VIDEO_QOS_CONTROL_CLIENT
0
Client-based control, which is for internal debugging of SDK and shall not be used by users.
VIDEO_QOS_CONTROL_SERVER
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
TRTC_VIDEO_QOS_PREFERENCE_SMOOTH
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.
TRTC_VIDEO_QOS_PREFERENCE_CLEAR
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
TRTC_QUALITY_UNKNOWN
0
Undefined
TRTC_QUALITY_Excellent
1
The current network is excellent
TRTC_QUALITY_Good
2
The current network is good
TRTC_QUALITY_Poor
3
The current network is fair
TRTC_QUALITY_Bad
4
The current network is bad
TRTC_QUALITY_Vbad
5
The current network is very bad
TRTC_QUALITY_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
TRTC_AUDIO_QUALITY_SPEECH
1
Speech mode: sample rate: 16 kHz; mono channel; bitrate: 16 Kbps. This mode has the best resistance among all modes and is suitable for audio call scenarios, such as online meeting and audio call.
TRTC_AUDIO_QUALITY_DEFAULT
2
Default mode: sample rate: 48 kHz; 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.
TRTC_AUDIO_QUALITY_MUSIC
3
Music mode: sample rate: 48 kHz; 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
TRTC_AUDIO_ROUTE_UNKNOWN
-1
Unknown：default router.
TRTC_AUDIO_ROUTE_SPEAKER
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.
TRTC_AUDIO_ROUTE_EARPIECE
1
Earpiece: the receiver at the top is used for playback. With relatively low volume, it is suitable for call scenarios that require privacy.
TRTC_AUDIO_ROUTE_WIRED_HEADSET
2
WiredHeadset：play using wired headphones.
TRTC_AUDIO_ROUTE_BLUETOOTH_HEADSET
3
BluetoothHeadset：play with bluetooth headphones.
TRTC_AUDIO_ROUTE_SOUND_CARD
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
TRTC_REVERB_TYPE_0
0
Disable reverb
TRTC_REVERB_TYPE_1
1
KTV
TRTC_REVERB_TYPE_2
2
Small room
TRTC_REVERB_TYPE_3
3
Hall
TRTC_REVERB_TYPE_4
4
Deep
TRTC_REVERB_TYPE_5
5
Resonant
TRTC_REVERB_TYPE_6
6
Metallic
TRTC_REVERB_TYPE_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
TRTC_VOICE_CHANGER_TYPE_0
0
Disable voice changing
TRTC_VOICE_CHANGER_TYPE_1
1
Child
TRTC_VOICE_CHANGER_TYPE_2
2
Girl
TRTC_VOICE_CHANGER_TYPE_3
3
Middle-Aged man
TRTC_VOICE_CHANGER_TYPE_4
4
Heavy metal
TRTC_VOICE_CHANGER_TYPE_5
5
Nasal
TRTC_VOICE_CHANGER_TYPE_6
6
Punk
TRTC_VOICE_CHANGER_TYPE_7
7
Trapped beast
TRTC_VOICE_CHANGER_TYPE_8
8
Otaku
TRTC_VOICE_CHANGER_TYPE_9
9
Electronic
TRTC_VOICE_CHANGER_TYPE_10
10
Robot
TRTC_VOICE_CHANGER_TYPE_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.

TRTCAudioFrameFormat

TRTCAudioFrameFormat

Audio frame content format

Enum
Value
DESC
TRTC_AUDIO_FRAME_FORMAT_PCM
1
Audio data in PCM format

TRTCAudioCapabilityType

TRTCAudioCapabilityType

Audio capability type supported by the system (only for Android devices)

The SDK currently provides two types of system audio capabilities to query whether they are supported: low-latency chorus capability and low-latency earmonitor capability.
Enum
Value
DESC
TRTCAudioCapabilityLowLatencyChorus
1
low-latency chorus capability
TRTCAudioCapabilityLowLatencyEarMonitor
2
low-latency earmonitor capability

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
TRTC_AUDIO_FRAME_OPERATION_MODE_READWRITE
0
Read-write mode: You can get and modify the audio data of the callback, the default mode.
TRTC_AUDIO_FRAME_OPERATION_MODE_READONLY
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
TRTC_LOG_LEVEL_VERBOSE
0
Output logs at all levels
TRTC_LOG_LEVEL_DEBUG
1
Output logs at the DEBUG, INFO, WARNING, ERROR, and FATAL levels
TRTC_LOG_LEVEL_INFO
2
Output logs at the INFO, WARNING, ERROR, and FATAL levels
TRTC_LOG_LEVEL_WARN
3
Output logs at the WARNING, ERROR, and FATAL levels
TRTC_LOG_LEVEL_ERROR
4
Output logs at the ERROR and FATAL levels
TRTC_LOG_LEVEL_FATAL
5
Output logs at the FATAL level
TRTC_LOG_LEVEL_NULL
6
Do not output any SDK logs

TRTCGSensorMode

TRTCGSensorMode

G-sensor switch (for mobile devices only)

Enum
Value
DESC
TRTC_GSENSOR_MODE_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.
TRTC_GSENSOR_MODE_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.
TRTC_GSENSOR_MODE_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.

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
TRTC_TranscodingConfigMode_Unknown
0
Undefined
TRTC_TranscodingConfigMode_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.
TRTC_TranscodingConfigMode_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.
TRTC_TranscodingConfigMode_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.
TRTC_TranscodingConfigMode_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
TRTC_RECORD_TYPE_AUDIO
0
Record audio only
TRTC_RECORD_TYPE_VIDEO
1
Record video only
TRTC_RECORD_TYPE_BOTH
2
Record both audio and video

TRTCMixInputType

TRTCMixInputType

Stream mix input type

Enum
Value
DESC
TRTC_MixInputType_Undefined
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
TRTC_MixInputType_AudioVideo
1
Mix both audio and video
TRTC_MixInputType_PureVideo
2
Mix video only
TRTC_MixInputType_PureAudio
3
Mix audio only
TRTC_MixInputType_Watermark
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.

TRTCDebugViewLevel

TRTCDebugViewLevel

Debugging information displayed in the rendering control

Enum
Value
DESC
TRTC_DEBUG_VIEW_LEVEL_GONE
0
Do not display debugging information in the rendering control
TRTC_DEBUG_VIEW_LEVEL_STATUS
1
Display audio/video statistics in the rendering control
TRTC_DEBUG_VIEW_LEVEL_ALL
2
Display audio/video statistics and key historical events in the rendering control

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
TRTC_AudioRecordingContent_All
0
Record both local and remote audio
TRTC_AudioRecordingContent_Local
1
Record local audio only
TRTC_AudioRecordingContent_Remote
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
TRTC_PublishMode_Unknown
0
Undefined
TRTC_PublishBigStream_ToCdn
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).
TRTC_PublishSubStream_ToCdn
2
Use this parameter to publish the substream (TRTCVideoStreamTypeSub) in the room to Tencent Cloud or a third-party CDN (only RTMP is supported).
TRTC_PublishMixStream_ToCdn
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).
TRTC_PublishMixStream_ToRoom
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
TRTC_EncryptionAlgorithm_Aes_128_Gcm
0
AES GCM 128。
TRTC_EncryptionAlgorithm_Aes_256_Gcm
1
AES GCM 256。

TRTCSpeedTestScene

TRTCSpeedTestScene

Speed Test Scene

This enumeration type is used for speed test scene selection.
Enum
Value
DESC
TRTC_SpeedTestScene_Delay_Testing
1
Delay testing.
TRTC_SpeedTestScene_Delay_Bandwidth_Testing
2
Delay and bandwidth testing.
TRTC_SpeedTestScene_Online_Chorus_Testing
3
Online chorus testing.

TRTCGravitySensorAdaptiveMode

TRTCGravitySensorAdaptiveMode

Set the adaptation mode of gravity sensing (only applicable to mobile terminals)

Enum
Value
DESC
TRTC_GRAVITY_SENSOR_ADAPTIVE_MODE_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.
TRTC_GRAVITY_SENSOR_ADAPTIVE_MODE_FILL_BY_CENTER_CROP
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.
TRTC_GRAVITY_SENSOR_ADAPTIVE_MODE_FIT_WITH_BLACK_BORDER
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
businessInfo
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 CDN Relayed Live Streaming.
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 and Playback.
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: false. 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.
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.
videoResolutionMode
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.

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 TRTC_VIDEO_QOS_PREFERENCE_SMOOTH
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 TRTC_VIDEO_QOS_PREFERENCE_CLEAR

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: TRTCVideoMirrorType_Auto
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 0ms - 100ms. -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 0ms - 100ms. -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.

TRTCTexture

TRTCTexture

Video texture data

EnumType
DESC
eglContext10
Field description: OpenGL context defined by (javax.microedition.khronos.egl.*)
eglContext14
Field description: OpenGL context defined by (android.opengl.*)
textureId
Field description: video texture ID

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
buffer
Field description: video data when bufferType is TRTCCloudDef#TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER, which carries the Direct Buffer used for the JNI layer.
bufferType
Field description: video data structure type
data
Field description: video data when bufferType is TRTCCloudDef#TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY, which carries the byte array used for the Java layer.
height
Field description: video height
Recommended value: please enter the height of the video data passed in.
pixelFormat
Field description: video pixel format
rotation
Field description: clockwise rotation angle of video pixels
texture
Field description: video data when bufferType is TRTCCloudDef#TRTC_VIDEO_PIXEL_FORMAT_Texture_2D, 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
channel
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
height
Field description: specify the height of this video image in px
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: false
Note
this field has been disused. We recommend you use the new field inputType introduced in v8.5.
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
width
Field description: specify the width of this video image in px
x
Field description: specify the X coordinate of this video image in px
y
Field description: specify the Y coordinate of this video image in px
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 .
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 .
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:
`json
{
"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.
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.

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
channel
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.

TRTCScreenShareParams

TRTCScreenShareParams

Screen sharing parameter (for Android only)

This parameter is used to specify the floating window and other related information during screen sharing in the screen sharing API startScreenCapture.
EnumType
DESC
enableForegroundService
@deprecated Begin from v11.8 version, in order to adapt to targetSdkVersion 34 and above, screen sharing will default to launching a built-in foreground service. This value setting will be invalid.
floatingView
Field description: you can set a floating view through this parameter.
Recommended value: starting from Android 7.0, applications running in the background with no session keep-alive configured will be force stopped by the Android system very soon.
However, when an application is sharing the screen, it will inevitably be switched to the system background. In this case, if a floating window can pop up, it can prevent the application from being force stopped by the system.
In addition, the pop-up floating window also informs the user of the ongoing screen sharing, helping remind the user to avoid the leakage of confidential information.
Note
you can also use the WindowsManager API of Android to achieve the same effect.
mediaProjection
Field description: you can set a MediaProjection to SDK through this parameter.
Recommended value: this parameter can be set as null normally.

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 (@link 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 (e 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.
height
Description: The height (in pixels) of the video.
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.
width
Description: The width (in pixels) of the video.
x
Description: The X coordinate (in pixels) of the video.
y
Description: The Y coordinate (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
height
Description: The height (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.
width
Description: The width (in pixels) of the watermark.
x
Description: The X coordinate (in pixels) of the watermark.
y
Description: The Y coordinate (in pixels) of the watermark.
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.