tencent cloud

1.1. Video Resolution

Here, only the landscape resolution is defined. If the portrait resolution (e.g., 360 * 640) needs to be used, Portrait must be selected for TRTCVideoResolutionMode.

Constant	Description
TRTC_VIDEO_RESOLUTION_120_120	[C] 80 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_160_160	[C] 100 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_270_270	[C] 200 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_480_480	[C] 350 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_160_120	[C] 100 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_240_180	[C] 150 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_280_210	[C] 200 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_320_240	[C] 250 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_400_300	[C] 300 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_480_360	[C] 400 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_640_480	[C] 600 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_960_720	[C] 1,000 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_160_90	[C] 100 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_256_144	[C] 150 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_320_180	[C] 250 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_480_270	[C] 350 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_640_360	[C] 550 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_960_540	[C] 850 Kbps bitrate is recommended.
TRTC_VIDEO_RESOLUTION_1280_720	[C] 1,200 Kbps bitrate is recommended.

1.2. Video Aspect Ratio Mode

• Landscape resolution: TRTCVideoResolution_640_360 + TRTCVideoResolutionModeLandscape = 640 * 360
• Portrait resolution: TRTCVideoResolution_640_360 + TRTCVideoResolutionModePortrait = 360 * 640

Constant	Description
TRTC_VIDEO_RESOLUTION_MODE_LANDSCAPE	Landscape resolution.
TRTC_VIDEO_RESOLUTION_MODE_PORTRAIT	Portrait resolution.

1.3. Video Stream Type

TRTC provides three different audio/video streams, including:

• Primary image: the most used channel, which is generally used to transmit video data from the camera.
• Small image: it is similar to the primary image, but with lower resolution and bitrate.
• Secondary stream image: it is generally used for screen sharing and remote video playback (for example, a teacher plays back a video to students).

• If the upstream network and performance of the anchor is good, the primary (big) and small images can be sent at the same time.
• The SDK does not support enabling only the small image, which must be enabled together with the primary image.

Constant	Description
TRTC_VIDEO_STREAM_TYPE_BIG	Primary image video stream.
TRTC_VIDEO_STREAM_TYPE_SMALL	Small image video stream.
TRTC_VIDEO_STREAM_TYPE_SUB	Secondary stream (screen sharing).

1.4. Video (or Network) Quality Constant Definition

The TRTC SDK defines six levels of image quality, among which "Excellent" stands for the best quality, and "Down" indicates that the image quality is unavailable.

Constant	Description
TRTC_QUALITY_UNKNOWN	Undefined.
TRTC_QUALITY_Excellent	Excellent.
TRTC_QUALITY_Good	Good.
TRTC_QUALITY_Poor	Poor.
TRTC_QUALITY_Bad	Bad.
TRTC_QUALITY_Vbad	Very bad.
TRTC_QUALITY_Down	Unavailable.

1.5. Video Image Fill Mode

If video image's display resolution is different from its original resolution, the fill mode needs to be set as below:

• TRTCVideoFillMode_Fill: the entire screen will be covered by the image, where parts that exceed the screen will be cropped, so the displayed image may be incomplete.
• TRTCVideoFillMode_Fit: the long side of the image will fit the screen, while the short side will be proportionally scaled with unmatched areas being filled with black color blocks, but the displayed image is definitely complete.

Constant	Description
TRTC_VIDEO_RENDER_MODE_FILL	The entire screen will be covered by the image, where parts that exceed the screen will be cropped.
TRTC_VIDEO_RENDER_MODE_FIT	The long side of the image will fit the screen, while the short side will be proportionally scaled with unmatched areas being filled with black color blocks.

1.6. Video Image Rotation Direction

The TRTC SDK provides rotation angle setting APIs for local and remote images. The following rotation angles are all clockwise.

Constant	Description
TRTC_VIDEO_ROTATION_0	No rotation.
TRTC_VIDEO_ROTATION_90	Rotates 90 degree clockwise.
TRTC_VIDEO_ROTATION_180	Rotates 180 degree clockwise.
TRTC_VIDEO_ROTATION_270	Rotates 270 degree clockwise.

1.7. Beauty (Skin Smoothing) Filter Algorithm

The TRTC SDK has multiple in-built skin smoothing algorithms. You can select the one most suitable for your product needs.

Constant	Description
TRTC_BEAUTY_STYLE_SMOOTH	Smooth, which is suitable for shows since it has more obvious effect.
TRTC_BEAUTY_STYLE_NATURE	Natural, which retains more facial details and seems more natural subjectively.

1.8. Video Pixel Format

The TRTC SDK provides custom capture and rendering features for video. In the custom capture feature, the following enumerated values can be used to describe the pixel format of the captured video. In the custom rendering feature, the video pixel format of the SDK callback can be specified.

Constant	Description
TRTC_VIDEO_PIXEL_FORMAT_UNKNOWN	Unknown.
TRTC_VIDEO_PIXEL_FORMAT_I420	YUV I420
TRTC_VIDEO_PIXEL_FORMAT_Texture_2D	OpenGL 2D texture.
TRTC_VIDEO_PIXEL_FORMAT_TEXTURE_EXTERNAL_OES	-
TRTC_VIDEO_PIXEL_FORMAT_NV21	-

1.9. Mirror Type for Local Video Image Preview

The TRTC SDK provides a mirror setting feature for the local preview image. The default mirror type is AUTO.

Constant	Description
TRTC_VIDEO_MIRROR_TYPE_AUTO	The SDK determines the mirror type: mirrors the front camera's image and does not mirror the rear camera's image.
TRTC_VIDEO_MIRROR_TYPE_ENABLE	Mirrors the images of both the front and rear cameras.
TRTC_VIDEO_MIRROR_TYPE_DISABLE	Does not mirror the images of both the front and rear cameras.

1.10. Video Data Container Format

In custom capture and rendering features, you need to use the following enumerated values to specify the type of container that is used to contain the video data.

Constant	Description
TRTC_VIDEO_BUFFER_TYPE_UNKNOWN	Unknown.
TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER	DirectBuffer, which carries buffers such as I420 and is used at the Native layer.
TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY	byte[], which carries buffers such as I420 and is used at the Java layer.
TRTC_VIDEO_BUFFER_TYPE_TEXTURE	Direct operation texture ID, which has the best performance and least image quality loss.

2.1. Application Scenarios

TRTC can be used in various application scenarios such as videoconferencing and live video broadcasting. The TRTC SDK provides different optimized configurations for different scenarios.

• VideoCall: video call, i.e., the scenario where most of the time there are two or more people on video calls, such as 1-to-1 online course, 1-to-N (N < 8) video conference, or a small class.
• LIVE: live video broadcasting (LVB), i.e., the scenario where most of the time there is only one person speaking or performing and occasionally multiple people interact with one another through video, such as co-anchoring in shows.

Constant	Description
TRTC_APP_SCENE_VIDEOCALL	Video call scenario, where optimization for internal encoders and network protocols focuses on smoothness to reduce call latency and lagging.
TRTC_APP_SCENE_LIVE	LVB scenario, where optimization for internal encoders and network protocols focuses on performance and compatibility to deliver better performance and definition.

2.2. Role for LVB Scenario Only (TRTCAppSceneLIVE)

In the LVB scenario, most users are viewers, and only several users are anchors. The differentiation in roles can help TRTC implement better and more specific optimization.

• Anchor: anchor, who can upstream video and audio. Up to 20 anchors are allowed to upstream videos at the same time in one room.
• Audience: viewer, who can only watch the video but cannot upstream video or audio. There is no upper limit for the number of viewers in one room.

Constant	Description
TRTCRoleAnchor	Anchor.
TRTCRoleAudience	Viewer.

2.3. Bandwidth Limit Mode

The TRTC SDK needs to adjust the internal codecs and network module based on the network conditions in real-time to respond to network changes. To support fast algorithm upgrade, the SDK provides two network bandwidth limit modes:

• ModeServer: on-cloud control, which is the default and recommended mode.
• ModeClient: client-based control, which is for internal debugging of SDK and shall not be used by users.

You are recommended to use on-cloud control, so that when the QoS algorithm is upgraded, you do not need to upgrade the SDK to get a better experience.

Constant	Description
VIDEO_QOS_CONTROL_CLIENT	Client-based control (which is for internal debugging of SDK and shall not be used by users).
VIDEO_QOS_CONTROL_SERVER	On-cloud control (default).

2.4. Image Quality Preference

This specifies whether to "ensure smoothness" or "ensure definition" when the TRTC SDK is used on a weak network:

• Smooth: smoothness is ensured on a weak network, i.e., ensuring the smoothness and sending of the audio, while the video image will have a lot of blurs but can be smooth with no lagging.
• Clear: definition is ensured on a weak network, i.e., the image will be as clear as possible but tend to lag.

Constant	Description
TRTC_VIDEO_QOS_PREFERENCE_SMOOTH	Ensures smoothness on a weak network.
TRTC_VIDEO_QOS_PREFERENCE_CLEAR	Ensures definition on a weak network.

3.1. 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 application scenario, TRTCAudioSampleRate48000 will be recommended.

Constant	Description
TRTCAudioSampleRate16000	16 kHz sample rate.
TRTCAudioSampleRate32000	32 kHz sample rate.
TRTCAudioSampleRate44100	44.1 kHz sample rate.
TRTCAudioSampleRate48000	48 kHz sample rate.

3.2. Audio Playback Mode (Audio Routing)

Both the video call features in WeChat and Mobile QQ have a hands-free mode during call. This mode is implemented based on audio routing. Generally, a mobile phone has two speakers, and the purpose of setting audio routing is to determine which speaker will be used:

• Speakerphone: speaker at the bottom of a phone, which has high volume and is suitable for playing back music.
• Earpiece: receiver at the top of a phone, which has low volume and is suitable for calls.

Constant	Description
TRTC_AUDIO_ROUTE_SPEAKER	Speaker.
TRTC_AUDIO_ROUTE_EARPIECE	Receiver.

3.3. Audio Reverb Mode

Constant	Description
TRTC_REVERB_TYPE_0	Disables reverb.
TRTC_REVERB_TYPE_1	KTV
TRTC_REVERB_TYPE_2	Small room.
TRTC_REVERB_TYPE_3	Big hall.
TRTC_REVERB_TYPE_4	Deep.
TRTC_REVERB_TYPE_5	Resonant.
TRTC_REVERB_TYPE_6	Metallic.
TRTC_REVERB_TYPE_7	Husky.

3.4. Voice Charger Type

Constant	Description
TRTC_VOICE_CHANGER_TYPE_0	Disables the voice changer.
TRTC_VOICE_CHANGER_TYPE_1	Naughty boy.
TRTC_VOICE_CHANGER_TYPE_2	Little girl.
TRTC_VOICE_CHANGER_TYPE_3	Middle-aged man.
TRTC_VOICE_CHANGER_TYPE_4	Heavy metal.
TRTC_VOICE_CHANGER_TYPE_5	Being cold.
TRTC_VOICE_CHANGER_TYPE_6	Non-native speaker.
TRTC_VOICE_CHANGER_TYPE_7	Furious animal.
TRTC_VOICE_CHANGER_TYPE_8	Chubby.
TRTC_VOICE_CHANGER_TYPE_9	Strong electric current.
TRTC_VOICE_CHANGER_TYPE_10	Robot.
TRTC_VOICE_CHANGER_TYPE_11	Ethereal voice.

3.5. Audio Frame Format

Constant	Description
TRTC_AUDIO_FRAME_FORMAT_PCM	PCM

3.6. System Volume Type

Smartphones usually have two system volume types, i.e., call volume and media volume.

• Call volume: volume type dedicated to the VoIP scenario. It will forcibly enable acoustic echo cancellation (AEC) and has low sound quality, and its volume level cannot be set to 0.
• Media volume: volume type dedicated to music and movie playback through players. It will not enable AEC and can be set to 0 with the volume buttons on the phone. In media volume mode, AEC can be implemented by using the SDK's built-in acoustic algorithm if needed.

Currently, the SDK provides two control modes of system volume types, including:

• TRTCSystemVolumeTypeAuto: "call volume with mic and media volume without mic", i.e., the call volume mode will be used in a call, while the media volume mode will be used when a user only watches the video without speaking.
• TRTCSystemVolumeTypeMedia: the media volume will be used all the time. If a user is on a call, the SDK will use the built-in algorithm for AEC.

Constant	Description
TRTCSystemVolumeTypeAuto	Default type, in which the SDK automatically selects the appropriate volume type.
TRTCSystemVolumeTypeMedia	Only the media volume is used, and the SDK does not use the call volume.

4.1. UI Debugging Log

Constant	Description
TRTC_DEBUG_VIEW_LEVEL_GONE	The UI does not display logs.
TRTC_DEBUG_VIEW_LEVEL_STATUS	The upper part of the UI displays the status logs.
TRTC_DEBUG_VIEW_LEVEL_ALL	The upper part of the UI displays the status logs, and the lower part displays the key events.

4.2. Log Level

Different log levels indicate different levels of details and number of logs. It is recommended to set the log level to TRTC_LOG_LEVEL_INFO generally.

Constant	Description
TRTC_LOG_LEVEL_VERBOSE	Outputs logs at all levels.
TRTC_LOG_LEVEL_DEBUG	Outputs logs at the DEBUG, INFO, WARNING, ERROR, and FATAL levels.
TRTC_LOG_LEVEL_INFO	Outputs logs at the INFO, WARNING, ERROR, and FATAL levels.
TRTC_LOG_LEVEL_WARN	Outputs logs at the WARNING, ERROR, and FATAL levels.
TRTC_LOG_LEVEL_ERROR	Outputs logs at the ERROR and FATAL levels.
TRTC_LOG_LEVEL_FATAL	Outputs logs at the FATAL level.
TRTC_LOG_LEVEL_NULL	No SDK logs will be output.

4.3. G-sensor Switch

This configuration item applies to mobile devices such as phones and tablets, and needs to be used together with the current UI layout mode

• Disable: you can select this value if you do not want the video image direction to be adjusted along with the sensed gravity direction.
• UIAutoLayout: the SDK will not automatically adjust the rotation direction of LocalVideoView, which will be adjusted by the system instead. It can be used only if the application UI supports gravity sensing.
• UIFixLayout: the SDK will automatically adjust the rotation direction of LocalVideoView and can be used if the application UI currently does not support gravity sensing.

Constant	Description
TRTC_GSENSOR_MODE_DISABLE	Disables the g-sensor.
TRTC_GSENSOR_MODE_UIAUTOLAYOUT	Enables the g-sensor, which can be used if the application UI supports gravity sensing.
TRTC_GSENSOR_MODE_UIFIXLAYOUT	Enables the g-sensor, which can be used if the application UI currently does not support gravity sensing.

4.4. MixTranscoding Parameter Configuration Mode

Currently, only the manual configuration mode is supported, i.e., all parameters of TRTC_TranscodingConfigMode_Manual need to be set manually.

Constant	Description
TRTC_TranscodingConfigMode_Unknown	Undefined.
TRTC_TranscodingConfigMode_Manual	Manual configuration of MixTranscoding parameters. All parameters of TRTCTranscodingConfig need to be specified.

TRTCParams

Feature

Room entry parameters.

Overview

As the room entry parameters in the TRTC SDK, only if these parameters are correctly set can the user successfully enter the audio/video room specified by roomId.

Attribute list

Attribute	Type	Description	Recommended Value
sdkAppId	int	Application ID, which is required. Tencent Cloud generates bills based on sdkAppId.	-
userId	String	User ID, which is required. It is the userId of the local user and acts as the username.	-
userSig	String	User signature, which is required. It is the authentication signature corresponding to the current userId and acts as the login password.	-
roomId	int	Room number, which is required. After the room number is specified, users (userIds) in the same room can see one another and make video calls.	-
role	int	Role, which applies only to the LVB scenario (TRTCAppSceneLIVE) and does not take effect in the video call scenario.	Default value: anchor (TRTCRoleAnchor).
privateMapKey	String	Room signature, which is optional. If you want users with only specified userIds to enter a room (roomId), you need to use privateMapKey to restrict the permission.	-
businessInfo	String	Business data, which is optional. This field applies only to some uncommon advanced features.	-

TRTCVideoEncParam

Feature

Encoding parameters.

Overview

Parameters related to video encoder. These settings determine the quality of image viewed by remote users, which is also the image quality of recorded video files in the cloud.

Attribute list

Attribute	Type	Description	Recommended Value	Remarks
videoResolution	int	Video resolution.	- For video calls, it is recommended to select 360 * 640 or lower for resolution and Portrait for resMode. - For MLVB, it is recommended to select 540 * 960 for resolution and Portrait for resMode. - For Window and macOS, it is recommended to select 640 * 360 or higher for resolution and Landscape for resMode.	Resolution set in TRTCVideoResolution is only in landscape mode by default, e.g., 640 * 360. If resolution in portrait mode is required, please select Portrait for resMode; for example, 640 * 360 will become 360 * 640 in portrait mode.
videoResolutionMode	int	Resolution mode (landscape/portrait).	For MLVB, it is recommended to select Portrait. For Window and macOS, it is recommended to select Landscape.	If 640 * 360 resolution is selected for videoResolution and Portrait is selected for resMode, then the final output resolution after encoding will be 360 * 640.
videoFps	int	Video capture frame rate.	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, too many resources will be wasted (the frame rate of movies is generally 24 fps).	The front cameras on many Android phones do not support a capture frame rate higher than 15 fps. For some Android phones that focus too much on beautification features, the capture frame rate of the front cameras may be lower than 10 fps.
videoBitrate	int	Video upstreaming bitrate.	For more information on the recommended settings, please see the description in the definition of TRTCVideoResolution in the first half of this document.	If the bitrate is too low, the video will have a lot of blurs.

TRTCNetworkQosParam

Feature

Network bandwidth limit parameters.

Overview

The settings determine the bandwidth limit practices of the SDK in various network conditions (e.g., whether to "ensure definition" or "ensure smoothness" on a weak network).

Attribute list

Attribute	Type	Description	Recommended Value	Remarks
preference	int	Whether to select "ensure definition" or "ensure smoothness" on a weak network.	-	- Smoothness on weak network: on a weak network, the video image will have a lot of blurs but can be smooth with no lagging - Definition on weak network: the image will be as clear as possible on a weak network but tend to have more lagging.
controlMode	int	Video resolution (on-cloud control - client-based control).	On-cloud control.	- Server mode (default): on-cloud control. If there are no special needs, please use this mode directly - Client mode: client-based control, which is for internal debugging of the SDK and shall not be used by users.

TRTCQuality

Feature

Video (or network) quality.

Attribute list

Attribute	Type	Description
userId	String	User ID.
quality	int	Network quality. For more information on the definition, please see Video (or Network) Quality Constant Definition.

TRTCTexture

Feature

Video texture data, including texture ID and EGL environment.

Attribute list

Attribute	Type	Description
textureId	int	Video texture ID.
eglContext10	javax.microedition.khronos.egl.EGLContext	OpenGL API defined by (javax.microedition.khronos.egl.*).
eglContext14	android.opengl.EGLContext	OpenGL API defined by (android.opengl.*).

TRTCVideoFrame

Feature

Video frame information.

Overview

TRTCVideoFrame is used to describe the raw data of a frame of video image, which is either the image before frame encoding or after frame decoding.

Attribute list

Attribute	Type	Description	Recommended Value
pixelFormat	int	Video pixel format. For more information on the definition, please see Video Pixel Format.	Custom capture: TRTC_VIDEO_PIXEL_FORMAT_Texture_2D or TRTC_VIDEO_PIXEL_FORMAT_NV21; Custom rendering: TRTC_VIDEO_PIXEL_FORMAT_I420.
bufferType	int	Video data container format. For more information on the definition, please see Video Data Container Format.	Custom capture: TRTC_VIDEO_BUFFER_TYPE_TEXTURE; Custom rendering: TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER or TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY.
texture	TRTCTexture	Video data when bufferType is TRTC_VIDEO_PIXEL_FORMAT_Texture_2D.	-
data	byte[]	Video data when bufferType is TRTC_VIDEO_BUFFER_TYPE_BYTE_ARRAY.	-
buffer	ByteBuffer	Video data when bufferType is TRTC_VIDEO_BUFFER_TYPE_BYTE_BUFFER, which is mainly used at the Native layer.	-
width	int	Video width.	Please strictly enter the width of the video data passed in.
height	int	Video height.	Please strictly enter the height of the video data passed in.
timestamp	long	Video frame timestamp in milliseconds.	This parameter can be set to 0 for custom video capture, and the SDK will automatically set the timestamp field. However, please "evenly" set the calling interval of sendCustomVideoData.
rotation	int	Clockwise rotation angle of video pixel.	This parameter can be left empty if custom video capture is enabled.

TRTCAudioFrame

Feature

Audio frame data.

Attribute list

Attribute	Type	Description
data	byte []	Audio data.
sampleRate	int	Sample rate.
channel	int	Number of sound channels.
timestamp	long	Timestamp.

TRTCVolumeInfo

Feature

Volume level.

Overview

This indicates the audio volume level, based on which corresponding icons can be displayed on UI to indicate whether a userId is speaking.

Attribute list

Attribute	Type	Description
userId	String	userId of the speaking user. An empty value indicates the local user.
volume	int	Volume level of the speaking user. Value range: 0–100.

TRTCSpeedTestResult

Feature

Network speed test result.

Overview

The startSpeedTest API of TRTCCloud can be used to test the network speed before a user enters a room (this API cannot be called during a call). The speed test result will be returned once every 2–3 seconds, and the test result of one IP address will be returned each time.

Attribute list

Attribute	Type	Description
ip	String	Server IP address.
quality	int	Network quality, which is tested and calculated based on the internal evaluation algorithm. The smaller the loss and round-trip time (RTT), the higher the network quality score.
upLostRate	float	Upstreaming 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.
downLostRate	float	Downstreaming 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.
rtt	int	Delay in milliseconds, which is the round-trip time between the current device and CVM instance. The smaller the value, the better. The normal value range is 10–100 ms.

TRTCMixUser

Feature

Position information of each channel of subimage in On-Cloud MixTranscoding.

Overview

TRTCMixUser is used to specify the detailed position of the video image of each channel (i.e., each userId).

Attribute list

Attribute	Type	Description
userId	String	userId that engages in mixing.
roomId	String	roomId of the userId that engages in mixing. The null value indicates the current room.
x	int	X coordinate of the layer position (absolute pixel value).
y	int	Y coordinate of the layer position (absolute pixel value).
width	int	Width of the layer position (absolute pixel value).
height	int	Height of the layer position (absolute pixel value).
zOrder	int	Layer number (1–15), which must be unique.
streamType	int	Whether the primary image (TRTC_VIDEO_STREAM_TYPE_BIG, which is default) or screen sharing image (TRTC_VIDEO_STREAM_TYPE_SUB) engages in mixing.
pureAudio	boolean	Whether the stream engages in mixing is pure audio stream.

TRTCTranscodingConfig

Feature

On-Cloud MixTranscoding configuration.

Overview

This contains the final encoding quality and the positions of images of each channel.

Attribute list

Attribute	Type	Description	Recommended Value
appId	int	Tencent Cloud LVB application ID.	Please select a created application in the TRTC Console, click Account Info, and get the ID in "LVB Info".
bizId	int	Tencent Cloud LVB business ID.	Please select a created application in the TRTC Console, click Account Info, and get the ID in "LVB Info".
mode	int	Transcoding config mode.	-
videoWidth	int	Width of video resolution in px after being transcoded.	-
videoHeight	int	Height of video resolution in px after being transcoded.	-
videoBitrate	int	Bitrate of video resolution in Kbps after being transcoded.	-
videoFramerate	int	Frame rate of video resolution in FPS after being transcoded.	15
videoGOP	int	GOP interval of video resolution in seconds after being transcoded.	3
audioSampleRate	int	Audio sample rate after being transcoded.	48000
audioBitrate	int	Audio bitrate in Kbps after being transcoded.	64
audioChannels	int	Number of sound channels after being transcoded.	2
mixUsers	ArrayListTRTCMixUser	Position information of each channel of subimage.	-

TRTCPublishCDNParam

Feature

Relayed push parameters.

Attribute list

Attribute	Type	Description
appId	int	Tencent Cloud application ID. Please select a created application in the TRTC Console, click Account Info, and get the ID in "LVB Info".
bizId	int	Tencent Cloud LVB business ID. Please select a created application in the TRTC Console, click Account Info, and get the ID in "LVB Info".
url	String	Relayed push URL.

TRTCAudioRecordingParams

Feature

Audio recording parameters.

Overview

Please set the parameters correctly to ensure that the audio recording file can be successfully generated.

Attribute list

Attribute	Type	Description	Remarks
filePath	String	File path, which is the path to the audio recording file and is required. The path needs to be specified by the user and must exist in the application and be writable.	The file name and its extension need to be specified in the path, and the extension determines the format of the audio recording file. Currently supported formats are PCM, WAV, and AAC. For example, if the specified path is path/to/audio.aac, a file in AAC format will be generated. Please specify a valid path that is readable/writable; otherwise, the audio recording file cannot be generated.

TRTCAudioEffectParam

Feature

Audio effect.

Attribute list

Attribute	Type	Description	Remarks
effectId	int	Audio effect ID. Each audio effect needs to be assigned with a unique ID, through which the audio effect can be manipulated, such as starting/stopping and adjusting volume level.	-
path	String	Absolute path to the audio effect file.	-
loopCount	int	Number of loop playback times of audio effect.	Value range: 0 or any positive integer. Default value: 0. 0 indicates that the audio effect will be played back once; 1 indicates that the audio effect will be played back twice; and so on.
publish	boolean	Whether to send audio effect to remote users.	YES: when the audio effect is played back locally, it will be upstreamed to the cloud and can be heard by remote users; NO: the audio effect will not be upstreamed to the cloud and can only be heard locally. Default value: NO.
volume	int	Volume level of audio effect.	Value range: 0–100. Default value: 100.

TRTCStatistics

Feature

Statistics.

Attribute list

Attribute	Type	Description
appCpu	int	CPU utilization of the current application in percent (%).
systemCpu	int	CPU utilization of the current system in percent (%).
rtt	int	Delay in milliseconds, which is the round-trip time between the SDK and server. The smaller the value, the better. Generally rtt lower than 50 ms is satisfactory, while rtt higher than 100 ms will result in long call latency. As data upstreaming and downstreaming share the same network connection, rtt is the same for the local user and remote user.
upLoss	int	Client-to-server upstream packet loss rate in percent (%). The smaller the value, the better. For example, if the packet loss rate is 0, it means the network conditions are good. If the value is 30%, it indicates that 30% of data packets sent to the server by the SDK were lost during upstreaming.
downLoss	int	Server-to-client downstream packet loss rate in percent (%). The smaller the value, the better. For example, if the packet loss rate is 0, it means the network conditions are good. If the value is 30%, it indicates that 30% of data packets sent to the server by the SDK were lost during downstreaming.
sendBytes	long	Total number of bytes sent. Note that this is the number of bytes but not bits.
receiveBytes	long	Total number of bytes received. Note that this is the number of bytes but not bits.
localArray	ArrayListTRTCLocalStatistics	Local audio/video statistics, which are an array since they may contain statistics of multiple channels, such as big image, small image, and secondary channel image.
remoteArray	ArrayListTRTCRemoteStatistics	Remote audio/video statistics, which are an array since they may contain statistics of multiple channels, such as big image, small image, and secondary channel image.

TRTCRemoteStatistics

Feature

Audio/video statistics of a remote user.

Attribute list

Attribute	Type	Description
userId	String	User ID, which specifies the user whose video stream is to be queried.
finalLoss	int	Total packet loss rate of this line in percent (%). A smaller value is preferred; for example, if the packet loss rate is 0, it means the network conditions are good. The packet loss rate is the total packet loss rate in one turn of upstreaming and downstreaming between the userId and server. If downLoss is 0% but finalLoss is not 0, it indicates that there was an irrecoverable packet loss in upstreaming from the userId.
width	int	Video width.
height	int	Video height.
frameRate	int	Receipt frame rate in fps.
videoBitrate	int	Video bitrate in Kbps.
audioSampleRate	int	Audio sample rate in Hz.
audioBitrate	int	Audio bitrate in Kbps.
streamType	int	Stream type (big image | small image | secondary channel image).

TRTCLocalStatistics

Feature

Local audio/video statistics.

Attribute list

Attribute	Type	Description
width	int	Video width.
height	int	Video height.
frameRate	int	Frame rate in fps.
videoBitrate	int	Video sending bitrate in Kbps.
audioSampleRate	int	Audio sample rate in Hz.
audioBitrate	int	Audio sending bitrate in Kbps.
streamType	int	Stream type (big image | small image | secondary channel image).